| QUIC | -M. Bishop, Ed. | -
| Internet-Draft | -Akamai | -
| Intended status: Standards Track | -October 22, 2019 | -
| Expires: April 24, 2020 | -- |
Hypertext Transfer Protocol Version 3 (HTTP/3)
- draft-ietf-quic-http-latest
The QUIC transport protocol has several features that are desirable in a transport for HTTP, such as stream multiplexing, per-stream flow control, and low-latency connection establishment. This document describes a mapping of HTTP semantics over QUIC. This document also identifies HTTP/2 features that are subsumed by QUIC, and describes how HTTP/2 extensions can be ported to HTTP/3.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-http.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
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- - - -HTTP semantics are used for a broad range of services on the Internet. These semantics have commonly been used with two different TCP mappings, HTTP/1.1 and HTTP/2. HTTP/3 supports the same semantics over a new transport protocol, QUIC.
-HTTP/1.1 is a TCP mapping which uses whitespace-delimited text fields to convey HTTP messages. While these exchanges are human-readable, using whitespace for message formatting leads to parsing difficulties and workarounds to be tolerant of variant behavior. Because each connection can transfer only a single HTTP request or response at a time in each direction, multiple parallel TCP connections are often used, reducing the ability of the congestion controller to accurately manage traffic between endpoints.
-HTTP/2 introduced a binary framing and multiplexing layer to improve latency without modifying the transport layer. However, because the parallel nature of HTTP/2’s multiplexing is not visible to TCP’s loss recovery mechanisms, a lost or reordered packet causes all active transactions to experience a stall regardless of whether that transaction was impacted by the lost packet.
-The QUIC transport protocol incorporates stream multiplexing and per-stream flow control, similar to that provided by the HTTP/2 framing layer. By providing reliability at the stream level and congestion control across the entire connection, it has the capability to improve the performance of HTTP compared to a TCP mapping. QUIC also incorporates TLS 1.3 at the transport layer, offering comparable security to running TLS over TCP, with the improved connection setup latency of TCP Fast Open [RFC7413].
-This document defines a mapping of HTTP semantics over the QUIC transport protocol, drawing heavily on the design of HTTP/2. While delegating stream lifetime and flow control issues to QUIC, a similar binary framing is used on each stream. Some HTTP/2 features are subsumed by QUIC, while other features are implemented atop QUIC.
-QUIC is described in [QUIC-TRANSPORT]. For a full description of HTTP/2, see [HTTP2].
-HTTP/3 provides a transport for HTTP semantics using the QUIC transport protocol and an internal framing layer similar to HTTP/2.
-Once a client knows that an HTTP/3 server exists at a certain endpoint, it opens a QUIC connection. QUIC provides protocol negotiation, stream-based multiplexing, and flow control. An HTTP/3 endpoint can be discovered using HTTP Alternative Services; this process is described in greater detail in Section 3.2.
-Within each stream, the basic unit of HTTP/3 communication is a frame (Section 7.2). Each frame type serves a different purpose. For example, HEADERS and DATA frames form the basis of HTTP requests and responses (Section 4.1).
-Multiplexing of requests is performed using the QUIC stream abstraction, described in Section 2 of [QUIC-TRANSPORT]. Each request and response consumes a single QUIC stream. Streams are independent of each other, so one stream that is blocked or suffers packet loss does not prevent progress on other streams.
-Server push is an interaction mode introduced in HTTP/2 [HTTP2] which permits a server to push a request-response exchange to a client in anticipation of the client making the indicated request. This trades off network usage against a potential latency gain. Several HTTP/3 frames are used to manage server push, such as PUSH_PROMISE, DUPLICATE_PUSH, MAX_PUSH_ID, and CANCEL_PUSH.
-As in HTTP/2, request and response headers are compressed for transmission. Because HPACK [HPACK] relies on in-order transmission of compressed header blocks (a guarantee not provided by QUIC), HTTP/3 replaces HPACK with QPACK [QPACK]. QPACK uses separate unidirectional streams to modify and track header table state, while header blocks refer to the state of the table without modifying it.
-The HTTP/3 specification is split into seven parts. The document begins with a detailed overview of the connection lifecycle and key concepts:
- - -The details of the wire protocol and interactions with the transport are described in subsequent sections:
- - -Additional resources are provided in the final sections:
- - -The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Field definitions are given in Augmented Backus-Naur Form (ABNF), as defined in [RFC5234].
-This document uses the variable-length integer encoding from [QUIC-TRANSPORT].
-The following terms are used:
- - -The term “payload body” is defined in Section 3.3 of [RFC7230].
-Finally, the terms “gateway”, “intermediary”, “proxy”, and “tunnel” are defined in Section 2.3 of [RFC7230]. Intermediaries act as both client and server at different times.
-HTTP/3 uses the token “h3” to identify itself in ALPN and Alt-Svc. Only implementations of the final, published RFC can identify themselves as “h3”. Until such an RFC exists, implementations MUST NOT identify themselves using this string.
-Implementations of draft versions of the protocol MUST add the string “-“ and the corresponding draft number to the identifier. For example, draft-ietf-quic-http-01 is identified using the string “h3-01”.
-Non-compatible experiments that are based on these draft versions MUST append the string “-“ and an experiment name to the identifier. For example, an experimental implementation based on draft-ietf-quic-http-09 which reserves an extra stream for unsolicited transmission of 1980s pop music might identify itself as “h3-09-rickroll”. Note that any label MUST conform to the “token” syntax defined in Section 3.2.6 of [RFC7230]. Experimenters are encouraged to coordinate their experiments on the quic@ietf.org mailing list.
-An HTTP origin advertises the availability of an equivalent HTTP/3 endpoint via the Alt-Svc HTTP response header field or the HTTP/2 ALTSVC frame ([ALTSVC]), using the ALPN token defined in Section 3.3.
-For example, an origin could indicate in an HTTP response that HTTP/3 was available on UDP port 50781 at the same hostname by including the following header field:
--Alt-Svc: h3=":50781" --
On receipt of an Alt-Svc record indicating HTTP/3 support, a client MAY attempt to establish a QUIC connection to the indicated host and port and, if successful, send HTTP requests using the mapping described in this document.
-Connectivity problems (e.g. firewall blocking UDP) can result in QUIC connection establishment failure, in which case the client SHOULD continue using the existing connection or try another alternative endpoint offered by the origin.
-Servers MAY serve HTTP/3 on any UDP port, since an alternative always includes an explicit port.
-HTTP/3 relies on QUIC as the underlying transport. The QUIC version being used MUST use TLS version 1.3 or greater as its handshake protocol. HTTP/3 clients MUST indicate the target domain name during the TLS handshake. This may be done using the Server Name Indication (SNI) [RFC6066] extension to TLS or using some other mechanism.
-QUIC connections are established as described in [QUIC-TRANSPORT]. During connection establishment, HTTP/3 support is indicated by selecting the ALPN token “h3” in the TLS handshake. Support for other application-layer protocols MAY be offered in the same handshake.
-While connection-level options pertaining to the core QUIC protocol are set in the initial crypto handshake, HTTP/3-specific settings are conveyed in the SETTINGS frame. After the QUIC connection is established, a SETTINGS frame (Section 7.2.4) MUST be sent by each endpoint as the initial frame of their respective HTTP control stream (see Section 6.2.1).
-Once a connection exists to a server endpoint, this connection MAY be reused for requests with multiple different URI authority components. The client MAY send any requests for which the client considers the server authoritative.
-An authoritative HTTP/3 endpoint is typically discovered because the client has received an Alt-Svc record from the request’s origin which nominates the endpoint as a valid HTTP Alternative Service for that origin. As required by [RFC7838], clients MUST check that the nominated server can present a valid certificate for the origin before considering it authoritative. Clients MUST NOT assume that an HTTP/3 endpoint is authoritative for other origins without an explicit signal.
-Prior to making requests for an origin whose scheme is not “https,” the client MUST ensure the server is willing to serve that scheme. If the client intends to make requests for an origin whose scheme is “http”, this means that it MUST obtain a valid http-opportunistic response for the origin as described in [RFC8164] prior to making any such requests. Other schemes might define other mechanisms.
-A server that does not wish clients to reuse connections for a particular origin can indicate that it is not authoritative for a request by sending a 421 (Misdirected Request) status code in response to the request (see Section 9.1.2 of [HTTP2]).
-The considerations discussed in Section 9.1 of [HTTP2] also apply to the management of HTTP/3 connections.
-A client sends an HTTP request on a client-initiated bidirectional QUIC stream. A client MUST send only a single request on a given stream. A server sends zero or more non-final HTTP responses on the same stream as the request, followed by a single final HTTP response, as detailed below.
-An HTTP message (request or response) consists of:
- - -A server MAY send one or more PUSH_PROMISE frames (see Section 7.2.5) before, after, or interleaved with the frames of a response message. These PUSH_PROMISE frames are not part of the response; see Section 4.4 for more details.
-Frames of unknown types (Section 9), including reserved frames (Section 7.2.9) MAY be sent on a request or push stream before, after, or interleaved with other frames described in this section.
-The HEADERS and PUSH_PROMISE frames might reference updates to the QPACK dynamic table. While these updates are not directly part of the message exchange, they must be received and processed before the message can be consumed. See Section 4.1.1 for more details.
-The “chunked” transfer encoding defined in Section 4.1 of [RFC7230] MUST NOT be used.
-A response MAY consist of multiple messages when and only when one or more informational responses (1xx; see [RFC7231], Section 6.2) precede a final response to the same request. Non-final responses do not contain a payload body or trailers.
-If an endpoint receives an invalid sequence of frames on either a request or a push stream, it MUST respond with a connection error of type HTTP_FRAME_UNEXPECTED (Section 8). In particular, a DATA frame before any HEADERS frame, or a HEADERS or DATA frame after the trailing HEADERS frame is considered invalid.
-An HTTP request/response exchange fully consumes a bidirectional QUIC stream. After sending a request, a client MUST close the stream for sending. Unless using the CONNECT method (see Section 4.2), clients MUST NOT make stream closure dependent on receiving a response to their request. After sending a final response, the server MUST close the stream for sending. At this point, the QUIC stream is fully closed.
-When a stream is closed, this indicates the end of an HTTP message. Because some messages are large or unbounded, endpoints SHOULD begin processing partial HTTP messages once enough of the message has been received to make progress. If a client stream terminates without enough of the HTTP message to provide a complete response, the server SHOULD abort its response with the error code HTTP_REQUEST_INCOMPLETE.
-A server can send a complete response prior to the client sending an entire request if the response does not depend on any portion of the request that has not been sent and received. When this is true, a server MAY abort reading the request stream with error code HTTP_EARLY_RESPONSE, send a complete response, and cleanly close the sending part of the stream. Clients MUST NOT discard complete responses as a result of having their request terminated abruptly, though clients can always discard responses at their discretion for other reasons.
-HTTP message headers carry information as a series of key-value pairs, called header fields. For a listing of registered HTTP header fields, see the “Message Header Field” registry maintained at https://www.iana.org/assignments/message-headers.
-Just as in previous versions of HTTP, header field names are strings of ASCII characters that are compared in a case-insensitive fashion. Properties of HTTP header field names and values are discussed in more detail in Section 3.2 of [RFC7230], though the wire rendering in HTTP/3 differs. As in HTTP/2, header field names MUST be converted to lowercase prior to their encoding. A request or response containing uppercase header field names MUST be treated as malformed (Section 4.1.3).
-As in HTTP/2, HTTP/3 uses special pseudo-header fields beginning with the ‘:’ character (ASCII 0x3a) to convey the target URI, the method of the request, and the status code for the response. These pseudo-header fields are defined in Section 8.1.2.3 and 8.1.2.4 of [HTTP2]. Pseudo-header fields are not HTTP header fields. Endpoints MUST NOT generate pseudo-header fields other than those defined in [HTTP2]. The restrictions on the use of pseudo-header fields in Section 8.1.2 of [HTTP2] also apply to HTTP/3. Messages which are considered malformed under these restrictions are handled as described in Section 4.1.3.
-HTTP/3 uses QPACK header compression as described in [QPACK], a variation of HPACK which allows the flexibility to avoid header-compression-induced head-of-line blocking. See that document for additional details.
-To allow for better compression efficiency, the cookie header field [RFC6265] MAY be split into separate header fields, each with one or more cookie-pairs, before compression. If a decompressed header list contains multiple cookie header fields, these MUST be concatenated before being passed into a non-HTTP/2, non-HTTP/3 context, as described in [HTTP2], Section 8.1.2.5.
-An HTTP/3 implementation MAY impose a limit on the maximum size of the message header it will accept on an individual HTTP message. A server that receives a larger header field list than it is willing to handle can send an HTTP 431 (Request Header Fields Too Large) status code [RFC6585]. A client can discard responses that it cannot process. The size of a header field list is calculated based on the uncompressed size of header fields, including the length of the name and value in bytes plus an overhead of 32 bytes for each header field.
-If an implementation wishes to advise its peer of this limit, it can be conveyed as a number of bytes in the SETTINGS_MAX_HEADER_LIST_SIZE parameter. An implementation which has received this parameter SHOULD NOT send an HTTP message header which exceeds the indicated size, as the peer will likely refuse to process it. However, because this limit is applied at each hop, messages below this limit are not guaranteed to be accepted.
-Clients can cancel requests by resetting and aborting the request stream with an error code of HTTP_REQUEST_CANCELLED (Section 8.1). When the client aborts reading a response, it indicates that this response is no longer of interest. Implementations SHOULD cancel requests by abruptly terminating any directions of a stream that are still open.
-When the server rejects a request without performing any application processing, it SHOULD abort its response stream with the error code HTTP_REQUEST_REJECTED. In this context, “processed” means that some data from the stream was passed to some higher layer of software that might have taken some action as a result. The client can treat requests rejected by the server as though they had never been sent at all, thereby allowing them to be retried later on a new connection. Servers MUST NOT use the HTTP_REQUEST_REJECTED error code for requests which were partially or fully processed. When a server abandons a response after partial processing, it SHOULD abort its response stream with the error code HTTP_REQUEST_CANCELLED.
-When a client resets a request with the error code HTTP_REQUEST_CANCELLED, a server MAY abruptly terminate the response using the error code HTTP_REQUEST_REJECTED if no processing was performed. Clients MUST NOT use the HTTP_REQUEST_REJECTED error code, except when a server has requested closure of the request stream with this error code.
-If a stream is cancelled after receiving a complete response, the client MAY ignore the cancellation and use the response. However, if a stream is cancelled after receiving a partial response, the response SHOULD NOT be used. Automatically retrying such requests is not possible, unless this is otherwise permitted (e.g., idempotent actions like GET, PUT, or DELETE).
-A malformed request or response is one that is an otherwise valid sequence of frames but is invalid due to the presence of extraneous frames, prohibited header fields, the absence of mandatory header fields, or the inclusion of uppercase header field names.
-A request or response that includes a payload body can include a content-length header field. A request or response is also malformed if the value of a content-length header field does not equal the sum of the DATA frame payload lengths that form the body. A response that is defined to have no payload, as described in Section 3.3.2 of [RFC7230] can have a non-zero content-length header field, even though no content is included in DATA frames.
-Intermediaries that process HTTP requests or responses (i.e., any intermediary not acting as a tunnel) MUST NOT forward a malformed request or response. Malformed requests or responses that are detected MUST be treated as a stream error (Section 8) of type HTTP_GENERAL_PROTOCOL_ERROR.
-For malformed requests, a server MAY send an HTTP response prior to closing or resetting the stream. Clients MUST NOT accept a malformed response. Note that these requirements are intended to protect against several types of common attacks against HTTP; they are deliberately strict because being permissive can expose implementations to these vulnerabilities.
-The pseudo-method CONNECT ([RFC7231], Section 4.3.6) is primarily used with HTTP proxies to establish a TLS session with an origin server for the purposes of interacting with “https” resources. In HTTP/1.x, CONNECT is used to convert an entire HTTP connection into a tunnel to a remote host. In HTTP/2, the CONNECT method is used to establish a tunnel over a single HTTP/2 stream to a remote host for similar purposes.
-A CONNECT request in HTTP/3 functions in the same manner as in HTTP/2. The request MUST be formatted as described in [HTTP2], Section 8.3. A CONNECT request that does not conform to these restrictions is malformed (see Section 4.1.3). The request stream MUST NOT be closed at the end of the request.
-A proxy that supports CONNECT establishes a TCP connection ([RFC0793]) to the server identified in the “:authority” pseudo-header field. Once this connection is successfully established, the proxy sends a HEADERS frame containing a 2xx series status code to the client, as defined in [RFC7231], Section 4.3.6.
-All DATA frames on the stream correspond to data sent or received on the TCP connection. Any DATA frame sent by the client is transmitted by the proxy to the TCP server; data received from the TCP server is packaged into DATA frames by the proxy. Note that the size and number of TCP segments is not guaranteed to map predictably to the size and number of HTTP DATA or QUIC STREAM frames.
-Once the CONNECT method has completed, only DATA frames are permitted to be sent on the stream. Extension frames MAY be used if specifically permitted by the definition of the extension. Receipt of any other frame type MUST be treated as a connection error of type HTTP_FRAME_UNEXPECTED.
-The TCP connection can be closed by either peer. When the client ends the request stream (that is, the receive stream at the proxy enters the “Data Recvd” state), the proxy will set the FIN bit on its connection to the TCP server. When the proxy receives a packet with the FIN bit set, it will terminate the send stream that it sends to the client. TCP connections which remain half-closed in a single direction are not invalid, but are often handled poorly by servers, so clients SHOULD NOT close a stream for sending while they still expect to receive data from the target of the CONNECT.
-A TCP connection error is signaled by abruptly terminating the stream. A proxy treats any error in the TCP connection, which includes receiving a TCP segment with the RST bit set, as a stream error of type HTTP_CONNECT_ERROR (Section 8.1). Correspondingly, if a proxy detects an error with the stream or the QUIC connection, it MUST close the TCP connection. If the underlying TCP implementation permits it, the proxy SHOULD send a TCP segment with the RST bit set.
-HTTP/3 does not support the HTTP Upgrade mechanism ([RFC7230], Section 6.7) or 101 (Switching Protocols) informational status code ([RFC7231], Section 6.2.2).
-Server push is an interaction mode introduced in HTTP/2 [HTTP2] which permits a server to push a request-response exchange to a client in anticipation of the client making the indicated request. This trades off network usage against a potential latency gain. HTTP/3 server push is similar to what is described in HTTP/2 [HTTP2], but uses different mechanisms.
-Each server push is identified by a unique Push ID. This Push ID is used in a single PUSH_PROMISE frame (see Section 7.2.5) which carries the request headers, possibly included in one or more DUPLICATE_PUSH frames (see Section 7.2.8), then included with the push stream which ultimately fulfills those promises.
-Server push is only enabled on a connection when a client sends a MAX_PUSH_ID frame (see Section 7.2.7). A server cannot use server push until it receives a MAX_PUSH_ID frame. A client sends additional MAX_PUSH_ID frames to control the number of pushes that a server can promise. A server SHOULD use Push IDs sequentially, starting at 0. A client MUST treat receipt of a push stream with a Push ID that is greater than the maximum Push ID as a connection error of type HTTP_ID_ERROR.
-The header of the request message is carried by a PUSH_PROMISE frame (see Section 7.2.5) on the request stream which generated the push. This allows the server push to be associated with a client request. Promised requests MUST conform to the requirements in Section 8.2 of [HTTP2].
-The same server push can be associated with additional client requests using a DUPLICATE_PUSH frame (see Section 7.2.8).
-Ordering of a PUSH_PROMISE or DUPLICATE_PUSH in relation to certain parts of the response is important. The server SHOULD send PUSH_PROMISE or DUPLICATE_PUSH frames prior to sending HEADERS or DATA frames that reference the promised responses. This reduces the chance that a client requests a resource that will be pushed by the server.
-When a server later fulfills a promise, the server push response is conveyed on a push stream (see Section 6.2.2). The push stream identifies the Push ID of the promise that it fulfills, then contains a response to the promised request using the same format described for responses in Section 4.1.
-Due to reordering, DUPLICATE_PUSH frames or push stream data can arrive before the corresponding PUSH_PROMISE frame. When a client receives a DUPLICATE_PUSH frame for an as-yet-unknown Push ID, the request headers of the push are not immediately available. The client can either delay generating new requests for content referenced following the DUPLICATE_PUSH frame until the request headers become available, or can initiate requests for discovered resources and cancel the requests if the requested resource is already being pushed. When a client receives a new push stream with an as-yet-unknown Push ID, both the associated client request and the pushed request headers are unknown. The client can buffer the stream data in expectation of the matching PUSH_PROMISE. The client can use stream flow control (see section 4.1 of [QUIC-TRANSPORT]) to limit the amount of data a server may commit to the pushed stream.
-If a promised server push is not needed by the client, the client SHOULD send a CANCEL_PUSH frame. If the push stream is already open or opens after sending the CANCEL_PUSH frame, the client can abort reading the stream with an error code of HTTP_REQUEST_CANCELLED. This asks the server not to transfer additional data and indicates that it will be discarded upon receipt.
-Once established, an HTTP/3 connection can be used for many requests and responses over time until the connection is closed. Connection closure can happen in any of several different ways.
-Each QUIC endpoint declares an idle timeout during the handshake. If the connection remains idle (no packets received) for longer than this duration, the peer will assume that the connection has been closed. HTTP/3 implementations will need to open a new connection for new requests if the existing connection has been idle for longer than the server’s advertised idle timeout, and SHOULD do so if approaching the idle timeout.
-HTTP clients are expected to request that the transport keep connections open while there are responses outstanding for requests or server pushes, as described in Section 19.2 of [QUIC-TRANSPORT]. If the client is not expecting a response from the server, allowing an idle connection to time out is preferred over expending effort maintaining a connection that might not be needed. A gateway MAY maintain connections in anticipation of need rather than incur the latency cost of connection establishment to servers. Servers SHOULD NOT actively keep connections open.
-Even when a connection is not idle, either endpoint can decide to stop using the connection and initiate a graceful connection close. Endpoints initiate the graceful shutdown of a connection by sending a GOAWAY frame (Section 7.2.6). The GOAWAY frame contains an identifier that indicates to the receiver the range of requests or pushes that were or might be processed in this connection. The server sends a client-initiated bidirectional Stream ID; the client sends a Push ID. Requests or pushes with the indicated identifier or greater are rejected by the sender of the GOAWAY. This identifier MAY be zero if no requests or pushes were processed.
-The information in the GOAWAY enables a client and server to agree on which requests or pushes were accepted prior to the connection shutdown. Endpoints SHOULD abruptly terminate any requests or pushes that have identifiers greater than or equal to the smallest identifier sent in a GOAWAY frame.
-Endpoints MUST NOT initiate new requests or pushes on the connection with an identifier greater than or equal to the smallest identifier received in a GOAWAY frame. Clients MAY establish a new connection to send additional requests.
-Some requests or pushes might already be in transit. If the endpoint has already sent requests or pushes with an identifier greater than or equal to that received in a GOAWAY frame, those requests or pushes will not be processed; requests MAY be retried by the client on a different connection. The endpoint that initiated these requests or pushes MAY cancel them. It is RECOMMENDED that the receiving endpoint explicitly reject such requests (see Section 4.1.2) or pushes (see Section 7.2.3) in order to clean up transport state for the affected streams.
-Requests on Stream IDs less than the Stream ID in a GOAWAY frame from the server might have been processed; their status cannot be known until a response is received, the stream is reset individually, or the connection terminates. Servers MAY reject individual requests on streams below the indicated ID if these requests were not processed.
-Servers SHOULD send a GOAWAY frame when the closing of a connection is known in advance, even if the advance notice is small, so that the remote peer can know whether a request has been partially processed or not. For example, if an HTTP client sends a POST at the same time that a server closes a QUIC connection, the client cannot know if the server started to process that POST request if the server does not send a GOAWAY frame to indicate what streams it might have acted on.
-A client that is unable to retry requests loses all requests that are in flight when the server closes the connection. An endpoint MAY send multiple GOAWAY frames indicating different identifiers, but MUST NOT increase the identifier value they send, in particular since clients might already have retried unprocessed requests on another connection. A server that is attempting to gracefully shut down a connection SHOULD send an initial GOAWAY frame with the last Stream ID set to the maximum value allowed by QUIC’s MAX_STREAMS and SHOULD NOT increase the MAX_STREAMS limit thereafter. This signals to the client that a shutdown is imminent and that initiating further requests is prohibited. After allowing time for any in-flight requests (at least one round-trip time), the server MAY send another GOAWAY frame with an updated last Stream ID. This ensures that a connection can be cleanly shut down without losing requests.
-A client has more flexibility in the value it chooses for the Push ID in a GOAWAY that it sends. A value of 2^62 - 1 indicates that the server can continue initiating pushes to complete outstanding requests, and the client can continue granting push credit as needed (see Section 7.2.7). A smaller value indicates the client will reject pushes with Push IDs greater than or equal to this value. Like the server, the client MAY send subsequent GOAWAY frames so long as the specified Push ID is strictly smaller than all previously sent values.
-Once all accepted requests and pushes have been processed, the endpoint can permit the connection to become idle, or MAY initiate an immediate closure of the connection. An endpoint that completes a graceful shutdown SHOULD use the HTTP_NO_ERROR code when closing the connection.
-If a client has consumed all available bidirectional stream IDs with requests, the server need not send a GOAWAY frame, since the client is unable to make further requests.
-An HTTP/3 implementation can immediately close the QUIC connection at any time. This results in sending a QUIC CONNECTION_CLOSE frame to the peer; the error code in this frame indicates to the peer why the connection is being closed. See Section 8 for error codes which can be used when closing a connection.
-Before closing the connection, a GOAWAY MAY be sent to allow the client to retry some requests. Including the GOAWAY frame in the same packet as the QUIC CONNECTION_CLOSE frame improves the chances of the frame being received by clients.
-For various reasons, the QUIC transport could indicate to the application layer that the connection has terminated. This might be due to an explicit closure by the peer, a transport-level error, or a change in network topology which interrupts connectivity.
-If a connection terminates without a GOAWAY frame, clients MUST assume that any request which was sent, whether in whole or in part, might have been processed.
-A QUIC stream provides reliable in-order delivery of bytes, but makes no guarantees about order of delivery with regard to bytes on other streams. On the wire, data is framed into QUIC STREAM frames, but this framing is invisible to the HTTP framing layer. The transport layer buffers and orders received QUIC STREAM frames, exposing the data contained within as a reliable byte stream to the application. Although QUIC permits out-of-order delivery within a stream, HTTP/3 does not make use of this feature.
-QUIC streams can be either unidirectional, carrying data only from initiator to receiver, or bidirectional. Streams can be initiated by either the client or the server. For more detail on QUIC streams, see Section 2 of [QUIC-TRANSPORT].
-When HTTP headers and data are sent over QUIC, the QUIC layer handles most of the stream management. HTTP does not need to do any separate multiplexing when using QUIC - data sent over a QUIC stream always maps to a particular HTTP transaction or connection context.
-All client-initiated bidirectional streams are used for HTTP requests and responses. A bidirectional stream ensures that the response can be readily correlated with the request. This means that the client’s first request occurs on QUIC stream 0, with subsequent requests on stream 4, 8, and so on. In order to permit these streams to open, an HTTP/3 server SHOULD configure non-zero minimum values for the number of permitted streams and the initial stream flow control window. It is RECOMMENDED that at least 100 requests be permitted at a time, so as to not unnecessarily limit parallelism.
-HTTP/3 does not use server-initiated bidirectional streams, though an extension could define a use for these streams. Clients MUST treat receipt of a server-initiated bidirectional stream as a connection error of type HTTP_STREAM_CREATION_ERROR unless such an extension has been negotiated.
-Unidirectional streams, in either direction, are used for a range of purposes. The purpose is indicated by a stream type, which is sent as a variable-length integer at the start of the stream. The format and structure of data that follows this integer is determined by the stream type.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream Type (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 1: Unidirectional Stream Header
-Some stream types are reserved (Section 6.2.3). Two stream types are defined in this document: control streams (Section 6.2.1) and push streams (Section 6.2.2). [QPACK] defines two additional stream types. Other stream types can be defined by extensions to HTTP/3; see Section 9 for more details.
-The performance of HTTP/3 connections in the early phase of their lifetime is sensitive to the creation and exchange of data on unidirectional streams. Endpoints that excessively restrict the number of streams or the flow control window of these streams will increase the chance that the remote peer reaches the limit early and becomes blocked. In particular, implementations should consider that remote peers may wish to exercise reserved stream behavior (Section 6.2.3) with some of the unidirectional streams they are permitted to use. To avoid blocking, the transport parameters sent by both clients and servers MUST allow the peer to create at least one unidirectional stream for the HTTP control stream plus the number of unidirectional streams required by mandatory extensions (three being the minimum number required for the base HTTP/3 protocol and QPACK), and SHOULD provide at least 1,024 bytes of flow control credit to each stream.
-Note that an endpoint is not required to grant additional credits to create more unidirectional streams if its peer consumes all the initial credits before creating the critical unidirectional streams. Endpoints SHOULD create the HTTP control stream as well as the unidirectional streams required by mandatory extensions (such as the QPACK encoder and decoder streams) first, and then create additional streams as allowed by their peer.
-If the stream header indicates a stream type which is not supported by the recipient, the remainder of the stream cannot be consumed as the semantics are unknown. Recipients of unknown stream types MAY abort reading of the stream with an error code of HTTP_STREAM_CREATION_ERROR, but MUST NOT consider such streams to be a connection error of any kind.
-Implementations MAY send stream types before knowing whether the peer supports them. However, stream types which could modify the state or semantics of existing protocol components, including QPACK or other extensions, MUST NOT be sent until the peer is known to support them.
-A sender can close or reset a unidirectional stream unless otherwise specified. A receiver MUST tolerate unidirectional streams being closed or reset prior to the reception of the unidirectional stream header.
-A control stream is indicated by a stream type of 0x00. Data on this stream consists of HTTP/3 frames, as defined in Section 7.2.
-Each side MUST initiate a single control stream at the beginning of the connection and send its SETTINGS frame as the first frame on this stream. If the first frame of the control stream is any other frame type, this MUST be treated as a connection error of type HTTP_MISSING_SETTINGS. Only one control stream per peer is permitted; receipt of a second stream which claims to be a control stream MUST be treated as a connection error of type HTTP_STREAM_CREATION_ERROR. The sender MUST NOT close the control stream, and the receiver MUST NOT request that the sender close the control stream. If either control stream is closed at any point, this MUST be treated as a connection error of type HTTP_CLOSED_CRITICAL_STREAM.
-A pair of unidirectional streams is used rather than a single bidirectional stream. This allows either peer to send data as soon as it is able. Depending on whether 0-RTT is enabled on the connection, either client or server might be able to send stream data first after the cryptographic handshake completes.
-Server push is an optional feature introduced in HTTP/2 that allows a server to initiate a response before a request has been made. See Section 4.4 for more details.
-A push stream is indicated by a stream type of 0x01, followed by the Push ID of the promise that it fulfills, encoded as a variable-length integer. The remaining data on this stream consists of HTTP/3 frames, as defined in Section 7.2, and fulfills a promised server push by zero or more non-final HTTP responses followed by a single final HTTP response, as defined in Section 4.1. Server push and Push IDs are described in Section 4.4.
-Only servers can push; if a server receives a client-initiated push stream, this MUST be treated as a connection error of type HTTP_STREAM_CREATION_ERROR.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| 0x01 (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 2: Push Stream Header
-Each Push ID MUST only be used once in a push stream header. If a push stream header includes a Push ID that was used in another push stream header, the client MUST treat this as a connection error of type HTTP_ID_ERROR.
-Stream types of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown types be ignored. These streams have no semantics, and can be sent when application-layer padding is desired. They MAY also be sent on connections where no data is currently being transferred. Endpoints MUST NOT consider these streams to have any meaning upon receipt.
-The payload and length of the stream are selected in any manner the implementation chooses.
-HTTP frames are carried on QUIC streams, as described in Section 6. HTTP/3 defines three stream types: control stream, request stream, and push stream. This section describes HTTP/3 frame formats and the streams types on which they are permitted; see Table 1 for an overview. A comparison between HTTP/2 and HTTP/3 frames is provided in Appendix A.2.
- - -| Frame | -Control Stream | -Request Stream | -Push Stream | -Section | -
|---|---|---|---|---|
| DATA | -No | -Yes | -Yes | -Section 7.2.1 | -
| HEADERS | -No | -Yes | -Yes | -Section 7.2.2 | -
| CANCEL_PUSH | -Yes | -No | -No | -Section 7.2.3 | -
| SETTINGS | -Yes (1) | -No | -No | -Section 7.2.4 | -
| PUSH_PROMISE | -No | -Yes | -No | -Section 7.2.5 | -
| GOAWAY | -Yes | -No | -No | -Section 7.2.6 | -
| MAX_PUSH_ID | -Yes | -No | -No | -Section 7.2.7 | -
| DUPLICATE_PUSH | -No | -Yes | -No | -Section 7.2.8 | -
| Reserved | -Yes | -Yes | -Yes | -Section 7.2.9 | -
Certain frames can only occur as the first frame of a particular stream type; these are indicated in Table 1 with a (1). Specific guidance is provided in the relevant section.
-Note that, unlike QUIC frames, HTTP/3 frames can span multiple packets.
-All frames have the following format:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Type (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Frame Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 3: HTTP/3 frame format
-A frame includes the following fields:
- - -Each frame’s payload MUST contain exactly the fields identified in its description. A frame payload that contains additional bytes after the identified fields or a frame payload that terminates before the end of the identified fields MUST be treated as a connection error of type HTTP_FRAME_ERROR.
-When a stream terminates cleanly, if the last frame on the stream was truncated, this MUST be treated as a connection error (Section 8) of type HTTP_FRAME_ERROR. Streams which terminate abruptly may be reset at any point in a frame.
-DATA frames (type=0x0) convey arbitrary, variable-length sequences of bytes associated with an HTTP request or response payload.
-DATA frames MUST be associated with an HTTP request or response. If a DATA frame is received on a control stream, the recipient MUST respond with a connection error (Section 8) of type HTTP_FRAME_UNEXPECTED.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 4: DATA frame payload
-The HEADERS frame (type=0x1) is used to carry a header block, compressed using QPACK. See [QPACK] for more details.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Header Block (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 5: HEADERS frame payload
-HEADERS frames can only be sent on request / push streams. If a HEADERS frame is received on a control stream, the recipient MUST respond with a connection error (Section 8) of type HTTP_FRAME_UNEXPECTED.
-The CANCEL_PUSH frame (type=0x3) is used to request cancellation of a server push prior to the push stream being received. The CANCEL_PUSH frame identifies a server push by Push ID (see Section 7.2.5), encoded as a variable-length integer.
-When a client sends CANCEL_PUSH, it is indicating that it does not wish to receive the promised resource. The server SHOULD abort sending the resource, but the mechanism to do so depends on the state of the corresponding push stream. If the server has not yet created a push stream, it does not create one. If the push stream is open, the server SHOULD abruptly terminate that stream. If the push stream has already ended, the server MAY still abruptly terminate the stream or MAY take no action.
-When a server sends CANCEL_PUSH, it is indicating that it will not be fulfilling a promise and has not created a push stream. The client should not expect the corresponding promise to be fulfilled.
-Sending CANCEL_PUSH has no direct effect on the state of existing push streams. A server SHOULD NOT send a CANCEL_PUSH when it has already created a corresponding push stream, and a client SHOULD NOT send a CANCEL_PUSH when it has already received a corresponding push stream.
-A CANCEL_PUSH frame is sent on the control stream. Receiving a CANCEL_PUSH frame on a stream other than the control stream MUST be treated as a connection error of type HTTP_FRAME_UNEXPECTED.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 6: CANCEL_PUSH frame payload
-The CANCEL_PUSH frame carries a Push ID encoded as a variable-length integer. The Push ID identifies the server push that is being cancelled (see Section 7.2.5). If a CANCEL_PUSH frame is received which references a Push ID greater than currently allowed on the connection, this MUST be treated as a connection error of type HTTP_ID_ERROR.
-If the client receives a CANCEL_PUSH frame, that frame might identify a Push ID that has not yet been mentioned by a PUSH_PROMISE frame due to reordering. If a server receives a CANCEL_PUSH frame for a Push ID that has not yet been mentioned by a PUSH_PROMISE frame, this MUST be treated as a connection error of type HTTP_ID_ERROR.
-The SETTINGS frame (type=0x4) conveys configuration parameters that affect how endpoints communicate, such as preferences and constraints on peer behavior. Individually, a SETTINGS parameter can also be referred to as a “setting”; the identifier and value of each setting parameter can be referred to as a “setting identifier” and a “setting value”.
-SETTINGS frames always apply to a connection, never a single stream. A SETTINGS frame MUST be sent as the first frame of each control stream (see Section 6.2.1) by each peer, and MUST NOT be sent subsequently. If an endpoint receives a second SETTINGS frame on the control stream, the endpoint MUST respond with a connection error of type HTTP_FRAME_UNEXPECTED.
-SETTINGS frames MUST NOT be sent on any stream other than the control stream. If an endpoint receives a SETTINGS frame on a different stream, the endpoint MUST respond with a connection error of type HTTP_FRAME_UNEXPECTED.
-SETTINGS parameters are not negotiated; they describe characteristics of the sending peer, which can be used by the receiving peer. However, a negotiation can be implied by the use of SETTINGS - each peer uses SETTINGS to advertise a set of supported values. The definition of the setting would describe how each peer combines the two sets to conclude which choice will be used. SETTINGS does not provide a mechanism to identify when the choice takes effect.
-Different values for the same parameter can be advertised by each peer. For example, a client might be willing to consume a very large response header, while servers are more cautious about request size.
-The same setting identifier MUST NOT occur more than once in the SETTINGS frame. A receiver MAY treat the presence of duplicate setting identifiers as a connection error of type HTTP_SETTINGS_ERROR.
-The payload of a SETTINGS frame consists of zero or more parameters. Each parameter consists of a setting identifier and a value, both encoded as QUIC variable-length integers.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Identifier (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Value (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 7: SETTINGS parameter format
-An implementation MUST ignore the contents for any SETTINGS identifier it does not understand.
-The following settings are defined in HTTP/3:
- - -Setting identifiers of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown identifiers be ignored. Such settings have no defined meaning. Endpoints SHOULD include at least one such setting in their SETTINGS frame. Endpoints MUST NOT consider such settings to have any meaning upon receipt.
-Because the setting has no defined meaning, the value of the setting can be any value the implementation selects.
-Additional settings can be defined by extensions to HTTP/3; see Section 9 for more details.
-An HTTP implementation MUST NOT send frames or requests which would be invalid based on its current understanding of the peer’s settings.
-All settings begin at an initial value. Each endpoint SHOULD use these initial values to send messages before the peer’s SETTINGS frame has arrived, as packets carrying the settings can be lost or delayed. When the SETTINGS frame arrives, any settings are changed to their new values.
-This removes the need to wait for the SETTINGS frame before sending messages. Endpoints MUST NOT require any data to be received from the peer prior to sending the SETTINGS frame; settings MUST be sent as soon as the transport is ready to send data.
-For servers, the initial value of each client setting is the default value.
-For clients using a 1-RTT QUIC connection, the initial value of each server setting is the default value. 1-RTT keys will always become available prior to SETTINGS arriving, even if the server sends SETTINGS immediately. Clients SHOULD NOT wait indefinitely for SETTINGS to arrive before sending requests, but SHOULD process received datagrams in order to increase the likelihood of processing SETTINGS before sending the first request.
-When a 0-RTT QUIC connection is being used, the initial value of each server setting is the value used in the previous session. Clients SHOULD store the settings the server provided in the connection where resumption information was provided, but MAY opt not to store settings in certain cases (e.g., if the session ticket is received before the SETTINGS frame). A client MUST comply with stored settings – or default values, if no values are stored – when attempting 0-RTT. Once a server has provided new settings, clients MUST comply with those values.
-A server can remember the settings that it advertised, or store an integrity-protected copy of the values in the ticket and recover the information when accepting 0-RTT data. A server uses the HTTP/3 settings values in determining whether to accept 0-RTT data. If the server cannot determine that the settings remembered by a client are compatible with its current settings, it MUST NOT accept 0-RTT data. Remembered settings are compatible if a client complying with those settings would not violate the server’s current settings.
-A server MAY accept 0-RTT and subsequently provide different settings in its SETTINGS frame. If 0-RTT data is accepted by the server, its SETTINGS frame MUST NOT reduce any limits or alter any values that might be violated by the client with its 0-RTT data. The server MUST include all settings which differ from their default values. If a server accepts 0-RTT, but then sends a SETTINGS frame which reduces a setting the client understands or omits a value that was previously specified to have a non-default value, this MUST be treated as a connection error of type HTTP_SETTINGS_ERROR.
-The PUSH_PROMISE frame (type=0x5) is used to carry a promised request header set from server to client on a request stream, as in HTTP/2.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Header Block (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 8: PUSH_PROMISE frame payload
-The payload consists of:
- - -A server MUST NOT use a Push ID that is larger than the client has provided in a MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat receipt of a PUSH_PROMISE frame that contains a larger Push ID than the client has advertised as a connection error of HTTP_ID_ERROR.
-A server MUST NOT use the same Push ID in multiple PUSH_PROMISE frames. A client MUST treat receipt of a Push ID which has already been promised as a connection error of type HTTP_ID_ERROR.
-If a PUSH_PROMISE frame is received on the control stream, the client MUST respond with a connection error (Section 8) of type HTTP_FRAME_UNEXPECTED.
-A client MUST NOT send a PUSH_PROMISE frame. A server MUST treat the receipt of a PUSH_PROMISE frame as a connection error of type HTTP_FRAME_UNEXPECTED.
-See Section 4.4 for a description of the overall server push mechanism.
-The GOAWAY frame (type=0x7) is used to initiate graceful shutdown of a connection by either endpoint. GOAWAY allows an endpoint to stop accepting new requests or pushes while still finishing processing of previously received requests and pushes. This enables administrative actions, like server maintenance. GOAWAY by itself does not close a connection.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID/Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 9: GOAWAY frame payload
-The GOAWAY frame is always sent on the control stream. In the server to client direction, it carries a QUIC Stream ID for a client-initiated bidirectional stream encoded as a variable-length integer. A client MUST treat receipt of a GOAWAY frame containing a Stream ID of any other type as a connection error of type HTTP_ID_ERROR.
-In the client to server direction, the GOAWAY frame carries a Push ID encoded as a variable-length integer.
-The GOAWAY frame applies to the connection, not a specific stream. A client MUST treat a GOAWAY frame on a stream other than the control stream as a connection error (Section 8) of type HTTP_FRAME_UNEXPECTED.
-See Section 5.2 for more information on the use of the GOAWAY frame.
-The MAX_PUSH_ID frame (type=0xD) is used by clients to control the number of server pushes that the server can initiate. This sets the maximum value for a Push ID that the server can use in PUSH_PROMISE and CANCEL_PUSH frames. Consequently, this also limits the number of push streams that the server can initiate in addition to the limit maintained by the QUIC transport.
-The MAX_PUSH_ID frame is always sent on the control stream. Receipt of a MAX_PUSH_ID frame on any other stream MUST be treated as a connection error of type HTTP_FRAME_UNEXPECTED.
-A server MUST NOT send a MAX_PUSH_ID frame. A client MUST treat the receipt of a MAX_PUSH_ID frame as a connection error of type HTTP_FRAME_UNEXPECTED.
-The maximum Push ID is unset when a connection is created, meaning that a server cannot push until it receives a MAX_PUSH_ID frame. A client that wishes to manage the number of promised server pushes can increase the maximum Push ID by sending MAX_PUSH_ID frames as the server fulfills or cancels server pushes.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 10: MAX_PUSH_ID frame payload
-The MAX_PUSH_ID frame carries a single variable-length integer that identifies the maximum value for a Push ID that the server can use (see Section 7.2.5). A MAX_PUSH_ID frame cannot reduce the maximum Push ID; receipt of a MAX_PUSH_ID that contains a smaller value than previously received MUST be treated as a connection error of type HTTP_ID_ERROR.
-The DUPLICATE_PUSH frame (type=0xE) is used by servers to indicate that an existing pushed resource is related to multiple client requests.
-The DUPLICATE_PUSH frame is always sent on a request stream. Receipt of a DUPLICATE_PUSH frame on any other stream MUST be treated as a connection error of type HTTP_FRAME_UNEXPECTED.
-A client MUST NOT send a DUPLICATE_PUSH frame. A server MUST treat the receipt of a DUPLICATE_PUSH frame as a connection error of type HTTP_FRAME_UNEXPECTED.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Push ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 11: DUPLICATE_PUSH frame payload
-The DUPLICATE_PUSH frame carries a single variable-length integer that identifies the Push ID of a resource that the server has previously promised (see Section 7.2.5), though that promise might not be received before this frame. A server MUST NOT use a Push ID that is larger than the client has provided in a MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat receipt of a DUPLICATE_PUSH that contains a larger Push ID than the client has advertised as a connection error of type HTTP_ID_ERROR.
-This frame allows the server to use the same server push in response to multiple concurrent requests. Referencing the same server push ensures that a promise can be made in relation to every response in which server push might be needed without duplicating request headers or pushed responses.
-Allowing duplicate references to the same Push ID is primarily to reduce duplication caused by concurrent requests. A server SHOULD avoid reusing a Push ID over a long period. Clients are likely to consume server push responses and not retain them for reuse over time. Clients that see a DUPLICATE_PUSH that uses a Push ID that they have since consumed and discarded are forced to ignore the DUPLICATE_PUSH.
-Frame types of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown types be ignored (Section 9). These frames have no semantics, and can be sent on any open stream when application-layer padding is desired. They MAY also be sent on connections where no data is currently being transferred. Endpoints MUST NOT consider these frames to have any meaning upon receipt.
-The payload and length of the frames are selected in any manner the implementation chooses.
-Frame types which were used in HTTP/2 where there is no corresponding HTTP/3 frame have also been reserved (Section 11.2). These frame types MUST NOT be sent, and receipt MAY be treated as an error of type HTTP_FRAME_UNEXPECTED.
-QUIC allows the application to abruptly terminate (reset) individual streams or the entire connection when an error is encountered. These are referred to as “stream errors” or “connection errors” and are described in more detail in [QUIC-TRANSPORT]. An endpoint MAY choose to treat a stream error as a connection error.
-Because new error codes can be defined without negotiation (see Section 9), receipt of an unknown error code or use of an error code in an unexpected context MUST NOT be treated as an error. However, closing a stream can constitute an error regardless of the error code (see Section 4.1).
-This section describes HTTP/3-specific error codes which can be used to express the cause of a connection or stream error.
-The following error codes are defined for use when abruptly terminating streams, aborting reading of streams, or immediately closing connections.
- - -HTTP/3 permits extension of the protocol. Within the limitations described in this section, protocol extensions can be used to provide additional services or alter any aspect of the protocol. Extensions are effective only within the scope of a single HTTP/3 connection.
-This applies to the protocol elements defined in this document. This does not affect the existing options for extending HTTP, such as defining new methods, status codes, or header fields.
-Extensions are permitted to use new frame types (Section 7.2), new settings (Section 7.2.4.1), new error codes (Section 8), or new unidirectional stream types (Section 6.2). Registries are established for managing these extension points: frame types (Section 11.2), settings (Section 11.3), error codes (Section 11.4), and stream types (Section 11.5).
-Implementations MUST ignore unknown or unsupported values in all extensible protocol elements. Implementations MUST discard frames and unidirectional streams that have unknown or unsupported types. This means that any of these extension points can be safely used by extensions without prior arrangement or negotiation. However, where a known frame type is required to be in a specific location, such as the SETTINGS frame as the first frame of the control stream (see Section 6.2.1), an unknown frame type does not satisfy that requirement and SHOULD be treated as an error.
-Extensions that could change the semantics of existing protocol components MUST be negotiated before being used. For example, an extension that changes the layout of the HEADERS frame cannot be used until the peer has given a positive signal that this is acceptable. In this case, it could also be necessary to coordinate when the revised layout comes into effect.
-This document doesn’t mandate a specific method for negotiating the use of an extension but notes that a setting (Section 7.2.4.1) could be used for that purpose. If both peers set a value that indicates willingness to use the extension, then the extension can be used. If a setting is used for extension negotiation, the default value MUST be defined in such a fashion that the extension is disabled if the setting is omitted.
-The security considerations of HTTP/3 should be comparable to those of HTTP/2 with TLS; the considerations from Section 10 of [HTTP2] apply in addition to those listed here.
-When HTTP Alternative Services is used for discovery for HTTP/3 endpoints, the security considerations of [ALTSVC] also apply.
-Where HTTP/2 employs PADDING frames and Padding fields in other frames to make a connection more resistant to traffic analysis, HTTP/3 can either rely on transport-layer padding or employ the reserved frame and stream types discussed in Section 7.2.9 and Section 6.2.3. These methods of padding produce different results in terms of the granularity of padding, the effect of packet loss and recovery, and how an implementation might control padding.
-Several protocol elements contain nested length elements, typically in the form of frames with an explicit length containing variable-length integers. This could pose a security risk to an incautious implementer. An implementation MUST ensure that the length of a frame exactly matches the length of the fields it contains.
-The use of 0-RTT with HTTP/3 creates an exposure to replay attack. The anti-replay mitigations in [HTTP-REPLAY] MUST be applied when using HTTP/3 with 0-RTT.
-Certain HTTP implementations use the client address for logging or access-control purposes. Since a QUIC client’s address might change during a connection (and future versions might support simultaneous use of multiple addresses), such implementations will need to either actively retrieve the client’s current address or addresses when they are relevant or explicitly accept that the original address might change.
-This document creates a new registration for the identification of HTTP/3 in the “Application Layer Protocol Negotiation (ALPN) Protocol IDs” registry established in [RFC7301].
-The “h3” string identifies HTTP/3:
- - -This document establishes a registry for HTTP/3 frame type codes. The “HTTP/3 Frame Type” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-While this registry is separate from the “HTTP/2 Frame Type” registry defined in [HTTP2], it is preferable that the assignments parallel each other where the code spaces overlap. If an entry is present in only one registry, every effort SHOULD be made to avoid assigning the corresponding value to an unrelated operation.
-New entries in this registry require the following information:
- - -The entries in the following table are registered by this document.
-| Frame Type | -Code | -Specification | -
|---|---|---|
| DATA | -0x0 | -Section 7.2.1 | -
| HEADERS | -0x1 | -Section 7.2.2 | -
| Reserved | -0x2 | -N/A | -
| CANCEL_PUSH | -0x3 | -Section 7.2.3 | -
| SETTINGS | -0x4 | -Section 7.2.4 | -
| PUSH_PROMISE | -0x5 | -Section 7.2.5 | -
| Reserved | -0x6 | -N/A | -
| GOAWAY | -0x7 | -Section 7.2.6 | -
| Reserved | -0x8 | -N/A | -
| Reserved | -0x9 | -N/A | -
| MAX_PUSH_ID | -0xD | -Section 7.2.7 | -
| DUPLICATE_PUSH | -0xE | -Section 7.2.8 | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-This document establishes a registry for HTTP/3 settings. The “HTTP/3 Settings” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126]. The designated experts are the same as those for the “HTTP/2 Settings” registry defined in [HTTP2].
-While this registry is separate from the “HTTP/2 Settings” registry defined in [HTTP2], it is preferable that the assignments parallel each other. If an entry is present in only one registry, every effort SHOULD be made to avoid assigning the corresponding value to an unrelated operation.
-New registrations are advised to provide the following information:
- - -The entries in the following table are registered by this document.
-| Setting Name | -Code | -Specification | -Default | -
|---|---|---|---|
| Reserved | -0x2 | -N/A | -N/A | -
| Reserved | -0x3 | -N/A | -N/A | -
| Reserved | -0x4 | -N/A | -N/A | -
| Reserved | -0x5 | -N/A | -N/A | -
| MAX_HEADER_LIST_SIZE | -0x6 | -Section 7.2.4.1 | -Unlimited | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-This document establishes a registry for HTTP/3 error codes. The “HTTP/3 Error Code” registry manages a 62-bit space. The “HTTP/3 Error Code” registry operates under the “Expert Review” policy [RFC8126].
-Registrations for error codes are required to include a description of the error code. An expert reviewer is advised to examine new registrations for possible duplication with existing error codes. Use of existing registrations is to be encouraged, but not mandated.
-New registrations are advised to provide the following information:
- - -The entries in the following table are registered by this document.
-| Name | -Code | -Description | -Specification | -
|---|---|---|---|
| HTTP_NO_ERROR | -0x0100 | -No error | -Section 8.1 | -
| HTTP_GENERAL_PROTOCOL_ERROR | -0x0101 | -General protocol error | -Section 8.1 | -
| HTTP_INTERNAL_ERROR | -0x0102 | -Internal error | -Section 8.1 | -
| HTTP_STREAM_CREATION_ERROR | -0x0103 | -Stream creation error | -Section 8.1 | -
| HTTP_CLOSED_CRITICAL_STREAM | -0x0104 | -Critical stream was closed | -Section 8.1 | -
| HTTP_FRAME_UNEXPECTED | -0x0105 | -Frame not permitted in the current state | -Section 8.1 | -
| HTTP_FRAME_ERROR | -0x0106 | -Frame violated layout or size rules | -Section 8.1 | -
| HTTP_EXCESSIVE_LOAD | -0x0107 | -Peer generating excessive load | -Section 8.1 | -
| HTTP_ID_ERROR | -0x0108 | -An identifier was used incorrectly | -Section 8.1 | -
| HTTP_SETTINGS_ERROR | -0x0109 | -SETTINGS frame contained invalid values | -Section 8.1 | -
| HTTP_MISSING_SETTINGS | -0x010A | -No SETTINGS frame received | -Section 8.1 | -
| HTTP_REQUEST_REJECTED | -0x010B | -Request not processed | -Section 8.1 | -
| HTTP_REQUEST_CANCELLED | -0x010C | -Data no longer needed | -Section 8.1 | -
| HTTP_REQUEST_INCOMPLETE | -0x010D | -Stream terminated early | -Section 8.1 | -
| HTTP_EARLY_RESPONSE | -0x010E | -Remainder of request not needed | -Section 8.1 | -
| HTTP_CONNECT_ERROR | -0x010F | -TCP reset or error on CONNECT request | -Section 8.1 | -
| HTTP_VERSION_FALLBACK | -0x0110 | -Retry over HTTP/1.1 | -Section 8.1 | -
This document establishes a registry for HTTP/3 unidirectional stream types. The “HTTP/3 Stream Type” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-New entries in this registry require the following information:
- - -The entries in the following table are registered by this document.
-| Stream Type | -Code | -Specification | -Sender | -
|---|---|---|---|
| Control Stream | -0x00 | -Section 6.2.1 | -Both | -
| Push Stream | -0x01 | -Section 4.4 | -Server | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-| [ALTSVC] | --Nottingham, M., McManus, P. and J. Reschke, "HTTP Alternative Services", RFC 7838, DOI 10.17487/RFC7838, April 2016. | -
| [HTTP-REPLAY] | --Thomson, M., Nottingham, M. and W. Tarreau, "Using Early Data in HTTP", RFC 8470, DOI 10.17487/RFC8470, September 2018. | -
| [HTTP2] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| [QPACK] | --Krasic, C., Bishop, M. and A. Frindell, "QPACK: Header Compression for HTTP over QUIC", Internet-Draft draft-ietf-quic-qpack, October 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, October 2019. | -
| [RFC0793] | --Postel, J., "Transmission Control Protocol", STD 7, RFC 793, DOI 10.17487/RFC0793, September 1981. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC5234] | --Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008. | -
| [RFC6066] | --Eastlake 3rd, D., "Transport Layer Security (TLS) Extensions: Extension Definitions", RFC 6066, DOI 10.17487/RFC6066, January 2011. | -
| [RFC6265] | --Barth, A., "HTTP State Management Mechanism", RFC 6265, DOI 10.17487/RFC6265, April 2011. | -
| [RFC7230] | --Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, DOI 10.17487/RFC7230, June 2014. | -
| [RFC7231] | --Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI 10.17487/RFC7231, June 2014. | -
| [RFC7838] | --Nottingham, M., McManus, P. and J. Reschke, "HTTP Alternative Services", RFC 7838, DOI 10.17487/RFC7838, April 2016. | -
| [RFC8126] | --Cotton, M., Leiba, B. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 8126, DOI 10.17487/RFC8126, June 2017. | -
| [RFC8164] | --Nottingham, M. and M. Thomson, "Opportunistic Security for HTTP/2", RFC 8164, DOI 10.17487/RFC8164, May 2017. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [HPACK] | --Peon, R. and H. Ruellan, "HPACK: Header Compression for HTTP/2", RFC 7541, DOI 10.17487/RFC7541, May 2015. | -
| [RFC6585] | --Nottingham, M. and R. Fielding, "Additional HTTP Status Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC7413] | --Cheng, Y., Chu, J., Radhakrishnan, S. and A. Jain, "TCP Fast Open", RFC 7413, DOI 10.17487/RFC7413, December 2014. | -
HTTP/3 is strongly informed by HTTP/2, and bears many similarities. This section describes the approach taken to design HTTP/3, points out important differences from HTTP/2, and describes how to map HTTP/2 extensions into HTTP/3.
-HTTP/3 begins from the premise that similarity to HTTP/2 is preferable, but not a hard requirement. HTTP/3 departs from HTTP/2 where QUIC differs from TCP, either to take advantage of QUIC features (like streams) or to accommodate important shortcomings (such as a lack of total ordering). These differences make HTTP/3 similar to HTTP/2 in key aspects, such as the relationship of requests and responses to streams. However, the details of the HTTP/3 design are substantially different than HTTP/2.
-These departures are noted in this section.
-HTTP/3 permits use of a larger number of streams (2^62-1) than HTTP/2. The considerations about exhaustion of stream identifier space apply, though the space is significantly larger such that it is likely that other limits in QUIC are reached first, such as the limit on the connection flow control window.
-Many framing concepts from HTTP/2 can be elided on QUIC, because the transport deals with them. Because frames are already on a stream, they can omit the stream number. Because frames do not block multiplexing (QUIC’s multiplexing occurs below this layer), the support for variable-maximum-length packets can be removed. Because stream termination is handled by QUIC, an END_STREAM flag is not required. This permits the removal of the Flags field from the generic frame layout.
-Frame payloads are largely drawn from [HTTP2]. However, QUIC includes many features (e.g., flow control) which are also present in HTTP/2. In these cases, the HTTP mapping does not re-implement them. As a result, several HTTP/2 frame types are not required in HTTP/3. Where an HTTP/2-defined frame is no longer used, the frame ID has been reserved in order to maximize portability between HTTP/2 and HTTP/3 implementations. However, even equivalent frames between the two mappings are not identical.
-Many of the differences arise from the fact that HTTP/2 provides an absolute ordering between frames across all streams, while QUIC provides this guarantee on each stream only. As a result, if a frame type makes assumptions that frames from different streams will still be received in the order sent, HTTP/3 will break them.
-Some examples of feature adaptations are described below, as well as general guidance to extension frame implementors converting an HTTP/2 extension to HTTP/3.
-HTTP/2 specifies priority assignments in PRIORITY frames and (optionally) in HEADERS frames. HTTP/3 does not provide a means of signaling priority.
-Note that while there is no explicit signaling for priority, this does not mean that prioritization is not important for achieving good performance.
-HPACK was designed with the assumption of in-order delivery. A sequence of encoded header blocks must arrive (and be decoded) at an endpoint in the same order in which they were encoded. This ensures that the dynamic state at the two endpoints remains in sync.
-Because this total ordering is not provided by QUIC, HTTP/3 uses a modified version of HPACK, called QPACK. QPACK uses a single unidirectional stream to make all modifications to the dynamic table, ensuring a total order of updates. All frames which contain encoded headers merely reference the table state at a given time without modifying it.
-[QPACK] provides additional details.
-Frame type definitions in HTTP/3 often use the QUIC variable-length integer encoding. In particular, Stream IDs use this encoding, which allows for a larger range of possible values than the encoding used in HTTP/2. Some frames in HTTP/3 use an identifier rather than a Stream ID (e.g., Push IDs). Redefinition of the encoding of extension frame types might be necessary if the encoding includes a Stream ID.
-Because the Flags field is not present in generic HTTP/3 frames, those frames which depend on the presence of flags need to allocate space for flags as part of their frame payload.
-Other than this issue, frame type HTTP/2 extensions are typically portable to QUIC simply by replacing Stream 0 in HTTP/2 with a control stream in HTTP/3. HTTP/3 extensions will not assume ordering, but would not be harmed by ordering, and would be portable to HTTP/2 in the same manner.
-Frame types defined by extensions to HTTP/2 need to be separately registered for HTTP/3 if still applicable. The IDs of frames defined in [HTTP2] have been reserved for simplicity. Note that the frame type space in HTTP/3 is substantially larger (62 bits versus 8 bits), so many HTTP/3 frame types have no equivalent HTTP/2 code points. See Section 11.2.
-An important difference from HTTP/2 is that settings are sent once, as the first frame of the control stream, and thereafter cannot change. This eliminates many corner cases around synchronization of changes.
-Some transport-level options that HTTP/2 specifies via the SETTINGS frame are superseded by QUIC transport parameters in HTTP/3. The HTTP-level options that are retained in HTTP/3 have the same value as in HTTP/2.
-Below is a listing of how each HTTP/2 SETTINGS parameter is mapped:
- - -In HTTP/3, setting values are variable-length integers (6, 14, 30, or 62 bits long) rather than fixed-length 32-bit fields as in HTTP/2. This will often produce a shorter encoding, but can produce a longer encoding for settings which use the full 32-bit space. Settings ported from HTTP/2 might choose to redefine the format of their settings to avoid using the 62-bit encoding.
-Settings need to be defined separately for HTTP/2 and HTTP/3. The IDs of settings defined in [HTTP2] have been reserved for simplicity. Note that the settings identifier space in HTTP/3 is substantially larger (62 bits versus 16 bits), so many HTTP/3 settings have no equivalent HTTP/2 code point. See Section 11.3.
-As QUIC streams might arrive out-of-order, endpoints are advised to not wait for the peers’ settings to arrive before responding to other streams. See Section 7.2.4.2.
-QUIC has the same concepts of “stream” and “connection” errors that HTTP/2 provides. However, there is no direct portability of HTTP/2 error codes to HTTP/3 error codes; the values are shifted in order to prevent accidental or implicit conversion.
-The HTTP/2 error codes defined in Section 7 of [HTTP2] logically map to the HTTP/3 error codes as follows:
- - -Error codes need to be defined for HTTP/2 and HTTP/3 separately. See Section 11.4.
-Substantial editorial reorganization; no technical changes.
-None.
-The original authors of this specification were Robbie Shade and Mike Warres.
-A substantial portion of Mike’s contribution was supported by Microsoft during his employment there.
-| QUIC | -M. Thomson | -
| Internet-Draft | -Mozilla | -
| Intended status: Standards Track | -October 22, 2019 | -
| Expires: April 24, 2020 | -- |
Version-Independent Properties of QUIC
- draft-ietf-quic-invariants-latest
This document defines the properties of the QUIC transport protocol that are expected to remain unchanged over time as new versions of the protocol are developed.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-invariants.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -In addition to providing secure, multiplexed transport, QUIC [QUIC-TRANSPORT] includes the ability to negotiate a version. This allows the protocol to change over time in response to new requirements. Many characteristics of the protocol will change between versions.
-This document describes the subset of QUIC that is intended to remain stable as new versions are developed and deployed. All of these invariants are IP-version-independent.
-The primary goal of this document is to ensure that it is possible to deploy new versions of QUIC. By documenting the properties that can’t change, this document aims to preserve the ability to change any other aspect of the protocol. Thus, unless specifically described in this document, any aspect of the protocol can change between different versions.
-Appendix A is a non-exhaustive list of some incorrect assumptions that might be made based on knowledge of QUIC version 1; these do not apply to every version of QUIC.
-The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-This document uses terms and notational conventions from [QUIC-TRANSPORT].
-QUIC is a connection-oriented protocol between two endpoints. Those endpoints exchange UDP datagrams. These UDP datagrams contain QUIC packets. QUIC endpoints use QUIC packets to establish a QUIC connection, which is shared protocol state between those endpoints.
-A QUIC packet is the content of the UDP datagrams exchanged by QUIC endpoints. This document describes the contents of those datagrams.
-QUIC defines two types of packet header: long and short. Packets with long headers are identified by the most significant bit of the first byte being set; packets with a short header have that bit cleared.
-Aside from the values described here, the payload of QUIC packets is version-specific and of arbitrary length.
-Long headers take the form described in Figure 1. Bits that have version-specific semantics are marked with an X.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|1|X X X X X X X| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -|X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 1: QUIC Long Header
-A QUIC packet with a long header has the high bit of the first byte set to 1. All other bits in that byte are version specific.
-The next four bytes include a 32-bit Version field (see Section 4.4).
-The next byte contains the length in bytes of the Destination Connection ID (see Section 4.3) field that follows it. This length is encoded as an 8-bit unsigned integer. The Destination Connection ID field follows the DCID Len field and is between 0 and 255 bytes in length.
-The next byte contains the length in bytes of the Source Connection ID field that follows it. This length is encoded as a 8-bit unsigned integer. The Source Connection ID field follows the SCID Len field and is between 0 and 255 bytes in length.
-The remainder of the packet contains version-specific content.
-Short headers take the form described in Figure 2. Bits that have version-specific semantics are marked with an X.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|0|X X X X X X X| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -|X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 2: QUIC Short Header
-A QUIC packet with a short header has the high bit of the first byte set to 0.
-A QUIC packet with a short header includes a Destination Connection ID immediately following the first byte. The short header does not include the Connection ID Lengths, Source Connection ID, or Version fields. The length of the Destination Connection ID is not specified in packets with a short header and is not constrained by this specification.
-The remainder of the packet has version-specific semantics.
-A connection ID is an opaque field of arbitrary length.
-The primary function of a connection ID is to ensure that changes in addressing at lower protocol layers (UDP, IP, and below) don’t cause packets for a QUIC connection to be delivered to the wrong endpoint. The connection ID is used by endpoints and the intermediaries that support them to ensure that each QUIC packet can be delivered to the correct instance of an endpoint. At the endpoint, the connection ID is used to identify which QUIC connection the packet is intended for.
-The connection ID is chosen by each endpoint using version-specific methods. Packets for the same QUIC connection might use different connection ID values.
-QUIC versions are identified with a 32-bit integer, encoded in network byte order. Version 0 is reserved for version negotiation (see Section 5). All other version numbers are potentially valid.
-The properties described in this document apply to all versions of QUIC. A protocol that does not conform to the properties described in this document is not QUIC. Future documents might describe additional properties which apply to a specific QUIC version, or to a range of QUIC versions.
-A QUIC endpoint that receives a packet with a long header and a version it either does not understand or does not support might send a Version Negotiation packet in response. Packets with a short header do not trigger version negotiation.
-A Version Negotiation packet sets the high bit of the first byte, and thus it conforms with the format of a packet with a long header as defined in Section 4.1. A Version Negotiation packet is identifiable as such by the Version field, which is set to 0x00000000.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|1|X X X X X X X| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) = 0 | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Supported Version 1 (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Supported Version 2 (32)] | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Supported Version N (32)] | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 3: Version Negotiation Packet
-The Version Negotiation packet contains a list of Supported Version fields, each identifying a version that the endpoint sending the packet supports. The Supported Version fields follow the Version field. A Version Negotiation packet contains no other fields. An endpoint MUST ignore a packet that contains no Supported Version fields, or a truncated Supported Version.
-Version Negotiation packets do not use integrity or confidentiality protection. A specific QUIC version might authenticate the packet as part of its connection establishment process.
-An endpoint MUST include the value from the Source Connection ID field of the packet it receives in the Destination Connection ID field. The value for Source Connection ID MUST be copied from the Destination Connection ID of the received packet, which is initially randomly selected by a client. Echoing both connection IDs gives clients some assurance that the server received the packet and that the Version Negotiation packet was not generated by an off-path attacker.
-An endpoint that receives a Version Negotiation packet might change the version that it decides to use for subsequent packets. The conditions under which an endpoint changes QUIC version will depend on the version of QUIC that it chooses.
-See [QUIC-TRANSPORT] for a more thorough description of how an endpoint that supports QUIC version 1 generates and consumes a Version Negotiation packet.
-It is possible that middleboxes could use traits of a specific version of QUIC and assume that when other versions of QUIC exhibit similar traits the same underlying semantic is being expressed. There are potentially many such traits (see Appendix A). Some effort has been made to either eliminate or obscure some observable traits in QUIC version 1, but many of these remain. Other QUIC versions might make different design decisions and so exhibit different traits.
-The QUIC version number does not appear in all QUIC packets, which means that reliably extracting information from a flow based on version-specific traits requires that middleboxes retain state for every connection ID they see.
-The Version Negotiation packet described in this document is not integrity-protected; it only has modest protection against insertion by off-path attackers. QUIC versions MUST define a mechanism that authenticates the values it contains.
-This document makes no request of IANA.
-| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, October 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using Transport Layer Security (TLS) to Secure QUIC", Internet-Draft draft-ietf-quic-tls, October 2019. | -
| [RFC5116] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
There are several traits of QUIC version 1 [QUIC-TRANSPORT] that are not protected from observation, but are nonetheless considered to be changeable when a new version is deployed.
-This section lists a sampling of incorrect assumptions that might be made based on knowledge of QUIC version 1. Some of these statements are not even true for QUIC version 1. This is not an exhaustive list, it is intended to be illustrative only.
-The following statements are NOT guaranteed to be true for every QUIC version:
- - -| QUIC | -C. Krasic | -
| Internet-Draft | -Netflix | -
| Intended status: Standards Track | -M. Bishop | -
| Expires: April 24, 2020 | -Akamai Technologies | -
| - | A. Frindell, Ed. | -
| - | |
| - | October 22, 2019 | -
QPACK: Header Compression for HTTP/3
- draft-ietf-quic-qpack-latest
This specification defines QPACK, a compression format for efficiently representing HTTP header fields, to be used in HTTP/3. This is a variation of HPACK header compression that seeks to reduce head-of-line blocking.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-qpack.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -The QUIC transport protocol [QUIC-TRANSPORT] is designed to support HTTP semantics, and its design subsumes many of the features of HTTP/2 [RFC7540]. HTTP/2 uses HPACK [RFC7541] for header compression. If HPACK were used for HTTP/3 [HTTP3], it would induce head-of-line blocking due to built-in assumptions of a total ordering across frames on all streams.
-QPACK reuses core concepts from HPACK, but is redesigned to allow correctness in the presence of out-of-order delivery, with flexibility for implementations to balance between resilience against head-of-line blocking and optimal compression ratio. The design goals are to closely approach the compression ratio of HPACK with substantially less head-of-line blocking under the same loss conditions.
-The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Definitions of terms that are used in this document:
- - -QPACK is a name, not an acronym.
-Diagrams use the format described in Section 3.1 of [RFC2360], with the following additional conventions:
- - -Like HPACK, QPACK uses two tables for associating header fields to indices. The static table (see Section 3.1) is predefined and contains common header fields (some of them with an empty value). The dynamic table (see Section 3.2) is built up over the course of the connection and can be used by the encoder to index header fields in the encoded header lists.
-QPACK defines unidirectional streams for sending instructions from encoder to decoder and vice versa.
-An encoder converts a header list into a header block by emitting either an indexed or a literal representation for each header field in the list (see Section 4.5). Indexed representations achieve high compression by replacing the literal name and possibly the value with an index to either the static or dynamic table. References to the static table and literal representations do not require any dynamic state and never risk head-of-line blocking. References to the dynamic table risk head-of-line blocking if the encoder has not received an acknowledgement indicating the entry is available at the decoder.
-An encoder MAY insert any entry in the dynamic table it chooses; it is not limited to header fields it is compressing.
-QPACK preserves the ordering of header fields within each header list. An encoder MUST emit header field representations in the order they appear in the input header list.
-QPACK is designed to contain the more complex state tracking to the encoder, while the decoder is relatively simple.
-An encoder MUST ensure that a header block which references a dynamic table entry is not processed by the decoder after the referenced entry has been evicted. Hence the encoder needs to retain information about each compressed header block that references the dynamic table until that header block is acknowledged by the decoder (see Section 4.4.1).
-A dynamic table entry is considered blocking and cannot be evicted until its insertion has been acknowledged and there are no outstanding unacknowledged references to the entry. In particular, a dynamic table entry that has never been referenced can still be blocking.
-An encoder MUST NOT insert an entry into the dynamic table (or duplicate an existing entry) if doing so would evict a blocking entry. In order to avoid this, an encoder that uses the dynamic table has to keep track of blocking entries.
- - -To ensure that the encoder is not prevented from adding new entries, the encoder can avoid referencing entries that are close to eviction. Rather than reference such an entry, the encoder can emit a Duplicate instruction (see Section 4.3.4), and reference the duplicate instead.
-Determining which entries are too close to eviction to reference is an encoder preference. One heuristic is to target a fixed amount of available space in the dynamic table: either unused space or space that can be reclaimed by evicting non-blocking entries. To achieve this, the encoder can maintain a draining index, which is the smallest absolute index (see Section 3.2.4) in the dynamic table that it will emit a reference for. As new entries are inserted, the encoder increases the draining index to maintain the section of the table that it will not reference. If the encoder does not create new references to entries with an absolute index lower than the draining index, the number of unacknowledged references to those entries will eventually become zero, allowing them to be evicted.
- - -- +----------+---------------------------------+--------+ - | Draining | Referenceable | Unused | - | Entries | Entries | Space | - +----------+---------------------------------+--------+ - ^ ^ ^ - | | | - Dropping Draining Index Insertion Point - Point --
Figure 1: Draining Dynamic Table Entries
-Because QUIC does not guarantee order between data on different streams, a decoder might encounter a header block that references a dynamic table entry that it has not yet received.
-Each header block contains a Required Insert Count (see Section 4.5.1), the lowest possible value for the Insert Count with which the header block can be decoded. For a header block with references to the dynamic table, the Required Insert Count is one larger than the largest absolute index of all referenced dynamic table entries. For a header block with no references to the dynamic table, the Required Insert Count is zero.
-When the decoder receives a header block with a Required Insert Count greater than its own Insert Count, the stream cannot be processed immediately, and is considered “blocked” (see Section 2.2.1).
-The decoder specifies an upper bound on the number of streams which can be blocked using the SETTINGS_QPACK_BLOCKED_STREAMS setting (see Section 5). An encoder MUST limit the number of streams which could become blocked to the value of SETTINGS_QPACK_BLOCKED_STREAMS at all times. If an decoder encounters more blocked streams than it promised to support, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-Note that the decoder might not become blocked on every stream which risks becoming blocked.
-An encoder can decide whether to risk having a stream become blocked. If permitted by the value of SETTINGS_QPACK_BLOCKED_STREAMS, compression efficiency can often be improved by referencing dynamic table entries that are still in transit, but if there is loss or reordering the stream can become blocked at the decoder. An encoder can avoid the risk of blocking by only referencing dynamic table entries which have been acknowledged, but this could mean using literals. Since literals make the header block larger, this can result in the encoder becoming blocked on congestion or flow control limits.
-The Known Received Count is the total number of dynamic table insertions and duplications acknowledged by the decoder. The encoder tracks the Known Received Count in order to identify which dynamic table entries can be referenced without potentially blocking a stream. The decoder tracks the Known Received Count in order to be able to send Insert Count Increment instructions (see Section 4.4.3).
-If a header block was potentially blocking, the encoder infers from receiving a Header Acknowledgement instruction (Section 4.4.1) that the decoder has received all dynamic table state necessary to process that header block. If the Required Insert Count of the acknowledged header block is greater than the current Known Received Count, the encoder updates the Known Received Count to the value of the Required Insert Count of the acknowledged header block.
-To acknowledge dynamic table entries which are not referenced by header blocks, for example because the encoder or the decoder have chosen not to risk blocked streams, the decoder sends an Insert Count Increment instruction (see Section 4.4.3).
-As in HPACK, the decoder processes header blocks and emits the corresponding header lists. It also processes instructions received on the encoder stream that modify the dynamic table. Note that header blocks and encoder stream instructions arrive on separate streams. This is unlike HPACK, where header blocks can contain instructions that modify the dynamic table, and there is no dedicated stream of HPACK instructions.
-The decoder MUST emit header fields in the order their representations appear in the input header block.
-Upon receipt of a header block, the decoder examines the Required Insert Count. When the Required Insert Count is less than or equal to the decoder’s Insert Count, the header block can be processed immediately. Otherwise, the stream on which the header block was received becomes blocked.
-While blocked, header block data SHOULD remain in the blocked stream’s flow control window. A stream becomes unblocked when the Insert Count becomes greater than or equal to the Required Insert Count for all header blocks the decoder has started reading from the stream.
-When processing header blocks, the decoder expects the Required Insert Count to exactly match the value defined in Section 2.1.3. If it encounters a smaller value than expected, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED (see Section 2.2.3). If it encounters a larger value than expected, it MAY treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-The decoder signals the following events by emitting decoder instructions (Section 4.4) on the decoder stream.
-After the decoder finishes decoding a header block containing dynamic table references, it MUST emit a Header Acknowledgement instruction (Section 4.4.1). A stream may carry multiple header blocks in the case of intermediate responses, trailers, and pushed requests. The encoder interprets each Header Acknowledgement instruction as acknowledging the earliest unacknowledged header block containing dynamic table references sent on the given stream.
-When an endpoint receives a stream reset before the end of a stream or before all header blocks are processed on that stream, or when it abandons reading of a stream, it generates a Stream Cancellation instruction (see Section 4.4.2). This signals to the encoder that all references to the dynamic table on that stream are no longer outstanding. A decoder with a maximum dynamic table capacity equal to zero (see Section 3.2.3) MAY omit sending Stream Cancellations, because the encoder cannot have any dynamic table references. An encoder cannot infer from this instruction that any updates to the dynamic table have been received.
-The Header Acknowledgement and Stream Cancellation instructions permit the encoder to remove references to entries in the dynamic table. When an entry with absolute index lower than the Known Received Count has zero references, then it is no longer considered blocking (see Section 2.1.2).
-After receiving new table entries on the encoder stream, the decoder chooses when to emit Insert Count Increment instructions (see Section 4.4.3). Emitting this instruction after adding each new dynamic table entry will provide the timeliest feedback to the encoder, but could be redundant with other decoder feedback. By delaying an Insert Count Increment instruction, the decoder might be able to coalesce multiple Insert Count Increment instructions, or replace them entirely with Header Acknowledgements (see Section 4.4.1). However, delaying too long may lead to compression inefficiencies if the encoder waits for an entry to be acknowledged before using it.
-If the decoder encounters a reference in a header block representation to a dynamic table entry which has already been evicted or which has an absolute index greater than or equal to the declared Required Insert Count (see Section 4.5.1), it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-If the decoder encounters a reference in an encoder instruction to a dynamic table entry which has already been evicted, it MUST treat this as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-Unlike in HPACK, entries in the QPACK static and dynamic tables are addressed separately. The following sections describe how entries in each table are addressed.
-The static table consists of a predefined static list of header fields, each of which has a fixed index over time. Its entries are defined in Appendix A.
-All entries in the static table have a name and a value. However, values can be empty (that is, have a length of 0). Each entry is identified by a unique index.
-Note that the QPACK static table is indexed from 0, whereas the HPACK static table is indexed from 1.
-When the decoder encounters an invalid static table index in a header block representation it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED. If this index is received on the encoder stream, this MUST be treated as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-The dynamic table consists of a list of header fields maintained in first-in, first-out order. Each HTTP/3 endpoint holds a dynamic table that is initially empty. Entries are added by encoder instructions received on the encoder stream (see Section 4.3).
-The dynamic table can contain duplicate entries (i.e., entries with the same name and same value). Therefore, duplicate entries MUST NOT be treated as an error by the decoder.
-Dynamic table entries can have empty values.
-The size of the dynamic table is the sum of the size of its entries.
-The size of an entry is the sum of its name’s length in bytes, its value’s length in bytes, and 32. The size of an entry is calculated using the length of its name and value without Huffman encoding applied.
-The encoder sets the capacity of the dynamic table, which serves as the upper limit on its size. The initial capacity of the dynamic table is zero. The encoder sends a Set Dynamic Table Capacity instruction (Section 4.3.1) with a non-zero capacity to begin using the dynamic table.
-Before a new entry is added to the dynamic table, entries are evicted from the end of the dynamic table until the size of the dynamic table is less than or equal to (table capacity - size of new entry). The encoder MUST NOT cause a blocking dynamic table entry to be evicted (see Section 2.1.2). The new entry is then added to the table. It is an error if the encoder attempts to add an entry that is larger than the dynamic table capacity; the decoder MUST treat this as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-A new entry can reference an entry in the dynamic table that will be evicted when adding this new entry into the dynamic table. Implementations are cautioned to avoid deleting the referenced name or value if the referenced entry is evicted from the dynamic table prior to inserting the new entry.
-Whenever the dynamic table capacity is reduced by the encoder (see Section 4.3.1), entries are evicted from the end of the dynamic table until the size of the dynamic table is less than or equal to the new table capacity. This mechanism can be used to completely clear entries from the dynamic table by setting a capacity of 0, which can subsequently be restored.
-To bound the memory requirements of the decoder, the decoder limits the maximum value the encoder is permitted to set for the dynamic table capacity. In HTTP/3, this limit is determined by the value of SETTINGS_QPACK_MAX_TABLE_CAPACITY sent by the decoder (see Section 5). The encoder MUST not set a dynamic table capacity that exceeds this maximum, but it can choose to use a lower dynamic table capacity (see Section 4.3.1).
-For clients using 0-RTT data in HTTP/3, the server’s maximum table capacity is the remembered value of the setting, or zero if the value was not previously sent. When the client’s 0-RTT value of the SETTING is zero, the server MAY set it to a non-zero value in its SETTINGS frame. If the remembered value is non-zero, the server MUST send the same non-zero value in its SETTINGS frame. If it specifies any other value, or omits SETTINGS_QPACK_MAX_TABLE_CAPACITY from SETTINGS, the encoder must treat this as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-For HTTP/3 servers and HTTP/3 clients when 0-RTT is not attempted or is rejected, the maximum table capacity is 0 until the encoder processes a SETTINGS frame with a non-zero value of SETTINGS_QPACK_MAX_TABLE_CAPACITY.
-When the maximum table capacity is zero, the encoder MUST NOT insert entries into the dynamic table, and MUST NOT send any encoder instructions on the encoder stream.
-Each entry possesses an absolute index which is fixed for the lifetime of that entry. The first entry inserted has an absolute index of “0”; indices increase by one with each insertion.
-Relative indices begin at zero and increase in the opposite direction from the absolute index. Determining which entry has a relative index of “0” depends on the context of the reference.
-In encoder instructions (see Section 4.3), a relative index of “0” refers to the most recently inserted value in the dynamic table. Note that this means the entry referenced by a given relative index will change while interpreting instructions on the encoder stream.
-- +-----+---------------+-------+ - | n-1 | ... | d | Absolute Index - + - - +---------------+ - - - + - | 0 | ... | n-d-1 | Relative Index - +-----+---------------+-------+ - ^ | - | V -Insertion Point Dropping Point - -n = count of entries inserted -d = count of entries dropped --
Example Dynamic Table Indexing - Encoder Stream
-Unlike in encoder instructions, relative indices in header block representations are relative to the Base at the beginning of the header block (see Section 4.5.1). This ensures that references are stable even if header blocks and dynamic table updates are processed out of order.
-In a header block a relative index of “0” refers to the entry with absolute index equal to Base - 1.
-- Required - Insert - Count Base - | | - V V - +-----+-----+-----+-----+-------+ - | n-1 | n-2 | n-3 | ... | d | Absolute Index - +-----+-----+ - +-----+ - + - | 0 | ... | n-d-3 | Relative Index - +-----+-----+-------+ - -n = count of entries inserted -d = count of entries dropped --
Example Dynamic Table Indexing - Relative Index in Header Block
-Post-Base indices are used in header block instructions for entries with absolute indices greater than or equal to Base, starting at 0 for the entry with absolute index equal to Base, and increasing in the same direction as the absolute index.
-Post-Base indices allow an encoder to process a header block in a single pass and include references to entries added while processing this (or other) header blocks.
-- Base - | - V - +-----+-----+-----+-----+-----+ - | n-1 | n-2 | n-3 | ... | d | Absolute Index - +-----+-----+-----+-----+-----+ - | 1 | 0 | Post-Base Index - +-----+-----+ - -n = count of entries inserted -d = count of entries dropped --
Example Dynamic Table Indexing - Post-Base Index in Header Block
-The prefixed integer from Section 5.1 of [RFC7541] is used heavily throughout this document. The format from [RFC7541] is used unmodified. Note, however, that QPACK uses some prefix sizes not actually used in HPACK.
-QPACK implementations MUST be able to decode integers up to 62 bits long.
-The string literal defined by Section 5.2 of [RFC7541] is also used throughout. This string format includes optional Huffman encoding.
-HPACK defines string literals to begin on a byte boundary. They begin with a single bit flag, denoted as ‘H’ in this document (indicating whether the string is Huffman-coded), followed by the Length encoded as a 7-bit prefix integer, and finally Length bytes of data. When Huffman encoding is enabled, the Huffman table from Appendix B of [RFC7541] is used without modification.
-This document expands the definition of string literals and permits them to begin other than on a byte boundary. An “N-bit prefix string literal” begins with the same Huffman flag, followed by the length encoded as an (N-1)-bit prefix integer. The prefix size, N, can have a value between 2 and 8 inclusive. The remainder of the string literal is unmodified.
-A string literal without a prefix length noted is an 8-bit prefix string literal and follows the definitions in [RFC7541] without modification.
-QPACK defines two unidirectional stream types:
- - -HTTP/3 endpoints contain a QPACK encoder and decoder. Each endpoint MUST initiate at most one encoder stream and at most one decoder stream. Receipt of a second instance of either stream type MUST be treated as a connection error of type HTTP_STREAM_CREATION_ERROR. These streams MUST NOT be closed. Closure of either unidirectional stream type MUST be treated as a connection error of type HTTP_CLOSED_CRITICAL_STREAM.
-An endpoint MAY avoid creating an encoder stream if it’s not going to be used (for example if its encoder doesn’t wish to use the dynamic table, or if the maximum size of the dynamic table permitted by the peer is zero).
-An endpoint MAY avoid creating a decoder stream if its decoder sets the maximum capacity of the dynamic table to zero.
-An endpoint MUST allow its peer to create an encoder stream and a decoder stream even if the connection’s settings prevent their use.
-An encoder sends encoder instructions on the encoder stream to set the capacity of the dynamic table and add dynamic table entries. Instructions adding table entries can use existing entries to avoid transmitting redundant information. The name can be transmitted as a reference to an existing entry in the static or the dynamic table or as a string literal. For entries which already exist in the dynamic table, the full entry can also be used by reference, creating a duplicate entry.
-This section specifies the following encoder instructions.
-An encoder informs the decoder of a change to the dynamic table capacity using an instruction which begins with the ‘001’ three-bit pattern. This is followed by the new dynamic table capacity represented as an integer with a 5-bit prefix (see Section 4.1.1).
- - -- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 0 | 0 | 1 | Capacity (5+) | -+---+---+---+-------------------+ --
Figure 2: Set Dynamic Table Capacity
-The new capacity MUST be lower than or equal to the limit described in Section 3.2.3. In HTTP/3, this limit is the value of the SETTINGS_QPACK_MAX_TABLE_CAPACITY parameter (see Section 5) received from the decoder. The decoder MUST treat a new dynamic table capacity value that exceeds this limit as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-Reducing the dynamic table capacity can cause entries to be evicted (see Section 3.2.2). This MUST NOT cause the eviction of blocking entries (see Section 2.1.2). Changing the capacity of the dynamic table is not acknowledged as this instruction does not insert an entry.
-An encoder adds an entry to the dynamic table where the header field name matches the header field name of an entry stored in the static or the dynamic table using an instruction that starts with the ‘1’ one-bit pattern. The second (‘T’) bit indicates whether the reference is to the static or dynamic table. The 6-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header name. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-The header name reference is followed by the header field value represented as a string literal (see Section 4.1.2).
-- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 1 | T | Name Index (6+) | - +---+---+-----------------------+ - | H | Value Length (7+) | - +---+---------------------------+ - | Value String (Length bytes) | - +-------------------------------+ --
Insert Header Field -- Indexed Name
-An encoder adds an entry to the dynamic table where both the header field name and the header field value are represented as string literals using an instruction that starts with the ‘01’ two-bit pattern.
-This is followed by the name represented as a 6-bit prefix string literal, and the value represented as an 8-bit prefix string literal (see Section 4.1.2).
-- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 0 | 1 | H | Name Length (5+) | - +---+---+---+-------------------+ - | Name String (Length bytes) | - +---+---------------------------+ - | H | Value Length (7+) | - +---+---------------------------+ - | Value String (Length bytes) | - +-------------------------------+ --
Insert Header Field -- New Name
-An encoder duplicates an existing entry in the dynamic table using an instruction that begins with the ‘000’ three-bit pattern. This is followed by the relative index of the existing entry represented as an integer with a 5-bit prefix (see Section 4.1.1.
- - -- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 0 | 0 | 0 | Index (5+) | - +---+---+---+-------------------+ --
Figure 3: Duplicate
-The existing entry is re-inserted into the dynamic table without resending either the name or the value. This is useful to avoid adding a reference to an older entry, which might block inserting new entries.
-A decoder sends decoder instructions on the decoder stream to inform the encoder about the processing of header blocks and table updates to ensure consistency of the dynamic table.
-This section specifies the following decoder instructions.
-After processing a header block whose declared Required Insert Count is not zero, the decoder emits a Header Acknowledgement instruction. The instruction begins with the ‘1’ one-bit pattern which is followed by the header block’s associated stream ID encoded as a 7-bit prefix integer (see Section 4.1.1).
-This instruction is used as described in Section 2.1.4 and in Section 2.2.2.
- - -- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 1 | Stream ID (7+) | -+---+---------------------------+ --
Figure 4: Header Acknowledgement
-If an encoder receives a Header Acknowledgement instruction referring to a stream on which every header block with a non-zero Required Insert Count has already been acknowledged, that MUST be treated as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-When a stream is reset or reading is abandoned, the decoder emits a Stream Cancellation instruction. The instruction begins with the ‘01’ two-bit pattern, which is followed by the stream ID of the affected stream encoded as a 6-bit prefix integer.
-This instruction is used as described in Section 2.2.2.
- - -- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 0 | 1 | Stream ID (6+) | -+---+---+-----------------------+ --
Figure 5: Stream Cancellation
-The Insert Count Increment instruction begins with the ‘00’ two-bit pattern, followed by the Increment encoded as a 6-bit prefix integer. The value of the Increment is the total number of dynamic table insertions and duplications processed by the decoder since the last time it sent a Header Acknowledgement instruction that increased the Known Received Count (see Section 2.1.4) or an Insert Count Increment instruction. The encoder uses this value to update the Known Received Count, as described in Section 2.2.2.
- - -- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 0 | 0 | Increment (6+) | -+---+---+-----------------------+ --
Figure 6: Insert Count Increment
-An encoder that receives an Increment field equal to zero, or one that increases the Known Received Count beyond what the encoder has sent MUST treat this as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-A header block consists of a prefix and a possibly empty sequence of representations defined in this section. Each representation corresponds to a single header field. These representations reference the static table or the dynamic table in a particular state, but do not modify that state.
-Header blocks are carried in frames on streams defined by the enclosing protocol.
-Each header block is prefixed with two integers. The Required Insert Count is encoded as an integer with an 8-bit prefix after the encoding described in Section 4.5.1.1). The Base is encoded as a sign bit (‘S’) and a Delta Base value with a 7-bit prefix (see Section 4.5.1.2).
- - -- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| Required Insert Count (8+) | -+---+---------------------------+ -| S | Delta Base (7+) | -+---+---------------------------+ -| Compressed Headers ... -+-------------------------------+ --
Figure 7: Header Block
-Required Insert Count identifies the state of the dynamic table needed to process the header block. Blocking decoders use the Required Insert Count to determine when it is safe to process the rest of the block.
-The encoder transforms the Required Insert Count as follows before encoding:
-- if ReqInsertCount == 0: - EncInsertCount = 0 - else: - EncInsertCount = (ReqInsertCount mod (2 * MaxEntries)) + 1 --
Here MaxEntries is the maximum number of entries that the dynamic table can have. The smallest entry has empty name and value strings and has the size of 32. Hence MaxEntries is calculated as
-- MaxEntries = floor( MaxTableCapacity / 32 ) --
MaxTableCapacity is the maximum capacity of the dynamic table as specified by the decoder (see Section 3.2.3).
-This encoding limits the length of the prefix on long-lived connections.
-The decoder can reconstruct the Required Insert Count using an algorithm such as the following. If the decoder encounters a value of EncodedInsertCount that could not have been produced by a conformant encoder, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-TotalNumberOfInserts is the total number of inserts into the decoder’s dynamic table.
-- FullRange = 2 * MaxEntries - if EncodedInsertCount == 0: - ReqInsertCount = 0 - else: - if EncodedInsertCount > FullRange: - Error - MaxValue = TotalNumberOfInserts + MaxEntries - - # MaxWrapped is the largest possible value of - # ReqInsertCount that is 0 mod 2*MaxEntries - MaxWrapped = floor(MaxValue / FullRange) * FullRange - ReqInsertCount = MaxWrapped + EncodedInsertCount - 1 - - # If ReqInsertCount exceeds MaxValue, the Encoder's value - # must have wrapped one fewer time - if ReqInsertCount > MaxValue: - if ReqInsertCount <= FullRange: - Error - ReqInsertCount -= FullRange - - # Value of 0 must be encoded as 0. - if ReqInsertCount == 0: - Error --
For example, if the dynamic table is 100 bytes, then the Required Insert Count will be encoded modulo 6. If a decoder has received 10 inserts, then an encoded value of 3 indicates that the Required Insert Count is 9 for the header block.
-The Base is used to resolve references in the dynamic table as described in Section 3.2.5.
-To save space, the Base is encoded relative to the Required Insert Count using a one-bit sign (‘S’) and the Delta Base value. A sign bit of 0 indicates that the Base is greater than or equal to the value of the Required Insert Count; the decoder adds the value of Delta Base to the Required Insert Count to determine the value of the Base. A sign bit of 1 indicates that the Base is less than the Required Insert Count; the decoder subtracts the value of Delta Base from the Required Insert Count and also subtracts one to determine the value of the Base. That is:
-- if S == 0: - Base = ReqInsertCount + DeltaBase - else: - Base = ReqInsertCount - DeltaBase - 1 --
A single-pass encoder determines the Base before encoding a header block. If the encoder inserted entries in the dynamic table while encoding the header block, Required Insert Count will be greater than the Base, so the encoded difference is negative and the sign bit is set to 1. If the header block did not reference the most recent entry in the table and did not insert any new entries, the Base will be greater than the Required Insert Count, so the delta will be positive and the sign bit is set to 0.
-An encoder that produces table updates before encoding a header block might set Base to the value of Required Insert Count. In such case, both the sign bit and the Delta Base will be set to zero.
-A header block that does not reference the dynamic table can use any value for the Base; setting Delta Base to zero is one of the most efficient encodings.
-For example, with a Required Insert Count of 9, a decoder receives a S bit of 1 and a Delta Base of 2. This sets the Base to 6 and enables post-base indexing for three entries. In this example, a relative index of 1 refers to the 5th entry that was added to the table; a post-base index of 1 refers to the 8th entry.
-An indexed header field representation identifies an entry in either the static table or the dynamic table and causes that header field to be added to the decoded header list.
-- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 1 | T | Index (6+) | -+---+---+-----------------------+ --
Indexed Header Field
-If the entry is in the static table, or in the dynamic table with an absolute index less than the Base, this representation starts with the ‘1’ 1-bit pattern, followed by the ‘T’ bit indicating whether the reference is into the static or dynamic table. The 6-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header field. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-If the entry is in the dynamic table with an absolute index greater than or equal to the Base, the representation starts with the ‘0001’ 4-bit pattern, followed by the post-base index (see Section 3.2.6) of the matching header field, represented as an integer with a 4-bit prefix (see Section 4.1.1).
-- 0 1 2 3 4 5 6 7 -+---+---+---+---+---+---+---+---+ -| 0 | 0 | 0 | 1 | Index (4+) | -+---+---+---+---+---------------+ --
Indexed Header Field with Post-Base Index
-A literal header field with name reference represents a header field where the header field name matches the header field name of an entry stored in the static table or the dynamic table.
-If the entry is in the static table, or in the dynamic table with an absolute index less than the Base, this representation starts with the ‘01’ two-bit pattern.
-The following bit, ‘N’, indicates whether an intermediary is permitted to add this header to the dynamic header table on subsequent hops. When the ‘N’ bit is set, the encoded header MUST always be encoded with a literal representation. In particular, when a peer sends a header field that it received represented as a literal header field with the ‘N’ bit set, it MUST use a literal representation to forward this header field. This bit is intended for protecting header field values that are not to be put at risk by compressing them (see Section 7 for more details).
-- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 0 | 1 | N | T |Name Index (4+)| - +---+---+---+---+---------------+ - | H | Value Length (7+) | - +---+---------------------------+ - | Value String (Length bytes) | - +-------------------------------+ --
Literal Header Field With Name Reference
-The fourth (‘T’) bit indicates whether the reference is to the static or dynamic table. The 4-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header name. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-Only the header field name is taken from the dynamic table entry; the header field value is encoded as an 8-bit prefix string literal (see Section 4.1.2).
-A literal header field with post-base name reference represents a header field where the name matches the header field name of a dynamic table entry with an absolute index greater than or equal to the Base.
-This representation starts with the ‘0000’ four-bit pattern. The fifth bit is the ‘N’ bit as described in Section 4.5.4. This is followed by a post-base index of the dynamic table entry (see Section 3.2.6) encoded as an integer with a 3-bit prefix (see Section 4.1.1).
-Only the header field name is taken from the dynamic table entry; the header field value is encoded as an 8-bit prefix string literal (see Section 4.1.2).
-- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 0 | 0 | 0 | 0 | N |NameIdx(3+)| - +---+---+---+---+---+-----------+ - | H | Value Length (7+) | - +---+---------------------------+ - | Value String (Length bytes) | - +-------------------------------+ --
Literal Header Field With Post-Base Name Reference
-The literal header field without name reference representation encodes a header field name and header field value as string literals.
-This representation begins with the ‘001’ three-bit pattern. The fourth bit is the ‘N’ bit as described in Section 4.5.4. The name follows, represented as a 4-bit prefix string literal, then the value, represented as an 8-bit prefix string literal (see Section 4.1.2).
-- 0 1 2 3 4 5 6 7 - +---+---+---+---+---+---+---+---+ - | 0 | 0 | 1 | N | H |NameLen(3+)| - +---+---+---+---+---+-----------+ - | Name String (Length bytes) | - +---+---------------------------+ - | H | Value Length (7+) | - +---+---------------------------+ - | Value String (Length bytes) | - +-------------------------------+ --
Literal Header Field Without Name Reference
-QPACK defines two settings which are included in the HTTP/3 SETTINGS frame.
- - -The following error codes are defined for HTTP/3 to indicate failures of QPACK which prevent the connection from continuing:
- - -TBD. Also see Section 7.1 of [RFC7541].
-While the negotiated limit on the dynamic table size accounts for much of the memory that can be consumed by a QPACK implementation, data which cannot be immediately sent due to flow control is not affected by this limit. Implementations should limit the size of unsent data, especially on the decoder stream where flexibility to choose what to send is limited. Possible responses to an excess of unsent data might include limiting the ability of the peer to open new streams, reading only from the encoder stream, or closing the connection.
-This document specifies two settings. The entries in the following table are registered in the “HTTP/3 Settings” registry established in [HTTP3].
-| Setting Name | -Code | -Specification | -Default | -
|---|---|---|---|
| QPACK_MAX_TABLE_CAPACITY | -0x1 | -Section 5 | -0 | -
| QPACK_BLOCKED_STREAMS | -0x7 | -Section 5 | -0 | -
This document specifies two stream types. The entries in the following table are registered in the “HTTP/3 Stream Type” registry established in [HTTP3].
-| Stream Type | -Code | -Specification | -Sender | -
|---|---|---|---|
| QPACK Encoder Stream | -0x02 | -Section 4.2 | -Both | -
| QPACK Decoder Stream | -0x03 | -Section 4.2 | -Both | -
This document specifies three error codes. The entries in the following table are registered in the “HTTP/3 Error Code” registry established in [HTTP3].
-| Name | -Code | -Description | -Specification | -
|---|---|---|---|
| HTTP_QPACK_DECOMPRESSION_FAILED | -0x200 | -Decompression of a header block failed | -Section 6 | -
| HTTP_QPACK_ENCODER_STREAM_ERROR | -0x201 | -Error on the encoder stream | -Section 6 | -
| HTTP_QPACK_DECODER_STREAM_ERROR | -0x202 | -Error on the decoder stream | -Section 6 | -
| [HTTP3] | --Bishop, M., "Hypertext Transfer Protocol Version 3 (HTTP/3)", Internet-Draft draft-ietf-quic-http, October 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, October 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC7541] | --Peon, R. and H. Ruellan, "HPACK: Header Compression for HTTP/2", RFC 7541, DOI 10.17487/RFC7541, May 2015. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC2360] | --Scott, G., "Guide for Internet Standards Writers", BCP 22, RFC 2360, DOI 10.17487/RFC2360, June 1998. | -
| [RFC7540] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| Index | -Name | -Value | -
|---|---|---|
| 0 | -:authority | -- |
| 1 | -:path | -/ | -
| 2 | -age | -0 | -
| 3 | -content-disposition | -- |
| 4 | -content-length | -0 | -
| 5 | -cookie | -- |
| 6 | -date | -- |
| 7 | -etag | -- |
| 8 | -if-modified-since | -- |
| 9 | -if-none-match | -- |
| 10 | -last-modified | -- |
| 11 | -link | -- |
| 12 | -location | -- |
| 13 | -referer | -- |
| 14 | -set-cookie | -- |
| 15 | -:method | -CONNECT | -
| 16 | -:method | -DELETE | -
| 17 | -:method | -GET | -
| 18 | -:method | -HEAD | -
| 19 | -:method | -OPTIONS | -
| 20 | -:method | -POST | -
| 21 | -:method | -PUT | -
| 22 | -:scheme | -http | -
| 23 | -:scheme | -https | -
| 24 | -:status | -103 | -
| 25 | -:status | -200 | -
| 26 | -:status | -304 | -
| 27 | -:status | -404 | -
| 28 | -:status | -503 | -
| 29 | -accept | -*/* | -
| 30 | -accept | -application/dns-message | -
| 31 | -accept-encoding | -gzip, deflate, br | -
| 32 | -accept-ranges | -bytes | -
| 33 | -access-control-allow-headers | -cache-control | -
| 34 | -access-control-allow-headers | -content-type | -
| 35 | -access-control-allow-origin | -* | -
| 36 | -cache-control | -max-age=0 | -
| 37 | -cache-control | -max-age=2592000 | -
| 38 | -cache-control | -max-age=604800 | -
| 39 | -cache-control | -no-cache | -
| 40 | -cache-control | -no-store | -
| 41 | -cache-control | -public, max-age=31536000 | -
| 42 | -content-encoding | -br | -
| 43 | -content-encoding | -gzip | -
| 44 | -content-type | -application/dns-message | -
| 45 | -content-type | -application/javascript | -
| 46 | -content-type | -application/json | -
| 47 | -content-type | -application/x-www-form-urlencoded | -
| 48 | -content-type | -image/gif | -
| 49 | -content-type | -image/jpeg | -
| 50 | -content-type | -image/png | -
| 51 | -content-type | -text/css | -
| 52 | -content-type | -text/html; charset=utf-8 | -
| 53 | -content-type | -text/plain | -
| 54 | -content-type | -text/plain;charset=utf-8 | -
| 55 | -range | -bytes=0- | -
| 56 | -strict-transport-security | -max-age=31536000 | -
| 57 | -strict-transport-security | -max-age=31536000; includesubdomains | -
| 58 | -strict-transport-security | -max-age=31536000; includesubdomains; preload | -
| 59 | -vary | -accept-encoding | -
| 60 | -vary | -origin | -
| 61 | -x-content-type-options | -nosniff | -
| 62 | -x-xss-protection | -1; mode=block | -
| 63 | -:status | -100 | -
| 64 | -:status | -204 | -
| 65 | -:status | -206 | -
| 66 | -:status | -302 | -
| 67 | -:status | -400 | -
| 68 | -:status | -403 | -
| 69 | -:status | -421 | -
| 70 | -:status | -425 | -
| 71 | -:status | -500 | -
| 72 | -accept-language | -- |
| 73 | -access-control-allow-credentials | -FALSE | -
| 74 | -access-control-allow-credentials | -TRUE | -
| 75 | -access-control-allow-headers | -* | -
| 76 | -access-control-allow-methods | -get | -
| 77 | -access-control-allow-methods | -get, post, options | -
| 78 | -access-control-allow-methods | -options | -
| 79 | -access-control-expose-headers | -content-length | -
| 80 | -access-control-request-headers | -content-type | -
| 81 | -access-control-request-method | -get | -
| 82 | -access-control-request-method | -post | -
| 83 | -alt-svc | -clear | -
| 84 | -authorization | -- |
| 85 | -content-security-policy | -script-src 'none'; object-src 'none'; base-uri 'none' | -
| 86 | -early-data | -1 | -
| 87 | -expect-ct | -- |
| 88 | -forwarded | -- |
| 89 | -if-range | -- |
| 90 | -origin | -- |
| 91 | -purpose | -prefetch | -
| 92 | -server | -- |
| 93 | -timing-allow-origin | -* | -
| 94 | -upgrade-insecure-requests | -1 | -
| 95 | -user-agent | -- |
| 96 | -x-forwarded-for | -- |
| 97 | -x-frame-options | -deny | -
| 98 | -x-frame-options | -sameorigin | -
Pseudo-code for single pass encoding, excluding handling of duplicates, non-blocking mode, and reference tracking.
--baseIndex = dynamicTable.baseIndex -largestReference = 0 -for header in headers: - staticIdx = staticTable.getIndex(header) - if staticIdx: - encodeIndexReference(streamBuffer, staticIdx) - continue - - dynamicIdx = dynamicTable.getIndex(header) - if !dynamicIdx: - # No matching entry. Either insert+index or encode literal - nameIdx = getNameIndex(header) - if shouldIndex(header) and dynamicTable.canIndex(header): - encodeLiteralWithIncrementalIndex(controlBuffer, nameIdx, - header) - dynamicTable.add(header) - dynamicIdx = dynamicTable.baseIndex - - if !dynamicIdx: - # Couldn't index it, literal - if nameIdx <= staticTable.size: - encodeLiteral(streamBuffer, nameIndex, header) - else: - # encode literal, possibly with nameIdx above baseIndex - encodeDynamicLiteral(streamBuffer, nameIndex, baseIndex, - header) - largestReference = max(largestReference, - dynamicTable.toAbsolute(nameIdx)) - else: - # Dynamic index reference - assert(dynamicIdx) - largestReference = max(largestReference, dynamicIdx) - # Encode dynamicIdx, possibly with dynamicIdx above baseIndex - encodeDynamicIndexReference(streamBuffer, dynamicIdx, - baseIndex) - -# encode the prefix -encodeInteger(prefixBuffer, 0x00, largestReference, 8) -if baseIndex >= largestReference: - encodeInteger(prefixBuffer, 0, baseIndex - largestReference, 7) -else: - encodeInteger(prefixBuffer, 0x80, - largestReference - baseIndex, 7) - -return controlBuffer, prefixBuffer + streamBuffer --
This draft draws heavily on the text of [RFC7541]. The indirect input of those authors is gratefully acknowledged, as well as ideas from:
- - -Buck’s contribution was supported by Google during his employment there.
-A substantial portion of Mike’s contribution was supported by Microsoft during his employment there.
-| QUIC | -J. Iyengar, Ed. | -
| Internet-Draft | -Fastly | -
| Intended status: Standards Track | -I. Swett, Ed. | -
| Expires: April 24, 2020 | -|
| - | October 22, 2019 | -
QUIC Loss Detection and Congestion Control
- draft-ietf-quic-recovery-latest
This document describes loss detection and congestion control mechanisms for QUIC.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-recovery.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades of transport and security experience, and implements mechanisms that make it attractive as a modern general-purpose transport. The QUIC protocol is described in [QUIC-TRANSPORT].
-QUIC implements the spirit of existing TCP loss recovery mechanisms, described in RFCs, various Internet-drafts, and also those prevalent in the Linux TCP implementation. This document describes QUIC congestion control and loss recovery, and where applicable, attributes the TCP equivalent in RFCs, Internet-drafts, academic papers, and/or TCP implementations.
-The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Definitions of terms that are used in this document:
- - -All transmissions in QUIC are sent with a packet-level header, which indicates the encryption level and includes a packet sequence number (referred to below as a packet number). The encryption level indicates the packet number space, as described in [QUIC-TRANSPORT]. Packet numbers never repeat within a packet number space for the lifetime of a connection. Packet numbers monotonically increase within a space, preventing ambiguity.
-This design obviates the need for disambiguating between transmissions and retransmissions and eliminates significant complexity from QUIC’s interpretation of TCP loss detection mechanisms.
-QUIC packets can contain multiple frames of different types. The recovery mechanisms ensure that data and frames that need reliable delivery are acknowledged or declared lost and sent in new packets as necessary. The types of frames contained in a packet affect recovery and congestion control logic:
- - -Readers familiar with TCP’s loss detection and congestion control will find algorithms here that parallel well-known TCP ones. Protocol differences between QUIC and TCP however contribute to algorithmic differences. We briefly describe these protocol differences below.
-QUIC uses separate packet number spaces for each encryption level, except 0-RTT and all generations of 1-RTT keys use the same packet number space. Separate packet number spaces ensures acknowledgement of packets sent with one level of encryption will not cause spurious retransmission of packets sent with a different encryption level. Congestion control and round-trip time (RTT) measurement are unified across packet number spaces.
-TCP conflates transmission order at the sender with delivery order at the receiver, which results in retransmissions of the same data carrying the same sequence number, and consequently leads to “retransmission ambiguity”. QUIC separates the two: QUIC uses a packet number to indicate transmission order, and any application data is sent in one or more streams, with delivery order determined by stream offsets encoded within STREAM frames.
-QUIC’s packet number is strictly increasing within a packet number space, and directly encodes transmission order. A higher packet number signifies that the packet was sent later, and a lower packet number signifies that the packet was sent earlier. When a packet containing ack-eliciting frames is detected lost, QUIC rebundles necessary frames in a new packet with a new packet number, removing ambiguity about which packet is acknowledged when an ACK is received. Consequently, more accurate RTT measurements can be made, spurious retransmissions are trivially detected, and mechanisms such as Fast Retransmit can be applied universally, based only on packet number.
-This design point significantly simplifies loss detection mechanisms for QUIC. Most TCP mechanisms implicitly attempt to infer transmission ordering based on TCP sequence numbers - a non-trivial task, especially when TCP timestamps are not available.
-QUIC ends a loss epoch when a packet sent after loss is declared is acknowledged. TCP waits for the gap in the sequence number space to be filled, and so if a segment is lost multiple times in a row, the loss epoch may not end for several round trips. Because both should reduce their congestion windows only once per epoch, QUIC will do it correctly once for every round trip that experiences loss, while TCP may only do it once across multiple round trips.
-QUIC ACKs contain information that is similar to TCP SACK, but QUIC does not allow any acked packet to be reneged, greatly simplifying implementations on both sides and reducing memory pressure on the sender.
-QUIC supports many ACK ranges, opposed to TCP’s 3 SACK ranges. In high loss environments, this speeds recovery, reduces spurious retransmits, and ensures forward progress without relying on timeouts.
-QUIC endpoints measure the delay incurred between when a packet is received and when the corresponding acknowledgment is sent, allowing a peer to maintain a more accurate round-trip time estimate (see Section 13.2 of [QUIC-TRANSPORT]).
-At a high level, an endpoint measures the time from when a packet was sent to when it is acknowledged as a round-trip time (RTT) sample. The endpoint uses RTT samples and peer-reported host delays (see Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical description of the connection’s RTT. An endpoint computes the following three values: the minimum value observed over the lifetime of the connection (min_rtt), an exponentially-weighted moving average (smoothed_rtt), and the variance in the observed RTT samples (rttvar).
-An endpoint generates an RTT sample on receiving an ACK frame that meets the following two conditions:
- - -The RTT sample, latest_rtt, is generated as the time elapsed since the largest acknowledged packet was sent:
--latest_rtt = ack_time - send_time_of_largest_acked --
An RTT sample is generated using only the largest acknowledged packet in the received ACK frame. This is because a peer reports host delays for only the largest acknowledged packet in an ACK frame. While the reported host delay is not used by the RTT sample measurement, it is used to adjust the RTT sample in subsequent computations of smoothed_rtt and rttvar Section 4.3.
-To avoid generating multiple RTT samples using the same packet, an ACK frame SHOULD NOT be used to update RTT estimates if it does not newly acknowledge the largest acknowledged packet.
-An RTT sample MUST NOT be generated on receiving an ACK frame that does not newly acknowledge at least one ack-eliciting packet. A peer does not send an ACK frame on receiving only non-ack-eliciting packets, so an ACK frame that is subsequently sent can include an arbitrarily large Ack Delay field. Ignoring such ACK frames avoids complications in subsequent smoothed_rtt and rttvar computations.
-A sender might generate multiple RTT samples per RTT when multiple ACK frames are received within an RTT. As suggested in [RFC6298], doing so might result in inadequate history in smoothed_rtt and rttvar. Ensuring that RTT estimates retain sufficient history is an open research question.
-min_rtt is the minimum RTT observed over the lifetime of the connection. min_rtt is set to the latest_rtt on the first sample in a connection, and to the lesser of min_rtt and latest_rtt on subsequent samples.
-An endpoint uses only locally observed times in computing the min_rtt and does not adjust for host delays reported by the peer. Doing so allows the endpoint to set a lower bound for the smoothed_rtt based entirely on what it observes (see Section 4.3), and limits potential underestimation due to erroneously-reported delays by the peer.
-smoothed_rtt is an exponentially-weighted moving average of an endpoint’s RTT samples, and rttvar is the endpoint’s estimated variance in the RTT samples.
-The calculation of smoothed_rtt uses path latency after adjusting RTT samples for host delays. For packets sent in the ApplicationData packet number space, a peer limits any delay in sending an acknowledgement for an ack-eliciting packet to no greater than the value it advertised in the max_ack_delay transport parameter. Consequently, when a peer reports an Ack Delay that is greater than its max_ack_delay, the delay is attributed to reasons out of the peer’s control, such as scheduler latency at the peer or loss of previous ACK frames. Any delays beyond the peer’s max_ack_delay are therefore considered effectively part of path delay and incorporated into the smoothed_rtt estimate.
-When adjusting an RTT sample using peer-reported acknowledgement delays, an endpoint:
- - -On the first RTT sample in a connection, the smoothed_rtt is set to the latest_rtt.
-smoothed_rtt and rttvar are computed as follows, similar to [RFC6298]. On the first RTT sample in a connection:
--smoothed_rtt = latest_rtt -rttvar = latest_rtt / 2 --
On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:
--ack_delay = min(Ack Delay in ACK Frame, max_ack_delay) -adjusted_rtt = latest_rtt -if (min_rtt + ack_delay < latest_rtt): - adjusted_rtt = latest_rtt - ack_delay -smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt -rttvar_sample = abs(smoothed_rtt - adjusted_rtt) -rttvar = 3/4 * rttvar + 1/4 * rttvar_sample --
QUIC senders use both ack information and timeouts to detect lost packets, and this section provides a description of these algorithms.
-If a packet is lost, the QUIC transport needs to recover from that loss, such as by retransmitting the data, sending an updated frame, or abandoning the frame. For more information, see Section 13.3 of [QUIC-TRANSPORT].
-Acknowledgement-based loss detection implements the spirit of TCP’s Fast Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK], SACK loss recovery [RFC6675], and RACK [RACK]. This section provides an overview of how these algorithms are implemented in QUIC.
-A packet is declared lost if it meets all the following conditions:
- - -The acknowledgement indicates that a packet sent later was delivered, while the packet and time thresholds provide some tolerance for packet reordering.
-Spuriously declaring packets as lost leads to unnecessary retransmissions and may result in degraded performance due to the actions of the congestion controller upon detecting loss. Implementations that detect spurious retransmissions and increase the reordering threshold in packets or time MAY choose to start with smaller initial reordering thresholds to minimize recovery latency.
-The RECOMMENDED initial value for the packet reordering threshold (kPacketThreshold) is 3, based on best practices for TCP loss detection [RFC5681] [RFC6675].
-Some networks may exhibit higher degrees of reordering, causing a sender to detect spurious losses. Implementers MAY use algorithms developed for TCP, such as TCP-NCR [RFC4653], to improve QUIC’s reordering resilience.
-Once a later packet packet within the same packet number space has been acknowledged, an endpoint SHOULD declare an earlier packet lost if it was sent a threshold amount of time in the past. To avoid declaring packets as lost too early, this time threshold MUST be set to at least kGranularity. The time threshold is:
--kTimeThreshold * max(smoothed_rtt, latest_rtt, kGranularity) --
If packets sent prior to the largest acknowledged packet cannot yet be declared lost, then a timer SHOULD be set for the remaining time.
-Using max(smoothed_rtt, latest_rtt) protects from the two following cases:
- - -The RECOMMENDED time threshold (kTimeThreshold), expressed as a round-trip time multiplier, is 9/8.
-Implementations MAY experiment with absolute thresholds, thresholds from previous connections, adaptive thresholds, or including RTT variance. Smaller thresholds reduce reordering resilience and increase spurious retransmissions, and larger thresholds increase loss detection delay.
-A Probe Timeout (PTO) triggers sending one or two probe datagrams when ack-eliciting packets are not acknowledged within the expected period of time or the handshake has not been completed. A PTO enables a connection to recover from loss of tail packets or acknowledgements. The PTO algorithm used in QUIC implements the reliability functions of Tail Loss Probe [TLP] [RACK], RTO [RFC5681] and F-RTO algorithms for TCP [RFC5682], and the timeout computation is based on TCP’s retransmission timeout period [RFC6298].
-When an ack-eliciting packet is transmitted, the sender schedules a timer for the PTO period as follows:
--PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay --
kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in Appendix A.2 and Appendix A.3.
-The PTO period is the amount of time that a sender ought to wait for an acknowledgement of a sent packet. This time period includes the estimated network roundtrip-time (smoothed_rtt), the variance in the estimate (4*rttvar), and max_ack_delay, to account for the maximum time by which a receiver might delay sending an acknowledgement.
-The PTO value MUST be set to at least kGranularity, to avoid the timer expiring immediately.
-When a PTO timer expires, the PTO period MUST be set to twice its current value. This exponential reduction in the sender’s rate is important because the PTOs might be caused by loss of packets or acknowledgements due to severe congestion. The life of a connection that is experiencing consecutive PTOs is limited by the endpoint’s idle timeout.
-A sender computes its PTO timer every time an ack-eliciting packet is sent. A sender might choose to optimize this by setting the timer fewer times if it knows that more ack-eliciting packets will be sent within a short period of time.
-The probe timer is not set if the time threshold Section 5.1.2 loss detection timer is set. The time threshold loss detection timer is expected to both expire earlier than the PTO and be less likely to spuriously retransmit data.
-The initial probe timeout for a new connection or new path SHOULD be set to twice the initial RTT. Resumed connections over the same network SHOULD use the previous connection’s final smoothed RTT value as the resumed connection’s initial RTT. If no previous RTT is available, the initial RTT SHOULD be set to 500ms, resulting in a 1 second initial timeout as recommended in [RFC6298].
-A connection MAY use the delay between sending a PATH_CHALLENGE and receiving a PATH_RESPONSE to seed initial_rtt for a new path, but the delay SHOULD NOT be considered an RTT sample.
-Until the server has validated the client’s address on the path, the amount of data it can send is limited, as specified in Section 8.1 of [QUIC-TRANSPORT]. Data at Initial encryption MUST be retransmitted before Handshake data and data at Handshake encryption MUST be retransmitted before any ApplicationData data. If no data can be sent, then the PTO alarm MUST NOT be armed until data has been received from the client.
-Since the server could be blocked until more packets are received from the client, it is the client’s responsibility to send packets to unblock the server until it is certain that the server has finished its address validation (see Section 8 of [QUIC-TRANSPORT]). That is, the client MUST set the probe timer if the client has not received an acknowledgement for one of its Handshake or 1-RTT packets.
-Prior to handshake completion, when few to none RTT samples have been generated, it is possible that the probe timer expiration is due to an incorrect RTT estimate at the client. To allow the client to improve its RTT estimate, the new packet that it sends MUST be ack-eliciting. If Handshake keys are available to the client, it MUST send a Handshake packet, and otherwise it MUST send an Initial packet in a UDP datagram of at least 1200 bytes.
-Initial packets and Handshake packets may never be acknowledged, but they are removed from bytes in flight when the Initial and Handshake keys are discarded.
-When a PTO timer expires, a sender MUST send at least one ack-eliciting packet as a probe, unless there is no data available to send. An endpoint MAY send up to two full-sized datagrams containing ack-eliciting packets, to avoid an expensive consecutive PTO expiration due to a single lost datagram.
-It is possible that the sender has no new or previously-sent data to send. As an example, consider the following sequence of events: new application data is sent in a STREAM frame, deemed lost, then retransmitted in a new packet, and then the original transmission is acknowledged. In the absence of any new application data, a PTO timer expiration now would find the sender with no new or previously-sent data to send.
-When there is no data to send, the sender SHOULD send a PING or other ack-eliciting frame in a single packet, re-arming the PTO timer.
-Alternatively, instead of sending an ack-eliciting packet, the sender MAY mark any packets still in flight as lost. Doing so avoids sending an additional packet, but increases the risk that loss is declared too aggressively, resulting in an unnecessary rate reduction by the congestion controller.
-Consecutive PTO periods increase exponentially, and as a result, connection recovery latency increases exponentially as packets continue to be dropped in the network. Sending two packets on PTO expiration increases resilience to packet drops, thus reducing the probability of consecutive PTO events.
-Probe packets sent on a PTO MUST be ack-eliciting. A probe packet SHOULD carry new data when possible. A probe packet MAY carry retransmitted unacknowledged data when new data is unavailable, when flow control does not permit new data to be sent, or to opportunistically reduce loss recovery delay. Implementations MAY use alternative strategies for determining the content of probe packets, including sending new or retransmitted data based on the application’s priorities.
-When the PTO timer expires multiple times and new data cannot be sent, implementations must choose between sending the same payload every time or sending different payloads. Sending the same payload may be simpler and ensures the highest priority frames arrive first. Sending different payloads each time reduces the chances of spurious retransmission.
-Delivery or loss of packets in flight is established when an ACK frame is received that newly acknowledges one or more packets.
-A PTO timer expiration event does not indicate packet loss and MUST NOT cause prior unacknowledged packets to be marked as lost. When an acknowledgement is received that newly acknowledges packets, loss detection proceeds as dictated by packet and time threshold mechanisms; see Section 5.1.
-A Retry or Version Negotiation packet causes a client to send another Initial packet, effectively restarting the connection process and resetting congestion control and loss recovery state, including resetting any pending timers. Either packet indicates that the Initial was received but not processed. Neither packet can be treated as an acknowledgment for the Initial.
-The client MAY however compute an RTT estimate to the server as the time period from when the first Initial was sent to when a Retry or a Version Negotiation packet is received. The client MAY use this value to seed the RTT estimator for a subsequent connection attempt to the server.
-When packet protection keys are discarded (see Section 4.9 of [QUIC-TLS]), all packets that were sent with those keys can no longer be acknowledged because their acknowledgements cannot be processed anymore. The sender MUST discard all recovery state associated with those packets and MUST remove them from the count of bytes in flight.
-Endpoints stop sending and receiving Initial packets once they start exchanging Handshake packets (see Section 17.2.2.1 of [QUIC-TRANSPORT]). At this point, recovery state for all in-flight Initial packets is discarded.
-When 0-RTT is rejected, recovery state for all in-flight 0-RTT packets is discarded.
-If a server accepts 0-RTT, but does not buffer 0-RTT packets that arrive before Initial packets, early 0-RTT packets will be declared lost, but that is expected to be infrequent.
-It is expected that keys are discarded after packets encrypted with them would be acknowledged or declared lost. Initial secrets however might be destroyed sooner, as soon as handshake keys are available (see Section 4.9.1 of [QUIC-TLS]).
-The majority of constants were derived from best common practices among widely deployed TCP implementations on the internet. Exceptions follow.
-A shorter delayed ack time of 25ms was chosen because longer delayed acks can delay loss recovery and for the small number of connections where less than packet per 25ms is delivered, acking every packet is beneficial to congestion control and loss recovery.
-QUIC’s congestion control is based on TCP NewReno [RFC6582]. NewReno is a congestion window based congestion control. QUIC specifies the congestion window in bytes rather than packets due to finer control and the ease of appropriate byte counting [RFC3465].
-QUIC hosts MUST NOT send packets if they would increase bytes_in_flight (defined in Appendix B.2) beyond the available congestion window, unless the packet is a probe packet sent after a PTO timer expires, as described in Section 5.2.
-Implementations MAY use other congestion control algorithms, such as Cubic [RFC8312], and endpoints MAY use different algorithms from one another. The signals QUIC provides for congestion control are generic and are designed to support different algorithms.
-If a path has been verified to support ECN, QUIC treats a Congestion Experienced codepoint in the IP header as a signal of congestion. This document specifies an endpoint’s response when its peer receives packets with the Congestion Experienced codepoint. As discussed in [RFC8311], endpoints are permitted to experiment with other response functions.
-QUIC begins every connection in slow start and exits slow start upon loss or upon increase in the ECN-CE counter. QUIC re-enters slow start anytime the congestion window is less than ssthresh, which only occurs after persistent congestion is declared. While in slow start, QUIC increases the congestion window by the number of bytes acknowledged when each acknowledgment is processed.
-Slow start exits to congestion avoidance. Congestion avoidance in NewReno uses an additive increase multiplicative decrease (AIMD) approach that increases the congestion window by one maximum packet size per congestion window acknowledged. When a loss is detected, NewReno halves the congestion window and sets the slow start threshold to the new congestion window.
-Recovery is a period of time beginning with detection of a lost packet or an increase in the ECN-CE counter. Because QUIC does not retransmit packets, it defines the end of recovery as a packet sent after the start of recovery being acknowledged. This is slightly different from TCP’s definition of recovery, which ends when the lost packet that started recovery is acknowledged.
-The recovery period limits congestion window reduction to once per round trip. During recovery, the congestion window remains unchanged irrespective of new losses or increases in the ECN-CE counter.
-During the handshake, some packet protection keys might not be available when a packet arrives. In particular, Handshake and 0-RTT packets cannot be processed until the Initial packets arrive, and 1-RTT packets cannot be processed until the handshake completes. Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might arrive before the peer has packet protection keys to process those packets.
-Probe packets MUST NOT be blocked by the congestion controller. A sender MUST however count these packets as being additionally in flight, since these packets add network load without establishing packet loss. Note that sending probe packets might cause the sender’s bytes in flight to exceed the congestion window until an acknowledgement is received that establishes loss or delivery of packets.
-When an ACK frame is received that establishes loss of all in-flight packets sent over a long enough period of time, the network is considered to be experiencing persistent congestion. Commonly, this can be established by consecutive PTOs, but since the PTO timer is reset when a new ack-eliciting packet is sent, an explicit duration must be used to account for those cases where PTOs do not occur or are substantially delayed. This duration is computed as follows:
--(smoothed_rtt + 4 * rttvar + max_ack_delay) * - kPersistentCongestionThreshold --
For example, assume:
-smoothed_rtt = 1 rttvar = 0 max_ack_delay = 0 kPersistentCongestionThreshold = 3
-If an ack-eliciting packet is sent at time = 0, the following scenario would illustrate persistent congestion:
-| t=0 | -Send Pkt #1 (App Data) | -
|---|---|
| t=1 | -Send Pkt #2 (PTO 1) | -
| t=3 | -Send Pkt #3 (PTO 2) | -
| t=7 | -Send Pkt #4 (PTO 3) | -
| t=8 | -Recv ACK of Pkt #4 | -
The first three packets are determined to be lost when the ACK of packet 4 is received at t=8. The congestion period is calculated as the time between the oldest and newest lost packets: (3 - 0) = 3. The duration for persistent congestion is equal to: (1 * kPersistentCongestionThreshold) = 3. Because the threshold was reached and because none of the packets between the oldest and the newest packets are acknowledged, the network is considered to have experienced persistent congestion.
-When persistent congestion is established, the sender’s congestion window MUST be reduced to the minimum congestion window (kMinimumWindow). This response of collapsing the congestion window on persistent congestion is functionally similar to a sender’s response on a Retransmission Timeout (RTO) in TCP [RFC5681] after Tail Loss Probes (TLP) [TLP].
-This document does not specify a pacer, but it is RECOMMENDED that a sender pace sending of all in-flight packets based on input from the congestion controller. For example, a pacer might distribute the congestion window over the smoothed RTT when used with a window-based controller, and a pacer might use the rate estimate of a rate-based controller.
-An implementation should take care to architect its congestion controller to work well with a pacer. For instance, a pacer might wrap the congestion controller and control the availability of the congestion window, or a pacer might pace out packets handed to it by the congestion controller. Timely delivery of ACK frames is important for efficient loss recovery. Packets containing only ACK frames should therefore not be paced, to avoid delaying their delivery to the peer.
-Sending multiple packets into the network without any delay between them creates a packet burst that might cause short-term congestion and losses. Implementations MUST either use pacing or limit such bursts to the minimum of 10 * kMaxDatagramSize and max(2* kMaxDatagramSize, 14720)), the same as the recommended initial congestion window.
-As an example of a well-known and publicly available implementation of a flow pacer, implementers are referred to the Fair Queue packet scheduler (fq qdisc) in Linux (3.11 onwards).
-When bytes in flight is smaller than the congestion window and sending is not pacing limited, the congestion window is under-utilized. When this occurs, the congestion window SHOULD NOT be increased in either slow start or congestion avoidance. This can happen due to insufficient application data or flow control credit.
-A sender MAY use the pipeACK method described in section 4.3 of [RFC7661] to determine if the congestion window is sufficiently utilized.
-A sender that paces packets (see Section 6.8) might delay sending packets and not fully utilize the congestion window due to this delay. A sender should not consider itself application limited if it would have fully utilized the congestion window without pacing delay.
-A sender MAY implement alternative mechanisms to update its congestion window after periods of under-utilization, such as those proposed for TCP in [RFC7661].
-Congestion control fundamentally involves the consumption of signals – both loss and ECN codepoints – from unauthenticated entities. On-path attackers can spoof or alter these signals. An attacker can cause endpoints to reduce their sending rate by dropping packets, or alter send rate by changing ECN codepoints.
-Packets that carry only ACK frames can be heuristically identified by observing packet size. Acknowledgement patterns may expose information about link characteristics or application behavior. Endpoints can use PADDING frames or bundle acknowledgments with other frames to reduce leaked information.
-A receiver can misreport ECN markings to alter the congestion response of a sender. Suppressing reports of ECN-CE markings could cause a sender to increase their send rate. This increase could result in congestion and loss.
-A sender MAY attempt to detect suppression of reports by marking occasional packets that they send with ECN-CE. If a packet marked with ECN-CE is not reported as having been marked when the packet is acknowledged, the sender SHOULD then disable ECN for that path.
-Reporting additional ECN-CE markings will cause a sender to reduce their sending rate, which is similar in effect to advertising reduced connection flow control limits and so no advantage is gained by doing so.
-Endpoints choose the congestion controller that they use. Though congestion controllers generally treat reports of ECN-CE markings as equivalent to loss [RFC8311], the exact response for each controller could be different. Failure to correctly respond to information about ECN markings is therefore difficult to detect.
-This document has no IANA actions. Yet.
-| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using TLS to Secure QUIC", Internet-Draft draft-ietf-quic-tls, October 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, October 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC8311] | --Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018. | -
We now describe an example implementation of the loss detection mechanisms described in Section 5.
-To correctly implement congestion control, a QUIC sender tracks every ack-eliciting packet until the packet is acknowledged or lost. It is expected that implementations will be able to access this information by packet number and crypto context and store the per-packet fields (Appendix A.1.1) for loss recovery and congestion control.
-After a packet is declared lost, the endpoint can track it for an amount of time comparable to the maximum expected packet reordering, such as 1 RTT. This allows for detection of spurious retransmissions.
-Sent packets are tracked for each packet number space, and ACK processing only applies to a single space.
-Constants used in loss recovery are based on a combination of RFCs, papers, and common practice. Some may need to be changed or negotiated in order to better suit a variety of environments.
- - -
- enum kPacketNumberSpace {
- Initial,
- Handshake,
- ApplicationData,
- }
-
-Variables required to implement the congestion control mechanisms are described in this section.
- - -At the beginning of the connection, initialize the loss detection variables as follows:
-- loss_detection_timer.reset() - pto_count = 0 - latest_rtt = 0 - smoothed_rtt = 0 - rttvar = 0 - min_rtt = 0 - max_ack_delay = 0 - time_of_last_sent_ack_eliciting_packet = 0 - for pn_space in [ Initial, Handshake, ApplicationData ]: - largest_acked_packet[pn_space] = infinite - loss_time[pn_space] = 0 --
After a packet is sent, information about the packet is stored. The parameters to OnPacketSent are described in detail above in Appendix A.1.1.
-Pseudocode for OnPacketSent follows:
-- OnPacketSent(packet_number, pn_space, ack_eliciting, - in_flight, sent_bytes): - sent_packets[pn_space][packet_number].packet_number = - packet_number - sent_packets[pn_space][packet_number].time_sent = now - sent_packets[pn_space][packet_number].ack_eliciting = - ack_eliciting - sent_packets[pn_space][packet_number].in_flight = in_flight - if (in_flight): - if (ack_eliciting): - time_of_last_sent_ack_eliciting_packet = now - OnPacketSentCC(sent_bytes) - sent_packets[pn_space][packet_number].size = sent_bytes - SetLossDetectionTimer() --
When an ACK frame is received, it may newly acknowledge any number of packets.
-Pseudocode for OnAckReceived and UpdateRtt follow:
--OnAckReceived(ack, pn_space): - if (largest_acked_packet[pn_space] == infinite): - largest_acked_packet[pn_space] = ack.largest_acked - else: - largest_acked_packet[pn_space] = - max(largest_acked_packet[pn_space], ack.largest_acked) - - // Nothing to do if there are no newly acked packets. - newly_acked_packets = DetermineNewlyAckedPackets(ack, pn_space) - if (newly_acked_packets.empty()): - return - - // If the largest acknowledged is newly acked and - // at least one ack-eliciting was newly acked, update the RTT. - if (sent_packets[pn_space].contains(ack.largest_acked) && - IncludesAckEliciting(newly_acked_packets)): - latest_rtt = - now - sent_packets[pn_space][ack.largest_acked].time_sent - ack_delay = 0 - if (pn_space == ApplicationData): - ack_delay = ack.ack_delay - UpdateRtt(ack_delay) - - // Process ECN information if present. - if (ACK frame contains ECN information): - ProcessECN(ack, pn_space) - - for acked_packet in newly_acked_packets: - OnPacketAcked(acked_packet.packet_number, pn_space) - - DetectLostPackets(pn_space) - - pto_count = 0 - - SetLossDetectionTimer() - - -UpdateRtt(ack_delay): - // First RTT sample. - if (smoothed_rtt == 0): - min_rtt = latest_rtt - smoothed_rtt = latest_rtt - rttvar = latest_rtt / 2 - return - - // min_rtt ignores ack delay. - min_rtt = min(min_rtt, latest_rtt) - // Limit ack_delay by max_ack_delay - ack_delay = min(ack_delay, max_ack_delay) - // Adjust for ack delay if plausible. - adjusted_rtt = latest_rtt - if (latest_rtt > min_rtt + ack_delay): - adjusted_rtt = latest_rtt - ack_delay - - rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt) - smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt --
When a packet is acknowledged for the first time, the following OnPacketAcked function is called. Note that a single ACK frame may newly acknowledge several packets. OnPacketAcked must be called once for each of these newly acknowledged packets.
-OnPacketAcked takes two parameters: acked_packet, which is the struct detailed in Appendix A.1.1, and the packet number space that this ACK frame was sent for.
-Pseudocode for OnPacketAcked follows:
-- OnPacketAcked(acked_packet, pn_space): - if (acked_packet.in_flight): - OnPacketAckedCC(acked_packet) - sent_packets[pn_space].remove(acked_packet.packet_number) --
QUIC loss detection uses a single timer for all timeout loss detection. The duration of the timer is based on the timer’s mode, which is set in the packet and timer events further below. The function SetLossDetectionTimer defined below shows how the single timer is set.
-This algorithm may result in the timer being set in the past, particularly if timers wake up late. Timers set in the past SHOULD fire immediately.
-Pseudocode for SetLossDetectionTimer follows:
--// Returns the earliest loss_time and the packet number -// space it's from. Returns 0 if all times are 0. -GetEarliestLossTime(): - time = loss_time[Initial] - space = Initial - for pn_space in [ Handshake, ApplicationData ]: - if (loss_time[pn_space] != 0 && - (time == 0 || loss_time[pn_space] < time)): - time = loss_time[pn_space]; - space = pn_space - return time, space - -PeerNotAwaitingAddressValidation(): - # Assume clients validate the server's address implicitly. - if (endpoint is server): - return true - # Servers complete address validation when a - # protected packet is received. - return has received Handshake ACK || - has received 1-RTT ACK - -SetLossDetectionTimer(): - loss_time, _ = GetEarliestLossTime() - if (loss_time != 0): - // Time threshold loss detection. - loss_detection_timer.update(loss_time) - return - - if (no ack-eliciting packets in flight && - PeerNotAwaitingAddressValidation()): - loss_detection_timer.cancel() - return - - // Use a default timeout if there are no RTT measurements - if (smoothed_rtt == 0): - timeout = 2 * kInitialRtt - else: - // Calculate PTO duration - timeout = smoothed_rtt + max(4 * rttvar, kGranularity) + - max_ack_delay - timeout = timeout * (2 ^ pto_count) - - loss_detection_timer.update( - time_of_last_sent_ack_eliciting_packet + timeout) --
When the loss detection timer expires, the timer’s mode determines the action to be performed.
-Pseudocode for OnLossDetectionTimeout follows:
--OnLossDetectionTimeout(): - loss_time, pn_space = GetEarliestLossTime() - if (loss_time != 0): - // Time threshold loss Detection - DetectLostPackets(pn_space) - SetLossDetectionTimer() - return - - if (endpoint is client without 1-RTT keys): - // Client sends an anti-deadlock packet: Initial is padded - // to earn more anti-amplification credit, - // a Handshake packet proves address ownership. - if (has Handshake keys): - SendOneAckElicitingHandshakePacket() - else: - SendOneAckElicitingPaddedInitialPacket() - else: - // PTO. Send new data if available, else retransmit old data. - // If neither is available, send a single PING frame. - SendOneOrTwoAckElicitingPackets() - - pto_count++ - SetLossDetectionTimer() --
DetectLostPackets is called every time an ACK is received and operates on the sent_packets for that packet number space.
-Pseudocode for DetectLostPackets follows:
-
-DetectLostPackets(pn_space):
- assert(largest_acked_packet[pn_space] != infinite)
- loss_time[pn_space] = 0
- lost_packets = {}
- loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)
-
- // Minimum time of kGranularity before packets are deemed lost.
- loss_delay = max(loss_delay, kGranularity)
-
- // Packets sent before this time are deemed lost.
- lost_send_time = now() - loss_delay
-
- foreach unacked in sent_packets[pn_space]:
- if (unacked.packet_number > largest_acked_packet[pn_space]):
- continue
-
- // Mark packet as lost, or set time when it should be marked.
- if (unacked.time_sent <= lost_send_time ||
- largest_acked_packet[pn_space] >=
- unacked.packet_number + kPacketThreshold):
- sent_packets[pn_space].remove(unacked.packet_number)
- if (unacked.in_flight):
- lost_packets.insert(unacked)
- else:
- if (loss_time[pn_space] == 0):
- loss_time[pn_space] = unacked.time_sent + loss_delay
- else:
- loss_time[pn_space] = min(loss_time[pn_space],
- unacked.time_sent + loss_delay)
-
- // Inform the congestion controller of lost packets and
- // let it decide whether to retransmit immediately.
- if (!lost_packets.empty()):
- OnPacketsLost(lost_packets)
-
-We now describe an example implementation of the congestion controller described in Section 6.
-Constants used in congestion control are based on a combination of RFCs, papers, and common practice. Some may need to be changed or negotiated in order to better suit a variety of environments.
- - -Variables required to implement the congestion control mechanisms are described in this section.
- - -At the beginning of the connection, initialize the congestion control variables as follows:
-- congestion_window = kInitialWindow - bytes_in_flight = 0 - congestion_recovery_start_time = 0 - ssthresh = infinite - for pn_space in [ Initial, Handshake, ApplicationData ]: - ecn_ce_counters[pn_space] = 0 --
Whenever a packet is sent, and it contains non-ACK frames, the packet increases bytes_in_flight.
-- OnPacketSentCC(bytes_sent): - bytes_in_flight += bytes_sent --
Invoked from loss detection’s OnPacketAcked and is supplied with the acked_packet from sent_packets.
-- InCongestionRecovery(sent_time): - return sent_time <= congestion_recovery_start_time - - OnPacketAckedCC(acked_packet): - // Remove from bytes_in_flight. - bytes_in_flight -= acked_packet.size - if (InCongestionRecovery(acked_packet.time_sent)): - // Do not increase congestion window in recovery period. - return - if (IsAppLimited()): - // Do not increase congestion_window if application - // limited. - return - if (congestion_window < ssthresh): - // Slow start. - congestion_window += acked_packet.size - else: - // Congestion avoidance. - congestion_window += kMaxDatagramSize * acked_packet.size - / congestion_window --
Invoked from ProcessECN and OnPacketsLost when a new congestion event is detected. May start a new recovery period and reduces the congestion window.
-- CongestionEvent(sent_time): - // Start a new congestion event if packet was sent after the - // start of the previous congestion recovery period. - if (!InCongestionRecovery(sent_time)): - congestion_recovery_start_time = Now() - congestion_window *= kLossReductionFactor - congestion_window = max(congestion_window, kMinimumWindow) - ssthresh = congestion_window --
Invoked when an ACK frame with an ECN section is received from the peer.
-- ProcessECN(ack, pn_space): - // If the ECN-CE counter reported by the peer has increased, - // this could be a new congestion event. - if (ack.ce_counter > ecn_ce_counters[pn_space]): - ecn_ce_counters[pn_space] = ack.ce_counter - CongestionEvent(sent_packets[ack.largest_acked].time_sent) --
Invoked from DetectLostPackets when packets are deemed lost.
-- InPersistentCongestion(largest_lost_packet): - pto = smoothed_rtt + max(4 * rttvar, kGranularity) + - max_ack_delay - congestion_period = pto * kPersistentCongestionThreshold - // Determine if all packets in the time period before the - // newest lost packet, including the edges, are marked - // lost - return AreAllPacketsLost(largest_lost_packet, - congestion_period) - - OnPacketsLost(lost_packets): - // Remove lost packets from bytes_in_flight. - for (lost_packet : lost_packets): - bytes_in_flight -= lost_packet.size - largest_lost_packet = lost_packets.last() - CongestionEvent(largest_lost_packet.time_sent) - - // Collapse congestion window if persistent congestion - if (InPersistentCongestion(largest_lost_packet)): - congestion_window = kMinimumWindow --
Issue and pull request numbers are listed with a leading octothorp.
-No significant changes.
-No significant changes.
-No significant changes.
-No significant changes.
-No significant changes.
-| QUIC | -M. Thomson, Ed. | -
| Internet-Draft | -Mozilla | -
| Intended status: Standards Track | -S. Turner, Ed. | -
| Expires: April 24, 2020 | -sn3rd | -
| - | October 22, 2019 | -
Using TLS to Secure QUIC
- draft-ietf-quic-tls-latest
This document describes how Transport Layer Security (TLS) is used to secure QUIC.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-tls.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -This document describes how QUIC [QUIC-TRANSPORT] is secured using TLS [TLS13].
-TLS 1.3 provides critical latency improvements for connection establishment over previous versions. Absent packet loss, most new connections can be established and secured within a single round trip; on subsequent connections between the same client and server, the client can often send application data immediately, that is, using a zero round trip setup.
-This document describes how TLS acts as a security component of QUIC.
-The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-This document uses the terminology established in [QUIC-TRANSPORT].
-For brevity, the acronym TLS is used to refer to TLS 1.3, though a newer version could be used (see Section 4.2).
-TLS provides two endpoints with a way to establish a means of communication over an untrusted medium (that is, the Internet) that ensures that messages they exchange cannot be observed, modified, or forged.
-Internally, TLS is a layered protocol, with the structure shown below:
--+--------------+--------------+--------------+ -| Handshake | Alerts | Application | -| Layer | | Data | -| | | | -+--------------+--------------+--------------+ -| | -| Record Layer | -| | -+--------------------------------------------+ --
Each upper layer (handshake, alerts, and application data) is carried as a series of typed TLS records. Records are individually cryptographically protected and then transmitted over a reliable transport (typically TCP) which provides sequencing and guaranteed delivery.
-Change Cipher Spec records cannot be sent in QUIC.
-The TLS authenticated key exchange occurs between two entities: client and server. The client initiates the exchange and the server responds. If the key exchange completes successfully, both client and server will agree on a secret. TLS supports both pre-shared key (PSK) and Diffie-Hellman (DH) key exchanges. PSK is the basis for 0-RTT; the latter provides perfect forward secrecy (PFS) when the DH keys are destroyed.
-After completing the TLS handshake, the client will have learned and authenticated an identity for the server and the server is optionally able to learn and authenticate an identity for the client. TLS supports X.509 [RFC5280] certificate-based authentication for both server and client.
-The TLS key exchange is resistant to tampering by attackers and it produces shared secrets that cannot be controlled by either participating peer.
-TLS provides two basic handshake modes of interest to QUIC:
- - -A simplified TLS handshake with 0-RTT application data is shown in Figure 1. Note that this omits the EndOfEarlyData message, which is not used in QUIC (see Section 8.3).
- - -
- Client Server
-
- ClientHello
- (0-RTT Application Data) -------->
- ServerHello
- {EncryptedExtensions}
- {Finished}
- <-------- [Application Data]
- {Finished} -------->
-
- [Application Data] <-------> [Application Data]
-
- () Indicates messages protected by early data (0-RTT) keys
- {} Indicates messages protected using handshake keys
- [] Indicates messages protected using application data
- (1-RTT) keys
-
-Figure 1: TLS Handshake with 0-RTT
-Data is protected using a number of encryption levels:
- - -Application data may appear only in the early data and application data levels. Handshake and Alert messages may appear in any level.
-The 0-RTT handshake is only possible if the client and server have previously communicated. In the 1-RTT handshake, the client is unable to send protected application data until it has received all of the handshake messages sent by the server.
-QUIC [QUIC-TRANSPORT] assumes responsibility for the confidentiality and integrity protection of packets. For this it uses keys derived from a TLS handshake [TLS13], but instead of carrying TLS records over QUIC (as with TCP), TLS Handshake and Alert messages are carried directly over the QUIC transport, which takes over the responsibilities of the TLS record layer, as shown below.
--+--------------+--------------+ +-------------+ -| TLS | TLS | | QUIC | -| Handshake | Alerts | | Applications| -| | | | (h3, etc.) | -+--------------+--------------+-+-------------+ -| | -| QUIC Transport | -| (streams, reliability, congestion, etc.) | -| | -+---------------------------------------------+ -| | -| QUIC Packet Protection | -| | -+---------------------------------------------+ --
QUIC also relies on TLS for authentication and negotiation of parameters that are critical to security and performance.
-Rather than a strict layering, these two protocols are co-dependent: QUIC uses the TLS handshake; TLS uses the reliability, ordered delivery, and record layer provided by QUIC.
-At a high level, there are two main interactions between the TLS and QUIC components:
- - -Figure 2 shows these interactions in more detail, with the QUIC packet protection being called out specially.
- - --+------------+ +------------+ -| |<---- Handshake Messages ----->| | -| |<- Validate 0-RTT parameters ->| | -| |<--------- 0-RTT Keys ---------| | -| QUIC |<------- Handshake Keys -------| TLS | -| |<--------- 1-RTT Keys ---------| | -| |<------- Handshake Done -------| | -+------------+ +------------+ - | ^ - | Protect | Protected - v | Packet -+------------+ -| QUIC | -| Packet | -| Protection | -+------------+ --
Figure 2: QUIC and TLS Interactions
-Unlike TLS over TCP, QUIC applications which want to send data do not send it through TLS “application_data” records. Rather, they send it as QUIC STREAM frames or other frame types which are then carried in QUIC packets.
-QUIC carries TLS handshake data in CRYPTO frames, each of which consists of a contiguous block of handshake data identified by an offset and length. Those frames are packaged into QUIC packets and encrypted under the current TLS encryption level. As with TLS over TCP, once TLS handshake data has been delivered to QUIC, it is QUIC’s responsibility to deliver it reliably. Each chunk of data that is produced by TLS is associated with the set of keys that TLS is currently using. If QUIC needs to retransmit that data, it MUST use the same keys even if TLS has already updated to newer keys.
-One important difference between TLS records (used with TCP) and QUIC CRYPTO frames is that in QUIC multiple frames may appear in the same QUIC packet as long as they are associated with the same encryption level. For instance, an implementation might bundle a Handshake message and an ACK for some Handshake data into the same packet.
-Some frames are prohibited in different encryption levels, others cannot be sent. The rules here generalize those of TLS, in that frames associated with establishing the connection can usually appear at any encryption level, whereas those associated with transferring data can only appear in the 0-RTT and 1-RTT encryption levels:
- - -Note that it is not possible to send the following frames in 0-RTT for various reasons: ACK, CRYPTO, NEW_TOKEN, PATH_RESPONSE, and RETIRE_CONNECTION_ID.
-Because packets could be reordered on the wire, QUIC uses the packet type to indicate which level a given packet was encrypted under, as shown in Table 1. When multiple packets of different encryption levels need to be sent, endpoints SHOULD use coalesced packets to send them in the same UDP datagram.
- - -| Packet Type | -Encryption Level | -PN Space | -
|---|---|---|
| Initial | -Initial secrets | -Initial | -
| 0-RTT Protected | -0-RTT | -0/1-RTT | -
| Handshake | -Handshake | -Handshake | -
| Retry | -N/A | -N/A | -
| Version Negotiation | -N/A | -N/A | -
| Short Header | -1-RTT | -0/1-RTT | -
Section 17 of [QUIC-TRANSPORT] shows how packets at the various encryption levels fit into the handshake process.
-As shown in Figure 2, the interface from QUIC to TLS consists of four primary functions:
- - -Additional functions might be needed to configure TLS.
-In this document, the TLS handshake is considered complete when the TLS stack has reported that the handshake is complete. This happens when the TLS stack has both sent a Finished message and verified the peer’s Finished message. Verifying the peer’s Finished provides the endpoints with an assurance that previous handshake messages have not been modified. Note that the handshake does not complete at both endpoints simultaneously. Consequently, any requirement that is based on the completion of the handshake depends on the perspective of the endpoint in question.
-In this document, the TLS handshake is considered confirmed at an endpoint when the following two conditions are met: the handshake is complete, and the endpoint has received an acknowledgment for a packet sent with 1-RTT keys. This second condition can be implemented by recording the lowest packet number sent with 1-RTT keys, and the highest value of the Largest Acknowledged field in any received 1-RTT ACK frame: once the latter is higher than or equal to the former, the handshake is confirmed.
-In order to drive the handshake, TLS depends on being able to send and receive handshake messages. There are two basic functions on this interface: one where QUIC requests handshake messages and one where QUIC provides handshake packets.
-Before starting the handshake QUIC provides TLS with the transport parameters (see Section 8.2) that it wishes to carry.
-A QUIC client starts TLS by requesting TLS handshake bytes from TLS. The client acquires handshake bytes before sending its first packet. A QUIC server starts the process by providing TLS with the client’s handshake bytes.
-At any time, the TLS stack at an endpoint will have a current sending encryption level and receiving encryption level. Each encryption level is associated with a different flow of bytes, which is reliably transmitted to the peer in CRYPTO frames. When TLS provides handshake bytes to be sent, they are appended to the current flow and any packet that includes the CRYPTO frame is protected using keys from the corresponding encryption level.
-QUIC takes the unprotected content of TLS handshake records as the content of CRYPTO frames. TLS record protection is not used by QUIC. QUIC assembles CRYPTO frames into QUIC packets, which are protected using QUIC packet protection.
-QUIC is only capable of conveying TLS handshake records in CRYPTO frames. TLS alerts are turned into QUIC CONNECTION_CLOSE error codes; see Section 4.9. TLS application data and other message types cannot be carried by QUIC at any encryption level and is an error if they are received from the TLS stack.
-When an endpoint receives a QUIC packet containing a CRYPTO frame from the network, it proceeds as follows:
- - -Each time that TLS is provided with new data, new handshake bytes are requested from TLS. TLS might not provide any bytes if the handshake messages it has received are incomplete or it has no data to send.
-Once the TLS handshake is complete, this is indicated to QUIC along with any final handshake bytes that TLS needs to send. TLS also provides QUIC with the transport parameters that the peer advertised during the handshake.
-Once the handshake is complete, TLS becomes passive. TLS can still receive data from its peer and respond in kind, but it will not need to send more data unless specifically requested - either by an application or QUIC. One reason to send data is that the server might wish to provide additional or updated session tickets to a client.
-When the handshake is complete, QUIC only needs to provide TLS with any data that arrives in CRYPTO streams. In the same way that is done during the handshake, new data is requested from TLS after providing received data.
-As keys for new encryption levels become available, TLS provides QUIC with those keys. Separately, as keys at a given encryption level become available to TLS, TLS indicates to QUIC that reading or writing keys at that encryption level are available. These events are not asynchronous; they always occur immediately after TLS is provided with new handshake bytes, or after TLS produces handshake bytes.
-TLS provides QUIC with three items as a new encryption level becomes available:
- - -These values are based on the values that TLS negotiates and are used by QUIC to generate packet and header protection keys (see Section 5 and Section 5.4).
-If 0-RTT is possible, it is ready after the client sends a TLS ClientHello message or the server receives that message. After providing a QUIC client with the first handshake bytes, the TLS stack might signal the change to 0-RTT keys. On the server, after receiving handshake bytes that contain a ClientHello message, a TLS server might signal that 0-RTT keys are available.
-Although TLS only uses one encryption level at a time, QUIC may use more than one level. For instance, after sending its Finished message (using a CRYPTO frame at the Handshake encryption level) an endpoint can send STREAM data (in 1-RTT encryption). If the Finished message is lost, the endpoint uses the Handshake encryption level to retransmit the lost message. Reordering or loss of packets can mean that QUIC will need to handle packets at multiple encryption levels. During the handshake, this means potentially handling packets at higher and lower encryption levels than the current encryption level used by TLS.
-In particular, server implementations need to be able to read packets at the Handshake encryption level at the same time as the 0-RTT encryption level. A client could interleave ACK frames that are protected with Handshake keys with 0-RTT data and the server needs to process those acknowledgments in order to detect lost Handshake packets.
-QUIC also needs access to keys that might not ordinarily be available to a TLS implementation. For instance, a client might need to acknowledge Handshake packets before it is ready to send CRYPTO frames at that encryption level. TLS therefore needs to provide keys to QUIC before it might produce them for its own use.
-Figure 3 summarizes the exchange between QUIC and TLS for both client and server. Each arrow is tagged with the encryption level used for that transmission.
- - --Client Server - -Get Handshake - Initial -------------> - Handshake Received -Install tx 0-RTT Keys - 0-RTT ---------------> - Get Handshake - <------------- Initial -Handshake Received -Install Handshake keys - Install rx 0-RTT keys - Install Handshake keys - Get Handshake - <----------- Handshake -Handshake Received - Install tx 1-RTT keys - <--------------- 1-RTT -Get Handshake -Handshake Complete - Handshake -----------> - Handshake Received - Install rx 1-RTT keys - Handshake Complete -Install 1-RTT keys - 1-RTT ---------------> - Get Handshake - <--------------- 1-RTT -Handshake Received --
Figure 3: Interaction Summary between QUIC and TLS
-Figure 3 shows the multiple packets that form a single “flight” of messages being processed individually, to show what incoming messages trigger different actions. New handshake messages are requested after all incoming packets have been processed. This process might vary depending on how QUIC implementations and the packets they receive are structured.
-This document describes how TLS 1.3 [TLS13] is used with QUIC.
-In practice, the TLS handshake will negotiate a version of TLS to use. This could result in a newer version of TLS than 1.3 being negotiated if both endpoints support that version. This is acceptable provided that the features of TLS 1.3 that are used by QUIC are supported by the newer version.
-A badly configured TLS implementation could negotiate TLS 1.2 or another older version of TLS. An endpoint MUST terminate the connection if a version of TLS older than 1.3 is negotiated.
-QUIC requires that the first Initial packet from a client contain an entire cryptographic handshake message, which for TLS is the ClientHello. Though a packet larger than 1200 bytes might be supported by the path, a client improves the likelihood that a packet is accepted if it ensures that the first ClientHello message is small enough to stay within this limit.
-QUIC packet and framing add at least 36 bytes of overhead to the ClientHello message. That overhead increases if the client chooses a connection ID without zero length. Overheads also do not include the token or a connection ID longer than 8 bytes, both of which might be required if a server sends a Retry packet.
-A typical TLS ClientHello can easily fit into a 1200 byte packet. However, in addition to the overheads added by QUIC, there are several variables that could cause this limit to be exceeded. Large session tickets, multiple or large key shares, and long lists of supported ciphers, signature algorithms, versions, QUIC transport parameters, and other negotiable parameters and extensions could cause this message to grow.
-For servers, in addition to connection IDs and tokens, the size of TLS session tickets can have an effect on a client’s ability to connect. Minimizing the size of these values increases the probability that they can be successfully used by a client.
-A client is not required to fit the ClientHello that it sends in response to a HelloRetryRequest message into a single UDP datagram.
-The TLS implementation does not need to ensure that the ClientHello is sufficiently large. QUIC PADDING frames are added to increase the size of the packet as necessary.
-The requirements for authentication depend on the application protocol that is in use. TLS provides server authentication and permits the server to request client authentication.
-A client MUST authenticate the identity of the server. This typically involves verification that the identity of the server is included in a certificate and that the certificate is issued by a trusted entity (see for example [RFC2818]).
-A server MAY request that the client authenticate during the handshake. A server MAY refuse a connection if the client is unable to authenticate when requested. The requirements for client authentication vary based on application protocol and deployment.
-A server MUST NOT use post-handshake client authentication (as defined in Section 4.6.2 of [TLS13]), because the multiplexing offered by QUIC prevents clients from correlating the certificate request with the application-level event that triggered it (see [HTTP2-TLS13]). More specifically, servers MUST NOT send post-handshake TLS CertificateRequest messages and clients MUST treat receipt of such messages as a connection error of type PROTOCOL_VIOLATION.
-To communicate their willingness to process 0-RTT data, servers send a NewSessionTicket message that contains the “early_data” extension with a max_early_data_size of 0xffffffff; the amount of data which the client can send in 0-RTT is controlled by the “initial_max_data” transport parameter supplied by the server. Servers MUST NOT send the “early_data” extension with a max_early_data_size set to any value other than 0xffffffff. A client MUST treat receipt of a NewSessionTicket that contains an “early_data” extension with any other value as a connection error of type PROTOCOL_VIOLATION.
-A client that wishes to send 0-RTT packets uses the “early_data” extension in the ClientHello message of a subsequent handshake (see Section 4.2.10 of [TLS13]). It then sends the application data in 0-RTT packets.
-A server accepts 0-RTT by sending an early_data extension in the EncryptedExtensions (see Section 4.2.10 of [TLS13]). The server then processes and acknowledges the 0-RTT packets that it receives.
-A server rejects 0-RTT by sending the EncryptedExtensions without an early_data extension. A server will always reject 0-RTT if it sends a TLS HelloRetryRequest. When rejecting 0-RTT, a server MUST NOT process any 0-RTT packets, even if it could. When 0-RTT was rejected, a client SHOULD treat receipt of an acknowledgement for a 0-RTT packet as a connection error of type PROTOCOL_VIOLATION, if it is able to detect the condition.
-When 0-RTT is rejected, all connection characteristics that the client assumed might be incorrect. This includes the choice of application protocol, transport parameters, and any application configuration. The client therefore MUST reset the state of all streams, including application state bound to those streams.
-A client MAY attempt to send 0-RTT again if it receives a Retry or Version Negotiation packet. These packets do not signify rejection of 0-RTT.
-When a server receives a ClientHello with the “early_data” extension, it has to decide whether to accept or reject early data from the client. Some of this decision is made by the TLS stack (e.g., checking that the cipher suite being resumed was included in the ClientHello; see Section 4.2.10 of [TLS13]). Even when the TLS stack has no reason to reject early data, the QUIC stack or the application protocol using QUIC might reject early data because the configuration of the transport or application associated with the resumed session is not compatible with the server’s current configuration.
-QUIC requires additional transport state to be associated with a 0-RTT session ticket. One common way to implement this is using stateless session tickets and storing this state in the session ticket. Application protocols that use QUIC might have similar requirements regarding associating or storing state. This associated state is used for deciding whether early data must be rejected. For example, HTTP/3 ([QUIC-HTTP]) settings determine how early data from the client is interpreted. Other applications using QUIC could have different requirements for determining whether to accept or reject early data.
-In TLS over TCP, the HelloRetryRequest feature (see Section 4.1.4 of [TLS13]) can be used to correct a client’s incorrect KeyShare extension as well as for a stateless round-trip check. From the perspective of QUIC, this just looks like additional messages carried in the Initial encryption level. Although it is in principle possible to use this feature for address verification in QUIC, QUIC implementations SHOULD instead use the Retry feature (see Section 8.1 of [QUIC-TRANSPORT]). HelloRetryRequest is still used to request key shares.
-If TLS experiences an error, it generates an appropriate alert as defined in Section 6 of [TLS13].
-A TLS alert is turned into a QUIC connection error by converting the one-byte alert description into a QUIC error code. The alert description is added to 0x100 to produce a QUIC error code from the range reserved for CRYPTO_ERROR. The resulting value is sent in a QUIC CONNECTION_CLOSE frame.
-The alert level of all TLS alerts is “fatal”; a TLS stack MUST NOT generate alerts at the “warning” level.
-After QUIC moves to a new encryption level, packet protection keys for previous encryption levels can be discarded. This occurs several times during the handshake, as well as when keys are updated; see Section 6.
-Packet protection keys are not discarded immediately when new keys are available. If packets from a lower encryption level contain CRYPTO frames, frames that retransmit that data MUST be sent at the same encryption level. Similarly, an endpoint generates acknowledgements for packets at the same encryption level as the packet being acknowledged. Thus, it is possible that keys for a lower encryption level are needed for a short time after keys for a newer encryption level are available.
-An endpoint cannot discard keys for a given encryption level unless it has both received and acknowledged all CRYPTO frames for that encryption level and when all CRYPTO frames for that encryption level have been acknowledged by its peer. However, this does not guarantee that no further packets will need to be received or sent at that encryption level because a peer might not have received all the acknowledgements necessary to reach the same state.
-Though an endpoint might retain older keys, new data MUST be sent at the highest currently-available encryption level. Only ACK frames and retransmissions of data in CRYPTO frames are sent at a previous encryption level. These packets MAY also include PADDING frames.
-Packets protected with Initial secrets (Section 5.2) are not authenticated, meaning that an attacker could spoof packets with the intent to disrupt a connection. To limit these attacks, Initial packet protection keys can be discarded more aggressively than other keys.
-The successful use of Handshake packets indicates that no more Initial packets need to be exchanged, as these keys can only be produced after receiving all CRYPTO frames from Initial packets. Thus, a client MUST discard Initial keys when it first sends a Handshake packet and a server MUST discard Initial keys when it first successfully processes a Handshake packet. Endpoints MUST NOT send Initial packets after this point.
-This results in abandoning loss recovery state for the Initial encryption level and ignoring any outstanding Initial packets.
-An endpoint MUST NOT discard its handshake keys until the TLS handshake is confirmed (Section 4.1.2). An endpoint SHOULD discard its handshake keys as soon as it has confirmed the handshake. Most application protocols will send data after the handshake, resulting in acknowledgements that allow both endpoints to discard their handshake keys promptly. Endpoints that do not have reason to send immediately after completing the handshake MAY send ack-eliciting frames, such as PING, which will cause the handshake to be confirmed when they are acknowledged.
-0-RTT and 1-RTT packets share the same packet number space, and clients do not send 0-RTT packets after sending a 1-RTT packet (Section 5.6).
-Therefore, a client SHOULD discard 0-RTT keys as soon as it installs 1-RTT keys, since they have no use after that moment.
-Additionally, a server MAY discard 0-RTT keys as soon as it receives a 1-RTT packet. However, due to packet reordering, a 0-RTT packet could arrive after a 1-RTT packet. Servers MAY temporarily retain 0-RTT keys to allow decrypting reordered packets without requiring their contents to be retransmitted with 1-RTT keys. After receiving a 1-RTT packet, servers MUST discard 0-RTT keys within a short time; the RECOMMENDED time period is three times the Probe Timeout (PTO, see [QUIC-RECOVERY]). A server MAY discard 0-RTT keys earlier if it determines that it has received all 0-RTT packets, which can be done by keeping track of missing packet numbers.
-As with TLS over TCP, QUIC protects packets with keys derived from the TLS handshake, using the AEAD algorithm negotiated by TLS.
-QUIC derives packet protection keys in the same way that TLS derives record protection keys.
-Each encryption level has separate secret values for protection of packets sent in each direction. These traffic secrets are derived by TLS (see Section 7.1 of [TLS13]) and are used by QUIC for all encryption levels except the Initial encryption level. The secrets for the Initial encryption level are computed based on the client’s initial Destination Connection ID, as described in Section 5.2.
-The keys used for packet protection are computed from the TLS secrets using the KDF provided by TLS. In TLS 1.3, the HKDF-Expand-Label function described in Section 7.1 of [TLS13] is used, using the hash function from the negotiated cipher suite. Other versions of TLS MUST provide a similar function in order to be used with QUIC.
-The current encryption level secret and the label “quic key” are input to the KDF to produce the AEAD key; the label “quic iv” is used to derive the IV; see Section 5.3. The header protection key uses the “quic hp” label; see Section 5.4. Using these labels provides key separation between QUIC and TLS; see Section 9.4.
-The KDF used for initial secrets is always the HKDF-Expand-Label function from TLS 1.3 (see Section 5.2).
-Initial packets are protected with a secret derived from the Destination Connection ID field from the client’s first Initial packet of the connection. Specifically:
--initial_salt = 0xc3eef712c72ebb5a11a7d2432bb46365bef9f502 -initial_secret = HKDF-Extract(initial_salt, - client_dst_connection_id) - -client_initial_secret = HKDF-Expand-Label(initial_secret, - "client in", "", - Hash.length) -server_initial_secret = HKDF-Expand-Label(initial_secret, - "server in", "", - Hash.length) --
The hash function for HKDF when deriving initial secrets and keys is SHA-256 [SHA].
-The connection ID used with HKDF-Expand-Label is the Destination Connection ID in the Initial packet sent by the client. This will be a randomly-selected value unless the client creates the Initial packet after receiving a Retry packet, where the Destination Connection ID is selected by the server.
-The value of initial_salt is a 20 byte sequence shown in the figure in hexadecimal notation. Future versions of QUIC SHOULD generate a new salt value, thus ensuring that the keys are different for each version of QUIC. This prevents a middlebox that only recognizes one version of QUIC from seeing or modifying the contents of packets from future versions.
-The HKDF-Expand-Label function defined in TLS 1.3 MUST be used for Initial packets even where the TLS versions offered do not include TLS 1.3.
-Appendix A contains test vectors for the initial packet encryption.
- - -The Authentication Encryption with Associated Data (AEAD) [AEAD] function used for QUIC packet protection is the AEAD that is negotiated for use with the TLS connection. For example, if TLS is using the TLS_AES_128_GCM_SHA256, the AEAD_AES_128_GCM function is used.
-Packets are protected prior to applying header protection (Section 5.4). The unprotected packet header is part of the associated data (A). When removing packet protection, an endpoint first removes the header protection.
-All QUIC packets other than Version Negotiation and Retry packets are protected with an AEAD algorithm [AEAD]. Prior to establishing a shared secret, packets are protected with AEAD_AES_128_GCM and a key derived from the Destination Connection ID in the client’s first Initial packet (see Section 5.2). This provides protection against off-path attackers and robustness against QUIC version unaware middleboxes, but not against on-path attackers.
-QUIC can use any of the ciphersuites defined in [TLS13] with the exception of TLS_AES_128_CCM_8_SHA256. A ciphersuite MUST NOT be negotiated unless a header protection scheme is defined for the ciphersuite. This document defines a header protection scheme for all ciphersuites defined in [TLS13] aside from TLS_AES_128_CCM_8_SHA256. These ciphersuites have a 16-byte authentication tag and produce an output 16 bytes larger than their input.
- - -The key and IV for the packet are computed as described in Section 5.1. The nonce, N, is formed by combining the packet protection IV with the packet number. The 62 bits of the reconstructed QUIC packet number in network byte order are left-padded with zeros to the size of the IV. The exclusive OR of the padded packet number and the IV forms the AEAD nonce.
-The associated data, A, for the AEAD is the contents of the QUIC header, starting from the flags byte in either the short or long header, up to and including the unprotected packet number.
-The input plaintext, P, for the AEAD is the payload of the QUIC packet, as described in [QUIC-TRANSPORT].
-The output ciphertext, C, of the AEAD is transmitted in place of P.
-Some AEAD functions have limits for how many packets can be encrypted under the same key and IV (see for example [AEBounds]). This might be lower than the packet number limit. An endpoint MUST initiate a key update (Section 6) prior to exceeding any limit set for the AEAD that is in use.
-Parts of QUIC packet headers, in particular the Packet Number field, are protected using a key that is derived separate to the packet protection key and IV. The key derived using the “quic hp” label is used to provide confidentiality protection for those fields that are not exposed to on-path elements.
-This protection applies to the least-significant bits of the first byte, plus the Packet Number field. The four least-significant bits of the first byte are protected for packets with long headers; the five least significant bits of the first byte are protected for packets with short headers. For both header forms, this covers the reserved bits and the Packet Number Length field; the Key Phase bit is also protected for packets with a short header.
-The same header protection key is used for the duration of the connection, with the value not changing after a key update (see Section 6). This allows header protection to be used to protect the key phase.
-This process does not apply to Retry or Version Negotiation packets, which do not contain a protected payload or any of the fields that are protected by this process.
-Header protection is applied after packet protection is applied (see Section 5.3). The ciphertext of the packet is sampled and used as input to an encryption algorithm. The algorithm used depends on the negotiated AEAD.
-The output of this algorithm is a 5 byte mask which is applied to the protected header fields using exclusive OR. The least significant bits of the first byte of the packet are masked by the least significant bits of the first mask byte, and the packet number is masked with the remaining bytes. Any unused bytes of mask that might result from a shorter packet number encoding are unused.
-Figure 4 shows a sample algorithm for applying header protection. Removing header protection only differs in the order in which the packet number length (pn_length) is determined.
- - --mask = header_protection(hp_key, sample) - -pn_length = (packet[0] & 0x03) + 1 -if (packet[0] & 0x80) == 0x80: - # Long header: 4 bits masked - packet[0] ^= mask[0] & 0x0f -else: - # Short header: 5 bits masked - packet[0] ^= mask[0] & 0x1f - -# pn_offset is the start of the Packet Number field. -packet[pn_offset:pn_offset+pn_length] ^= mask[1:1+pn_length] --
Figure 4: Header Protection Pseudocode
-Figure 5 shows the protected fields of long and short headers marked with an E. Figure 5 also shows the sampled fields.
- - --Long Header: -+-+-+-+-+-+-+-+-+ -|1|1|T T|E E E E| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version -> Length Fields ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - -Short Header: -+-+-+-+-+-+-+-+-+ -|0|1|S|E E E E E| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0/32..144) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - -Common Fields: -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -|E E E E E E E E E Packet Number (8/16/24/32) E E E E E E E E... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Protected Payload (8/16/24)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Sampled part of Protected Payload (128) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Protected Payload Remainder (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 5: Header Protection and Ciphertext Sample
-Before a TLS ciphersuite can be used with QUIC, a header protection algorithm MUST be specified for the AEAD used with that ciphersuite. This document defines algorithms for AEAD_AES_128_GCM, AEAD_AES_128_CCM, AEAD_AES_256_GCM (all AES AEADs are defined in [AEAD]), and AEAD_CHACHA20_POLY1305 [CHACHA]. Prior to TLS selecting a ciphersuite, AES header protection is used (Section 5.4.3), matching the AEAD_AES_128_GCM packet protection.
-The header protection algorithm uses both the header protection key and a sample of the ciphertext from the packet Payload field.
-The same number of bytes are always sampled, but an allowance needs to be made for the endpoint removing protection, which will not know the length of the Packet Number field. In sampling the packet ciphertext, the Packet Number field is assumed to be 4 bytes long (its maximum possible encoded length).
-An endpoint MUST discard packets that are not long enough to contain a complete sample.
-To ensure that sufficient data is available for sampling, packets are padded so that the combined lengths of the encoded packet number and protected payload is at least 4 bytes longer than the sample required for header protection. The ciphersuites defined in [TLS13] - other than TLS_AES_128_CCM_8_SHA256, for which a header protection scheme is not defined in this document - have 16-byte expansions and 16-byte header protection samples. This results in needing at least 3 bytes of frames in the unprotected payload if the packet number is encoded on a single byte, or 2 bytes of frames for a 2-byte packet number encoding.
-The sampled ciphertext for a packet with a short header can be determined by the following pseudocode:
--sample_offset = 1 + len(connection_id) + 4 - -sample = packet[sample_offset..sample_offset+sample_length] --
For example, for a packet with a short header, an 8 byte connection ID, and protected with AEAD_AES_128_GCM, the sample takes bytes 13 to 28 inclusive (using zero-based indexing).
-A packet with a long header is sampled in the same way, noting that multiple QUIC packets might be included in the same UDP datagram and that each one is handled separately.
--sample_offset = 7 + len(destination_connection_id) + - len(source_connection_id) + - len(payload_length) + 4 -if packet_type == Initial: - sample_offset += len(token_length) + - len(token) - -sample = packet[sample_offset..sample_offset+sample_length] --
This section defines the packet protection algorithm for AEAD_AES_128_GCM, AEAD_AES_128_CCM, and AEAD_AES_256_GCM. AEAD_AES_128_GCM and AEAD_AES_128_CCM use 128-bit AES [AES] in electronic code-book (ECB) mode. AEAD_AES_256_GCM uses 256-bit AES in ECB mode.
-This algorithm samples 16 bytes from the packet ciphertext. This value is used as the input to AES-ECB. In pseudocode:
--mask = AES-ECB(hp_key, sample) --
When AEAD_CHACHA20_POLY1305 is in use, header protection uses the raw ChaCha20 function as defined in Section 2.4 of [CHACHA]. This uses a 256-bit key and 16 bytes sampled from the packet protection output.
-The first 4 bytes of the sampled ciphertext are the block counter. A ChaCha20 implementation could take a 32-bit integer in place of a byte sequence, in which case the byte sequence is interpreted as a little-endian value.
-The remaining 12 bytes are used as the nonce. A ChaCha20 implementation might take an array of three 32-bit integers in place of a byte sequence, in which case the nonce bytes are interpreted as a sequence of 32-bit little-endian integers.
-The encryption mask is produced by invoking ChaCha20 to protect 5 zero bytes. In pseudocode:
-
-counter = sample[0..3]
-nonce = sample[4..15]
-mask = ChaCha20(hp_key, counter, nonce, {0,0,0,0,0})
-
-Once an endpoint successfully receives a packet with a given packet number, it MUST discard all packets in the same packet number space with higher packet numbers if they cannot be successfully unprotected with either the same key, or - if there is a key update - the next packet protection key (see Section 6). Similarly, a packet that appears to trigger a key update, but cannot be unprotected successfully MUST be discarded.
-Failure to unprotect a packet does not necessarily indicate the existence of a protocol error in a peer or an attack. The truncated packet number encoding used in QUIC can cause packet numbers to be decoded incorrectly if they are delayed significantly.
-If 0-RTT keys are available (see Section 4.5), the lack of replay protection means that restrictions on their use are necessary to avoid replay attacks on the protocol.
-A client MUST only use 0-RTT keys to protect data that is idempotent. A client MAY wish to apply additional restrictions on what data it sends prior to the completion of the TLS handshake. A client otherwise treats 0-RTT keys as equivalent to 1-RTT keys, except that it MUST NOT send ACKs with 0-RTT keys.
-A client that receives an indication that its 0-RTT data has been accepted by a server can send 0-RTT data until it receives all of the server’s handshake messages. A client SHOULD stop sending 0-RTT data if it receives an indication that 0-RTT data has been rejected.
-A server MUST NOT use 0-RTT keys to protect packets; it uses 1-RTT keys to protect acknowledgements of 0-RTT packets. A client MUST NOT attempt to decrypt 0-RTT packets it receives and instead MUST discard them.
-Once a client has installed 1-RTT keys, it MUST NOT send any more 0-RTT packets.
- - -Due to reordering and loss, protected packets might be received by an endpoint before the final TLS handshake messages are received. A client will be unable to decrypt 1-RTT packets from the server, whereas a server will be able to decrypt 1-RTT packets from the client.
-Even though 1-RTT keys are available to a server after receiving the first handshake messages from a client, it is missing assurances on the client state:
- - -Therefore, the server’s use of 1-RTT keys is limited before the handshake is complete. A server MUST NOT process data from incoming 1-RTT protected packets before the TLS handshake is complete. Because sending acknowledgments indicates that all frames in a packet have been processed, a server cannot send acknowledgments for 1-RTT packets until the TLS handshake is complete. Received packets protected with 1-RTT keys MAY be stored and later decrypted and used once the handshake is complete.
-The requirement for the server to wait for the client Finished message creates a dependency on that message being delivered. A client can avoid the potential for head-of-line blocking that this implies by sending its 1-RTT packets coalesced with a handshake packet containing a copy of the CRYPTO frame that carries the Finished message, until one of the handshake packets is acknowledged. This enables immediate server processing for those packets.
-A server could receive packets protected with 0-RTT keys prior to receiving a TLS ClientHello. The server MAY retain these packets for later decryption in anticipation of receiving a ClientHello.
-Once the handshake is confirmed, it is possible to update the keys. The KEY_PHASE bit in the short header is used to indicate whether key updates have occurred. The KEY_PHASE bit is initially set to 0 and then inverted with each key update.
-The KEY_PHASE bit allows a recipient to detect a change in keying material without necessarily needing to receive the first packet that triggered the change. An endpoint that notices a changed KEY_PHASE bit can update keys and decrypt the packet that contains the changed bit.
-This mechanism replaces the TLS KeyUpdate message. Endpoints MUST NOT send a TLS KeyUpdate message. Endpoints MUST treat the receipt of a TLS KeyUpdate message as a connection error of type 0x10a, equivalent to a fatal TLS alert of unexpected_message (see Section 4.9).
-An endpoint MUST NOT initiate the first key update until the handshake is confirmed (Section 4.1.2). An endpoint MUST NOT initiate a subsequent key update until it has received an acknowledgment for a packet sent at the current KEY_PHASE. This can be implemented by tracking the lowest packet number sent with each KEY_PHASE, and the highest acknowledged packet number in the 1-RTT space: once the latter is higher than or equal to the former, another key update can be initiated.
-Endpoints MAY limit the number of keys they retain to two sets for removing packet protection and one set for protecting packets. Older keys can be discarded. Updating keys multiple times rapidly can cause packets to be effectively lost if packets are significantly reordered. Therefore, an endpoint SHOULD NOT initiate a key update for some time after it has last updated keys; the RECOMMENDED time period is three times the PTO. This avoids valid reordered packets being dropped by the peer as a result of the peer discarding older keys.
-A receiving endpoint detects an update when the KEY_PHASE bit does not match what it is expecting. It creates a new secret (see Section 7.2 of [TLS13]) and the corresponding read key and IV using the KDF function provided by TLS. The header protection key is not updated.
-If the packet can be decrypted and authenticated using the updated key and IV, then the keys the endpoint uses for packet protection are also updated. The next packet sent by the endpoint MUST then use the new keys. Once an endpoint has sent a packet encrypted with a given key phase, it MUST NOT send a packet encrypted with an older key phase.
-An endpoint does not always need to send packets when it detects that its peer has updated keys. The next packet that it sends will simply use the new keys. If an endpoint detects a second update before it has sent any packets with updated keys, it indicates that its peer has updated keys twice without awaiting a reciprocal update. An endpoint MUST treat consecutive key updates as a fatal error and abort the connection.
-An endpoint SHOULD retain old keys for a period of no more than three times the PTO. After this period, old keys and their corresponding secrets SHOULD be discarded. Retaining keys allow endpoints to process packets that were sent with old keys and delayed in the network. Packets with higher packet numbers always use the updated keys and MUST NOT be decrypted with old keys.
-This ensures that once the handshake is complete, packets with the same KEY_PHASE will have the same packet protection keys, unless there are multiple key updates in a short time frame succession and significant packet reordering.
- - -- Initiating Peer Responding Peer - -@M QUIC Frames - New Keys -> @N -@N QUIC Frames - --------> - QUIC Frames @M - New Keys -> @N - QUIC Frames @N - <-------- --
Figure 6: Key Update
-A packet that triggers a key update could arrive after the receiving endpoint successfully processed a packet with a higher packet number. This is only possible if there is a key compromise and an attack, or if the peer is incorrectly reverting to use of old keys. Because the latter cannot be differentiated from an attack, an endpoint MUST immediately terminate the connection if it detects this condition.
-In deciding when to update keys, endpoints MUST NOT exceed the limits for use of specific keys, as described in Section 5.5 of [TLS13].
-Initial packets are not protected with a secret key, so they are subject to potential tampering by an attacker. QUIC provides protection against attackers that cannot read packets, but does not attempt to provide additional protection against attacks where the attacker can observe and inject packets. Some forms of tampering – such as modifying the TLS messages themselves – are detectable, but some – such as modifying ACKs – are not.
-For example, an attacker could inject a packet containing an ACK frame that makes it appear that a packet had not been received or to create a false impression of the state of the connection (e.g., by modifying the ACK Delay). Note that such a packet could cause a legitimate packet to be dropped as a duplicate. Implementations SHOULD use caution in relying on any data which is contained in Initial packets that is not otherwise authenticated.
-It is also possible for the attacker to tamper with data that is carried in Handshake packets, but because that tampering requires modifying TLS handshake messages, that tampering will cause the TLS handshake to fail.
-QUIC uses the TLS handshake for more than just negotiation of cryptographic parameters. The TLS handshake provides preliminary values for QUIC transport parameters and allows a server to perform return routability checks on clients.
-QUIC requires that the cryptographic handshake provide authenticated protocol negotiation. TLS uses Application Layer Protocol Negotiation (ALPN) [RFC7301] to select an application protocol. Unless another mechanism is used for agreeing on an application protocol, endpoints MUST use ALPN for this purpose. When using ALPN, endpoints MUST immediately close a connection (see Section 10.3 in [QUIC-TRANSPORT]) if an application protocol is not negotiated with a no_application_protocol TLS alert (QUIC error code 0x178, see Section 4.9). While [RFC7301] only specifies that servers use this alert, QUIC clients MUST also use it to terminate a connection when ALPN negotiation fails.
-An application-layer protocol MAY restrict the QUIC versions that it can operate over. Servers MUST select an application protocol compatible with the QUIC version that the client has selected. If the server cannot select a compatible combination of application protocol and QUIC version, it MUST abort the connection. A client MUST abort a connection if the server picks an application protocol incompatible with the protocol version being used.
-QUIC transport parameters are carried in a TLS extension. Different versions of QUIC might define a different format for this struct.
-Including transport parameters in the TLS handshake provides integrity protection for these values.
-
- enum {
- quic_transport_parameters(0xffa5), (65535)
- } ExtensionType;
-
-The extension_data field of the quic_transport_parameters extension contains a value that is defined by the version of QUIC that is in use. The quic_transport_parameters extension carries a TransportParameters struct when the version of QUIC defined in [QUIC-TRANSPORT] is used.
-The quic_transport_parameters extension is carried in the ClientHello and the EncryptedExtensions messages during the handshake. Endpoints MUST send the quic_transport_parameters extension; endpoints that receive ClientHello or EncryptedExtensions messages without the quic_transport_parameters extension MUST close the connection with an error of type 0x16d (equivalent to a fatal TLS missing_extension alert, see Section 4.9).
-While the transport parameters are technically available prior to the completion of the handshake, they cannot be fully trusted until the handshake completes, and reliance on them should be minimized. However, any tampering with the parameters will cause the handshake to fail.
-Endpoints MUST NOT send this extension in a TLS connection that does not use QUIC (such as the use of TLS with TCP defined in [TLS13]). A fatal unsupported_extension alert MUST be sent by an implementation that supports this extension if the extension is received when the transport is not QUIC.
-The TLS EndOfEarlyData message is not used with QUIC. QUIC does not rely on this message to mark the end of 0-RTT data or to signal the change to Handshake keys.
-Clients MUST NOT send the EndOfEarlyData message. A server MUST treat receipt of a CRYPTO frame in a 0-RTT packet as a connection error of type PROTOCOL_VIOLATION.
-As a result, EndOfEarlyData does not appear in the TLS handshake transcript.
-There are likely to be some real clangers here eventually, but the current set of issues is well captured in the relevant sections of the main text.
-Never assume that because it isn’t in the security considerations section it doesn’t affect security. Most of this document does.
-As described in Section 8 of [TLS13], use of TLS early data comes with an exposure to replay attack. The use of 0-RTT in QUIC is similarly vulnerable to replay attack.
-Endpoints MUST implement and use the replay protections described in [TLS13], however it is recognized that these protections are imperfect. Therefore, additional consideration of the risk of replay is needed.
-QUIC is not vulnerable to replay attack, except via the application protocol information it might carry. The management of QUIC protocol state based on the frame types defined in [QUIC-TRANSPORT] is not vulnerable to replay. Processing of QUIC frames is idempotent and cannot result in invalid connection states if frames are replayed, reordered or lost. QUIC connections do not produce effects that last beyond the lifetime of the connection, except for those produced by the application protocol that QUIC serves.
- - -A server that accepts 0-RTT on a connection incurs a higher cost than accepting a connection without 0-RTT. This includes higher processing and computation costs. Servers need to consider the probability of replay and all associated costs when accepting 0-RTT.
-Ultimately, the responsibility for managing the risks of replay attacks with 0-RTT lies with an application protocol. An application protocol that uses QUIC MUST describe how the protocol uses 0-RTT and the measures that are employed to protect against replay attack. An analysis of replay risk needs to consider all QUIC protocol features that carry application semantics.
-Disabling 0-RTT entirely is the most effective defense against replay attack.
-QUIC extensions MUST describe how replay attacks affect their operation, or prohibit their use in 0-RTT. Application protocols MUST either prohibit the use of extensions that carry application semantics in 0-RTT or provide replay mitigation strategies.
-A small ClientHello that results in a large block of handshake messages from a server can be used in packet reflection attacks to amplify the traffic generated by an attacker.
-QUIC includes three defenses against this attack. First, the packet containing a ClientHello MUST be padded to a minimum size. Second, if responding to an unverified source address, the server is forbidden to send more than three UDP datagrams in its first flight (see Section 8.1 of [QUIC-TRANSPORT]). Finally, because acknowledgements of Handshake packets are authenticated, a blind attacker cannot forge them. Put together, these defenses limit the level of amplification.
-[NAN] analyzes authenticated encryption algorithms which provide nonce privacy, referred to as “Hide Nonce” (HN) transforms. The general header protection construction in this document is one of those algorithms (HN1). Header protection uses the output of the packet protection AEAD to derive sample, and then encrypts the header field using a pseudorandom function (PRF) as follows:
--protected_field = field XOR PRF(hp_key, sample) --
The header protection variants in this document use a pseudorandom permutation (PRP) in place of a generic PRF. However, since all PRPs are also PRFs [IMC], these variants do not deviate from the HN1 construction.
-As hp_key is distinct from the packet protection key, it follows that header protection achieves AE2 security as defined in [NAN] and therefore guarantees privacy of field, the protected packet header. Future header protection variants based on this construction MUST use a PRF to ensure equivalent security guarantees.
-Use of the same key and ciphertext sample more than once risks compromising header protection. Protecting two different headers with the same key and ciphertext sample reveals the exclusive OR of the protected fields. Assuming that the AEAD acts as a PRF, if L bits are sampled, the odds of two ciphertext samples being identical approach 2^(-L/2), that is, the birthday bound. For the algorithms described in this document, that probability is one in 2^64.
- - -To prevent an attacker from modifying packet headers, the header is transitively authenticated using packet protection; the entire packet header is part of the authenticated additional data. Protected fields that are falsified or modified can only be detected once the packet protection is removed.
-An attacker could guess values for packet numbers and have an endpoint confirm guesses through timing side channels. Similarly, guesses for the packet number length can be trialed and exposed. If the recipient of a packet discards packets with duplicate packet numbers without attempting to remove packet protection they could reveal through timing side-channels that the packet number matches a received packet. For authentication to be free from side-channels, the entire process of header protection removal, packet number recovery, and packet protection removal MUST be applied together without timing and other side-channels.
-For the sending of packets, construction and protection of packet payloads and packet numbers MUST be free from side-channels that would reveal the packet number or its encoded size.
-In using TLS, the central key schedule of TLS is used. As a result of the TLS handshake messages being integrated into the calculation of secrets, the inclusion of the QUIC transport parameters extension ensures that handshake and 1-RTT keys are not the same as those that might be produced by a server running TLS over TCP. To avoid the possibility of cross-protocol key synchronization, additional measures are provided to improve key separation.
-The QUIC packet protection keys and IVs are derived using a different label than the equivalent keys in TLS.
-To preserve this separation, a new version of QUIC SHOULD define new labels for key derivation for packet protection key and IV, plus the header protection keys. This version of QUIC uses the string “quic”. Other versions can use a version-specific label in place of that string.
-The initial secrets use a key that is specific to the negotiated QUIC version. New QUIC versions SHOULD define a new salt value used in calculating initial secrets.
-This document does not create any new IANA registries, but it registers the values in the following registries:
- - -| [AEAD] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
| [AES] | -"Advanced encryption standard (AES)", National Institute of Standards and Technology report, DOI 10.6028/nist.fips.197, November 2001. | -
| [CHACHA] | --Nir, Y. and A. Langley, "ChaCha20 and Poly1305 for IETF Protocols", RFC 8439, DOI 10.17487/RFC8439, June 2018. | -
| [QUIC-RECOVERY] | --Iyengar, J. and I. Swett, "QUIC Loss Detection and Congestion Control", Internet-Draft draft-ietf-quic-recovery, October 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, October 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [SHA] | --Dang, Q., "Secure Hash Standard", National Institute of Standards and Technology report, DOI 10.6028/nist.fips.180-4, July 2015. | -
| [TLS-REGISTRIES] | --Salowey, J. and S. Turner, "IANA Registry Updates for TLS and DTLS", RFC 8447, DOI 10.17487/RFC8447, August 2018. | -
| [TLS13] | --Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018. | -
| [AEBounds] | --Luykx, A. and K. Paterson, "Limits on Authenticated Encryption Use in TLS", March 2016. | -
| [HTTP2-TLS13] | --Benjamin, D., "Using TLS 1.3 with HTTP/2", Internet-Draft draft-ietf-httpbis-http2-tls13-03, October 2019. | -
| [IMC] | --Katz, J. and Y. Lindell, "Introduction to Modern Cryptography, Second Edition", ISBN 978-1466570269, November 2014. | -
| [NAN] | --Bellare, M., Ng, R. and B. Tackmann, "Nonces Are Noticed: AEAD Revisited", Advances in Cryptology – CRYPTO 2019 pp. 235-265, DOI 10.1007/978-3-030-26948-7_9, 2019. | -
| [QUIC-HTTP] | --Bishop, M., "Hypertext Transfer Protocol (HTTP) over QUIC", Internet-Draft draft-ietf-quic-http, October 2019. | -
| [RFC2818] | --Rescorla, E., "HTTP Over TLS", RFC 2818, DOI 10.17487/RFC2818, May 2000. | -
| [RFC5280] | --Cooper, D., Santesson, S., Farrell, S., Boeyen, S., Housley, R. and W. Polk, "Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile", RFC 5280, DOI 10.17487/RFC5280, May 2008. | -
This section shows examples of packet protection for Initial packets so that implementations can be verified incrementally. These packets use an 8-byte client-chosen Destination Connection ID of 0x8394c8f03e515708. Values for both server and client packet protection are shown together with values in hexadecimal.
-The labels generated by the HKDF-Expand-Label function are:
- - -The initial secret is common:
--initial_secret = HKDF-Extract(initial_salt, cid) - = 524e374c6da8cf8b496f4bcb69678350 - 7aafee6198b202b4bc823ebf7514a423 --
The secrets for protecting client packets are:
--client_initial_secret - = HKDF-Expand-Label(initial_secret, "client in", _, 32) - = fda3953aecc040e48b34e27ef87de3a6 - 098ecf0e38b7e032c5c57bcbd5975b84 - -key = HKDF-Expand-Label(client_initial_secret, "quic key", _, 16) - = af7fd7efebd21878ff66811248983694 - -iv = HKDF-Expand-Label(client_initial_secret, "quic iv", _, 12) - = 8681359410a70bb9c92f0420 - -hp = HKDF-Expand-Label(client_initial_secret, "quic hp", _, 16) - = a980b8b4fb7d9fbc13e814c23164253d --
The secrets for protecting server packets are:
--server_initial_secret - = HKDF-Expand-Label(initial_secret, "server in", _, 32) - = 554366b81912ff90be41f17e80222130 - 90ab17d8149179bcadf222f29ff2ddd5 - -key = HKDF-Expand-Label(server_initial_secret, "quic key", _, 16) - = 5d51da9ee897a21b2659ccc7e5bfa577 - -iv = HKDF-Expand-Label(server_initial_secret, "quic iv", _, 12) - = 5e5ae651fd1e8495af13508b - -hp = HKDF-Expand-Label(server_initial_secret, "quic hp", _, 16) - = a8ed82e6664f865aedf6106943f95fb8 --
The client sends an Initial packet. The unprotected payload of this packet contains the following CRYPTO frame, plus enough PADDING frames to make a 1162 byte payload:
--060040c4010000c003036660261ff947 cea49cce6cfad687f457cf1b14531ba1 -4131a0e8f309a1d0b9c4000006130113 031302010000910000000b0009000006 -736572766572ff01000100000a001400 12001d00170018001901000101010201 -03010400230000003300260024001d00 204cfdfcd178b784bf328cae793b136f -2aedce005ff183d7bb14952072366470 37002b0003020304000d0020001e0403 -05030603020308040805080604010501 060102010402050206020202002d0002 -0101001c00024001 --
The unprotected header includes the connection ID and a 4 byte packet number encoding for a packet number of 2:
--c3ff000017088394c8f03e5157080000449e00000002 --
Protecting the payload produces output that is sampled for header protection. Because the header uses a 4 byte packet number encoding, the first 16 bytes of the protected payload is sampled, then applied to the header:
--sample = 535064a4268a0d9d7b1c9d250ae35516 - -mask = AES-ECB(hp, sample)[0..4] - = 833b343aaa - -header[0] ^= mask[0] & 0x0f - = c0 -header[17..20] ^= mask[1..4] - = 3b343aa8 -header = c0ff000017088394c8f03e5157080000449e3b343aa8 --
The resulting protected packet is:
--c0ff000017088394c8f03e5157080000 449e3b343aa8535064a4268a0d9d7b1c -9d250ae355162276e9b1e3011ef6bbc0 ab48ad5bcc2681e953857ca62becd752 -4daac473e68d7405fbba4e9ee616c870 38bdbe908c06d9605d9ac49030359eec -b1d05a14e117db8cede2bb09d0dbbfee 271cb374d8f10abec82d0f59a1dee29f -e95638ed8dd41da07487468791b719c5 5c46968eb3b54680037102a28e53dc1d -12903db0af5821794b41c4a93357fa59 ce69cfe7f6bdfa629eef78616447e1d6 -11c4baf71bf33febcb03137c2c75d253 17d3e13b684370f668411c0f00304b50 -1c8fd422bd9b9ad81d643b20da89ca05 25d24d2b142041cae0af205092e43008 -0cd8559ea4c5c6e4fa3f66082b7d303e 52ce0162baa958532b0bbc2bc785681f -cf37485dff6595e01e739c8ac9efba31 b985d5f656cc092432d781db95221724 -87641c4d3ab8ece01e39bc85b1543661 4775a98ba8fa12d46f9b35e2a55eb72d -7f85181a366663387ddc20551807e007 673bd7e26bf9b29b5ab10a1ca87cbb7a -d97e99eb66959c2a9bc3cbde4707ff77 20b110fa95354674e395812e47a0ae53 -b464dcb2d1f345df360dc227270c7506 76f6724eb479f0d2fbb6124429990457 -ac6c9167f40aab739998f38b9eccb24f d47c8410131bf65a52af841275d5b3d1 -880b197df2b5dea3e6de56ebce3ffb6e 9277a82082f8d9677a6767089b671ebd -244c214f0bde95c2beb02cd1172d58bd f39dce56ff68eb35ab39b49b4eac7c81 -5ea60451d6e6ab82119118df02a58684 4a9ffe162ba006d0669ef57668cab38b -62f71a2523a084852cd1d079b3658dc2 f3e87949b550bab3e177cfc49ed190df -f0630e43077c30de8f6ae081537f1e83 da537da980afa668e7b7fb25301cf741 -524be3c49884b42821f17552fbd1931a 813017b6b6590a41ea18b6ba49cd48a4 -40bd9a3346a7623fb4ba34a3ee571e3c 731f35a7a3cf25b551a680fa68763507 -b7fde3aaf023c50b9d22da6876ba337e b5e9dd9ec3daf970242b6c5aab3aa4b2 -96ad8b9f6832f686ef70fa938b31b4e5 ddd7364442d3ea72e73d668fb0937796 -f462923a81a47e1cee7426ff6d922126 9b5a62ec03d6ec94d12606cb485560ba -b574816009e96504249385bb61a819be 04f62c2066214d8360a2022beb316240 -b6c7d78bbe56c13082e0ca272661210a bf020bf3b5783f1426436cf9ff418405 -93a5d0638d32fc51c5c65ff291a3a7a5 2fd6775e623a4439cc08dd25582febc9 -44ef92d8dbd329c91de3e9c9582e41f1 7f3d186f104ad3f90995116c682a2a14 -a3b4b1f547c335f0be710fc9fc03e0e5 87b8cda31ce65b969878a4ad4283e6d5 -b0373f43da86e9e0ffe1ae0fddd35162 55bd74566f36a38703d5f34249ded1f6 -6b3d9b45b9af2ccfefe984e13376b1b2 c6404aa48c8026132343da3f3a33659e -c1b3e95080540b28b7f3fcd35fa5d843 b579a84c089121a60d8c1754915c344e -eaf45a9bf27dc0c1e784161691220913 13eb0e87555abd706626e557fc36a04f -cd191a58829104d6075c5594f627ca50 6bf181daec940f4a4f3af0074eee89da -acde6758312622d4fa675b39f728e062 d2bee680d8f41a597c262648bb18bcfc -13c8b3d97b1a77b2ac3af745d61a34cc 4709865bac824a94bb19058015e4e42d -c9be6c7803567321829dd85853396269 --
The server sends the following payload in response, including an ACK frame, a CRYPTO frame, and no PADDING frames:
--0d0000000018410a020000560303eefc e7f7b37ba1d1632e96677825ddf73988 -cfc79825df566dc5430b9a045a120013 0100002e00330024001d00209d3c940d -89690b84d08a60993c144eca684d1081 287c834d5311bcf32bb9da1a002b0002 -0304 --
The header from the server includes a new connection ID and a 2-byte packet number encoding for a packet number of 1:
--c1ff0000170008f067a5502a4262b50040740001 --
As a result, after protection, the header protection sample is taken starting from the third protected octet:
--sample = 7002596f99ae67abf65a5852f54f58c3 -mask = 38168a0c25 -header = c9ff0000170008f067a5502a4262b5004074168b --
The final protected packet is then:
--c9ff0000170008f067a5502a4262b500 4074168bf22b7002596f99ae67abf65a -5852f54f58c37c808682e2e40492d8a3 899fb04fc0afe9aabc8767b18a0aa493 -537426373b48d502214dd856d63b78ce e37bc664b3fe86d487ac7a77c53038a3 -cd32f0b5004d9f5754c4f7f2d1f35cf3 f7116351c92b9cf9bb6d091ddfc8b32d -432348a2c413 --
Issue and pull request numbers are listed with a leading octothorp.
-No significant changes.
-No significant changes.
-This document has benefited from input from Dragana Damjanovic, Christian Huitema, Jana Iyengar, Adam Langley, Roberto Peon, Eric Rescorla, Ian Swett, and many others.
-Ryan Hamilton was originally an author of this specification.
-| QUIC | -J. Iyengar, Ed. | -
| Internet-Draft | -Fastly | -
| Intended status: Standards Track | -M. Thomson, Ed. | -
| Expires: April 24, 2020 | -Mozilla | -
| - | October 22, 2019 | -
QUIC: A UDP-Based Multiplexed and Secure Transport
- draft-ietf-quic-transport-latest
This document defines the core of the QUIC transport protocol. Accompanying documents describe QUIC’s loss detection and congestion control and the use of TLS for key negotiation.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at <https://mailarchive.ietf.org/arch/search/?email_list=quic>.
-Working Group information can be found at <https://github.com/quicwg>; source code and issues list for this draft can be found at <https://github.com/quicwg/base-drafts/labels/-transport>.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on April 24, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -QUIC is a multiplexed and secure general-purpose transport protocol that provides:
- - -QUIC uses UDP as a substrate to avoid requiring changes to legacy client operating systems and middleboxes. QUIC authenticates all of its headers and encrypts most of the data it exchanges, including its signaling, to avoid incurring a dependency on middleboxes.
-This document describes the core QUIC protocol and is structured as follows.
- - -Accompanying documents describe QUIC’s loss detection and congestion control [QUIC-RECOVERY], and the use of TLS for key negotiation [QUIC-TLS].
-This document defines QUIC version 1, which conforms to the protocol invariants in [QUIC-INVARIANTS].
-The keywords “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Commonly used terms in the document are described below.
- - -Packet and frame diagrams in this document use the format described in Section 3.1 of [RFC2360], with the following additional conventions:
- - -Streams in QUIC provide a lightweight, ordered byte-stream abstraction to an application. Streams can be unidirectional or bidirectional. An alternative view of QUIC unidirectional streams is a “message” abstraction of practically unlimited length.
-Streams can be created by sending data. Other processes associated with stream management - ending, cancelling, and managing flow control - are all designed to impose minimal overheads. For instance, a single STREAM frame (Section 19.8) can open, carry data for, and close a stream. Streams can also be long-lived and can last the entire duration of a connection.
-Streams can be created by either endpoint, can concurrently send data interleaved with other streams, and can be cancelled. QUIC does not provide any means of ensuring ordering between bytes on different streams.
-QUIC allows for an arbitrary number of streams to operate concurrently and for an arbitrary amount of data to be sent on any stream, subject to flow control constraints (see Section 4) and stream limits.
-Streams can be unidirectional or bidirectional. Unidirectional streams carry data in one direction: from the initiator of the stream to its peer. Bidirectional streams allow for data to be sent in both directions.
-Streams are identified within a connection by a numeric value, referred to as the stream ID. A stream ID is a 62-bit integer (0 to 2^62-1) that is unique for all streams on a connection. Stream IDs are encoded as variable-length integers (see Section 16). A QUIC endpoint MUST NOT reuse a stream ID within a connection.
-The least significant bit (0x1) of the stream ID identifies the initiator of the stream. Client-initiated streams have even-numbered stream IDs (with the bit set to 0), and server-initiated streams have odd-numbered stream IDs (with the bit set to 1).
-The second least significant bit (0x2) of the stream ID distinguishes between bidirectional streams (with the bit set to 0) and unidirectional streams (with the bit set to 1).
-The least significant two bits from a stream ID therefore identify a stream as one of four types, as summarized in Table 1.
- - -| Bits | -Stream Type | -
|---|---|
| 0x0 | -Client-Initiated, Bidirectional | -
| 0x1 | -Server-Initiated, Bidirectional | -
| 0x2 | -Client-Initiated, Unidirectional | -
| 0x3 | -Server-Initiated, Unidirectional | -
Within each type, streams are created with numerically increasing stream IDs. A stream ID that is used out of order results in all streams of that type with lower-numbered stream IDs also being opened.
-The first bidirectional stream opened by the client has a stream ID of 0.
-STREAM frames (Section 19.8) encapsulate data sent by an application. An endpoint uses the Stream ID and Offset fields in STREAM frames to place data in order.
-Endpoints MUST be able to deliver stream data to an application as an ordered byte-stream. Delivering an ordered byte-stream requires that an endpoint buffer any data that is received out of order, up to the advertised flow control limit.
-QUIC makes no specific allowances for delivery of stream data out of order. However, implementations MAY choose to offer the ability to deliver data out of order to a receiving application.
-An endpoint could receive data for a stream at the same stream offset multiple times. Data that has already been received can be discarded. The data at a given offset MUST NOT change if it is sent multiple times; an endpoint MAY treat receipt of different data at the same offset within a stream as a connection error of type PROTOCOL_VIOLATION.
-Streams are an ordered byte-stream abstraction with no other structure visible to QUIC. STREAM frame boundaries are not expected to be preserved when data is transmitted, retransmitted after packet loss, or delivered to the application at a receiver.
-An endpoint MUST NOT send data on any stream without ensuring that it is within the flow control limits set by its peer. Flow control is described in detail in Section 4.
-Stream multiplexing can have a significant effect on application performance if resources allocated to streams are correctly prioritized.
-QUIC does not provide a mechanism for exchanging prioritization information. Instead, it relies on receiving priority information from the application that uses QUIC.
-A QUIC implementation SHOULD provide ways in which an application can indicate the relative priority of streams. When deciding which streams to dedicate resources to, the implementation SHOULD use the information provided by the application.
-There are certain operations which an application MUST be able to perform when interacting with QUIC streams. This document does not specify an API, but any implementation of this version of QUIC MUST expose the ability to perform the operations described in this section on a QUIC stream.
-On the sending part of a stream, application protocols need to be able to:
- - -On the receiving part of a stream, application protocols need to be able to:
- - -Applications also need to be informed of state changes on streams, including when the peer has opened or reset a stream, when a peer aborts reading on a stream, when new data is available, and when data can or cannot be written to the stream due to flow control.
-This section describes streams in terms of their send or receive components. Two state machines are described: one for the streams on which an endpoint transmits data (Section 3.1), and another for streams on which an endpoint receives data (Section 3.2).
-Unidirectional streams use the applicable state machine directly. Bidirectional streams use both state machines. For the most part, the use of these state machines is the same whether the stream is unidirectional or bidirectional. The conditions for opening a stream are slightly more complex for a bidirectional stream because the opening of either send or receive sides causes the stream to open in both directions.
-An endpoint MUST open streams of the same type in increasing order of stream ID.
- - -Figure 1 shows the states for the part of a stream that sends data to a peer.
- - -- o - | Create Stream (Sending) - | Peer Creates Bidirectional Stream - v - +-------+ - | Ready | Send RESET_STREAM - | |-----------------------. - +-------+ | - | | - | Send STREAM / | - | STREAM_DATA_BLOCKED | - | | - | Peer Creates | - | Bidirectional Stream | - v | - +-------+ | - | Send | Send RESET_STREAM | - | |---------------------->| - +-------+ | - | | - | Send STREAM + FIN | - v v - +-------+ +-------+ - | Data | Send RESET_STREAM | Reset | - | Sent |------------------>| Sent | - +-------+ +-------+ - | | - | Recv All ACKs | Recv ACK - v v - +-------+ +-------+ - | Data | | Reset | - | Recvd | | Recvd | - +-------+ +-------+ --
Figure 1: States for Sending Parts of Streams
-The sending part of stream that the endpoint initiates (types 0 and 2 for clients, 1 and 3 for servers) is opened by the application. The “Ready” state represents a newly created stream that is able to accept data from the application. Stream data might be buffered in this state in preparation for sending.
-Sending the first STREAM or STREAM_DATA_BLOCKED frame causes a sending part of a stream to enter the “Send” state. An implementation might choose to defer allocating a stream ID to a stream until it sends the first STREAM frame and enters this state, which can allow for better stream prioritization.
-The sending part of a bidirectional stream initiated by a peer (type 0 for a server, type 1 for a client) enters the “Ready” state then immediately transitions to the “Send” state if the receiving part enters the “Recv” state (Section 3.2).
-In the “Send” state, an endpoint transmits - and retransmits as necessary - stream data in STREAM frames. The endpoint respects the flow control limits set by its peer, and continues to accept and process MAX_STREAM_DATA frames. An endpoint in the “Send” state generates STREAM_DATA_BLOCKED frames if it is blocked from sending by stream or connection flow control limits Section 4.1.
-After the application indicates that all stream data has been sent and a STREAM frame containing the FIN bit is sent, the sending part of the stream enters the “Data Sent” state. From this state, the endpoint only retransmits stream data as necessary. The endpoint does not need to check flow control limits or send STREAM_DATA_BLOCKED frames for a stream in this state. MAX_STREAM_DATA frames might be received until the peer receives the final stream offset. The endpoint can safely ignore any MAX_STREAM_DATA frames it receives from its peer for a stream in this state.
-Once all stream data has been successfully acknowledged, the sending part of the stream enters the “Data Recvd” state, which is a terminal state.
-From any of the “Ready”, “Send”, or “Data Sent” states, an application can signal that it wishes to abandon transmission of stream data. Alternatively, an endpoint might receive a STOP_SENDING frame from its peer. In either case, the endpoint sends a RESET_STREAM frame, which causes the stream to enter the “Reset Sent” state.
-An endpoint MAY send a RESET_STREAM as the first frame that mentions a stream; this causes the sending part of that stream to open and then immediately transition to the “Reset Sent” state.
-Once a packet containing a RESET_STREAM has been acknowledged, the sending part of the stream enters the “Reset Recvd” state, which is a terminal state.
-Figure 2 shows the states for the part of a stream that receives data from a peer. The states for a receiving part of a stream mirror only some of the states of the sending part of the stream at the peer. The receiving part of a stream does not track states on the sending part that cannot be observed, such as the “Ready” state. Instead, the receiving part of a stream tracks the delivery of data to the application, some of which cannot be observed by the sender.
- - -- o - | Recv STREAM / STREAM_DATA_BLOCKED / RESET_STREAM - | Create Bidirectional Stream (Sending) - | Recv MAX_STREAM_DATA / STOP_SENDING (Bidirectional) - | Create Higher-Numbered Stream - v - +-------+ - | Recv | Recv RESET_STREAM - | |-----------------------. - +-------+ | - | | - | Recv STREAM + FIN | - v | - +-------+ | - | Size | Recv RESET_STREAM | - | Known |---------------------->| - +-------+ | - | | - | Recv All Data | - v v - +-------+ Recv RESET_STREAM +-------+ - | Data |--- (optional) --->| Reset | - | Recvd | Recv All Data | Recvd | - +-------+<-- (optional) ----+-------+ - | | - | App Read All Data | App Read RST - v v - +-------+ +-------+ - | Data | | Reset | - | Read | | Read | - +-------+ +-------+ --
Figure 2: States for Receiving Parts of Streams
-The receiving part of a stream initiated by a peer (types 1 and 3 for a client, or 0 and 2 for a server) is created when the first STREAM, STREAM_DATA_BLOCKED, or RESET_STREAM is received for that stream. For bidirectional streams initiated by a peer, receipt of a MAX_STREAM_DATA or STOP_SENDING frame for the sending part of the stream also creates the receiving part. The initial state for the receiving part of stream is “Recv”.
-The receiving part of a stream enters the “Recv” state when the sending part of a bidirectional stream initiated by the endpoint (type 0 for a client, type 1 for a server) enters the “Ready” state.
-An endpoint opens a bidirectional stream when a MAX_STREAM_DATA or STOP_SENDING frame is received from the peer for that stream. Receiving a MAX_STREAM_DATA frame for an unopened stream indicates that the remote peer has opened the stream and is providing flow control credit. Receiving a STOP_SENDING frame for an unopened stream indicates that the remote peer no longer wishes to receive data on this stream. Either frame might arrive before a STREAM or STREAM_DATA_BLOCKED frame if packets are lost or reordered.
-Before a stream is created, all streams of the same type with lower-numbered stream IDs MUST be created. This ensures that the creation order for streams is consistent on both endpoints.
-In the “Recv” state, the endpoint receives STREAM and STREAM_DATA_BLOCKED frames. Incoming data is buffered and can be reassembled into the correct order for delivery to the application. As data is consumed by the application and buffer space becomes available, the endpoint sends MAX_STREAM_DATA frames to allow the peer to send more data.
-When a STREAM frame with a FIN bit is received, the final size of the stream is known (see Section 4.4). The receiving part of the stream then enters the “Size Known” state. In this state, the endpoint no longer needs to send MAX_STREAM_DATA frames, it only receives any retransmissions of stream data.
-Once all data for the stream has been received, the receiving part enters the “Data Recvd” state. This might happen as a result of receiving the same STREAM frame that causes the transition to “Size Known”. After all data has been received, any STREAM or STREAM_DATA_BLOCKED frames for the stream can be discarded.
-The “Data Recvd” state persists until stream data has been delivered to the application. Once stream data has been delivered, the stream enters the “Data Read” state, which is a terminal state.
-Receiving a RESET_STREAM frame in the “Recv” or “Size Known” states causes the stream to enter the “Reset Recvd” state. This might cause the delivery of stream data to the application to be interrupted.
-It is possible that all stream data is received when a RESET_STREAM is received (that is, from the “Data Recvd” state). Similarly, it is possible for remaining stream data to arrive after receiving a RESET_STREAM frame (the “Reset Recvd” state). An implementation is free to manage this situation as it chooses.
-Sending RESET_STREAM means that an endpoint cannot guarantee delivery of stream data; however there is no requirement that stream data not be delivered if a RESET_STREAM is received. An implementation MAY interrupt delivery of stream data, discard any data that was not consumed, and signal the receipt of the RESET_STREAM. A RESET_STREAM signal might be suppressed or withheld if stream data is completely received and is buffered to be read by the application. If the RESET_STREAM is suppressed, the receiving part of the stream remains in “Data Recvd”.
-Once the application receives the signal indicating that the stream was reset, the receiving part of the stream transitions to the “Reset Read” state, which is a terminal state.
-The sender of a stream sends just three frame types that affect the state of a stream at either sender or receiver: STREAM (Section 19.8), STREAM_DATA_BLOCKED (Section 19.13), and RESET_STREAM (Section 19.4).
-A sender MUST NOT send any of these frames from a terminal state (“Data Recvd” or “Reset Recvd”). A sender MUST NOT send STREAM or STREAM_DATA_BLOCKED after sending a RESET_STREAM; that is, in the terminal states and in the “Reset Sent” state. A receiver could receive any of these three frames in any state, due to the possibility of delayed delivery of packets carrying them.
-The receiver of a stream sends MAX_STREAM_DATA (Section 19.10) and STOP_SENDING frames (Section 19.5).
-The receiver only sends MAX_STREAM_DATA in the “Recv” state. A receiver can send STOP_SENDING in any state where it has not received a RESET_STREAM frame; that is states other than “Reset Recvd” or “Reset Read”. However there is little value in sending a STOP_SENDING frame in the “Data Recvd” state, since all stream data has been received. A sender could receive either of these two frames in any state as a result of delayed delivery of packets.
-A bidirectional stream is composed of sending and receiving parts. Implementations may represent states of the bidirectional stream as composites of sending and receiving stream states. The simplest model presents the stream as “open” when either sending or receiving parts are in a non-terminal state and “closed” when both sending and receiving streams are in terminal states.
-Table 2 shows a more complex mapping of bidirectional stream states that loosely correspond to the stream states in HTTP/2 [HTTP2]. This shows that multiple states on sending or receiving parts of streams are mapped to the same composite state. Note that this is just one possibility for such a mapping; this mapping requires that data is acknowledged before the transition to a “closed” or “half-closed” state.
- - -| Sending Part | -Receiving Part | -Composite State | -
|---|---|---|
| No Stream/Ready | -No Stream/Recv *1 | -idle | -
| Ready/Send/Data Sent | -Recv/Size Known | -open | -
| Ready/Send/Data Sent | -Data Recvd/Data Read | -half-closed (remote) | -
| Ready/Send/Data Sent | -Reset Recvd/Reset Read | -half-closed (remote) | -
| Data Recvd | -Recv/Size Known | -half-closed (local) | -
| Reset Sent/Reset Recvd | -Recv/Size Known | -half-closed (local) | -
| Reset Sent/Reset Recvd | -Data Recvd/Data Read | -closed | -
| Reset Sent/Reset Recvd | -Reset Recvd/Reset Read | -closed | -
| Data Recvd | -Data Recvd/Data Read | -closed | -
| Data Recvd | -Reset Recvd/Reset Read | -closed | -
If an application is no longer interested in the data it is receiving on a stream, it can abort reading the stream and specify an application error code.
-If the stream is in the “Recv” or “Size Known” states, the transport SHOULD signal this by sending a STOP_SENDING frame to prompt closure of the stream in the opposite direction. This typically indicates that the receiving application is no longer reading data it receives from the stream, but it is not a guarantee that incoming data will be ignored.
-STREAM frames received after sending STOP_SENDING are still counted toward connection and stream flow control, even though these frames can be discarded upon receipt.
-A STOP_SENDING frame requests that the receiving endpoint send a RESET_STREAM frame. An endpoint that receives a STOP_SENDING frame MUST send a RESET_STREAM frame if the stream is in the Ready or Send state. If the stream is in the Data Sent state and any outstanding data is declared lost, an endpoint SHOULD send a RESET_STREAM frame in lieu of a retransmission.
-An endpoint SHOULD copy the error code from the STOP_SENDING frame to the RESET_STREAM frame it sends, but MAY use any application error code. The endpoint that sends a STOP_SENDING frame MAY ignore the error code carried in any RESET_STREAM frame it receives.
-If the STOP_SENDING frame is received on a stream that is already in the “Data Sent” state, an endpoint that wishes to cease retransmission of previously-sent STREAM frames on that stream MUST first send a RESET_STREAM frame.
-STOP_SENDING SHOULD only be sent for a stream that has not been reset by the peer. STOP_SENDING is most useful for streams in the “Recv” or “Size Known” states.
-An endpoint is expected to send another STOP_SENDING frame if a packet containing a previous STOP_SENDING is lost. However, once either all stream data or a RESET_STREAM frame has been received for the stream - that is, the stream is in any state other than “Recv” or “Size Known” - sending a STOP_SENDING frame is unnecessary.
-An endpoint that wishes to terminate both directions of a bidirectional stream can terminate one direction by sending a RESET_STREAM, and it can encourage prompt termination in the opposite direction by sending a STOP_SENDING frame.
-It is necessary to limit the amount of data that a receiver could buffer, to prevent a fast sender from overwhelming a slow receiver, or to prevent a malicious sender from consuming a large amount of memory at a receiver. To enable a receiver to limit memory commitment to a connection and to apply back pressure on the sender, streams are flow controlled both individually and as an aggregate. A QUIC receiver controls the maximum amount of data the sender can send on a stream at any time, as described in Section 4.1 and Section 4.2
-Similarly, to limit concurrency within a connection, a QUIC endpoint controls the maximum cumulative number of streams that its peer can initiate, as described in Section 4.5.
-Data sent in CRYPTO frames is not flow controlled in the same way as stream data. QUIC relies on the cryptographic protocol implementation to avoid excessive buffering of data; see [QUIC-TLS]. The implementation SHOULD provide an interface to QUIC to tell it about its buffering limits so that there is not excessive buffering at multiple layers.
-QUIC employs a credit-based flow-control scheme similar to that in HTTP/2 [HTTP2], where a receiver advertises the number of bytes it is prepared to receive on a given stream and for the entire connection. This leads to two levels of data flow control in QUIC:
- - -A receiver sets initial credits for all streams by sending transport parameters during the handshake (Section 7.3). A receiver sends MAX_STREAM_DATA (Section 19.10) or MAX_DATA (Section 19.9) frames to the sender to advertise additional credit.
-A receiver advertises credit for a stream by sending a MAX_STREAM_DATA frame with the Stream ID field set appropriately. A MAX_STREAM_DATA frame indicates the maximum absolute byte offset of a stream. A receiver could use the current offset of data consumed to determine the flow control offset to be advertised. A receiver MAY send MAX_STREAM_DATA frames in multiple packets in order to make sure that the sender receives an update before running out of flow control credit, even if one of the packets is lost.
-A receiver advertises credit for a connection by sending a MAX_DATA frame, which indicates the maximum of the sum of the absolute byte offsets of all streams. A receiver maintains a cumulative sum of bytes received on all streams, which is used to check for flow control violations. A receiver might use a sum of bytes consumed on all streams to determine the maximum data limit to be advertised.
-A receiver can advertise a larger offset by sending MAX_STREAM_DATA or MAX_DATA frames. Once a receiver advertises an offset, it MAY advertise a smaller offset, but this has no effect.
-A receiver MUST close the connection with a FLOW_CONTROL_ERROR error (Section 11) if the sender violates the advertised connection or stream data limits.
-A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do not increase flow control limits.
-If a sender runs out of flow control credit, it will be unable to send new data and is considered blocked. A sender SHOULD send a STREAM_DATA_BLOCKED or DATA_BLOCKED frame to indicate it has data to write but is blocked by flow control limits. These frames are expected to be sent infrequently in common cases, but they are considered useful for debugging and monitoring purposes.
-A sender SHOULD NOT send multiple STREAM_DATA_BLOCKED or DATA_BLOCKED frames for the same data limit, unless the original frame is determined to be lost. Another STREAM_DATA_BLOCKED or DATA_BLOCKED frame can be sent after the data limit is increased.
-This document leaves when and how many bytes to advertise in a MAX_STREAM_DATA or MAX_DATA frame to implementations, but offers a few considerations. These frames contribute to connection overhead. Therefore frequently sending frames with small changes is undesirable. At the same time, larger increments to limits are necessary to avoid blocking if updates are less frequent, requiring larger resource commitments at the receiver. Thus there is a trade-off between resource commitment and overhead when determining how large a limit is advertised.
-A receiver can use an autotuning mechanism to tune the frequency and amount of advertised additional credit based on a round-trip time estimate and the rate at which the receiving application consumes data, similar to common TCP implementations. As an optimization, sending frames related to flow control only when there are other frames to send or when a peer is blocked ensures that flow control doesn’t cause extra packets to be sent.
-If a sender runs out of flow control credit, it will be unable to send new data and is considered blocked. It is generally considered best to not let the sender become blocked. To avoid blocking a sender, and to reasonably account for the possibility of loss, a receiver should send a MAX_DATA or MAX_STREAM_DATA frame at least two round trips before it expects the sender to get blocked.
-A receiver MUST NOT wait for a STREAM_DATA_BLOCKED or DATA_BLOCKED frame before sending MAX_STREAM_DATA or MAX_DATA, since doing so will mean that a sender will be blocked for at least an entire round trip, and potentially for longer if the peer chooses to not send STREAM_DATA_BLOCKED or DATA_BLOCKED frames.
-Endpoints need to eventually agree on the amount of flow control credit that has been consumed, to avoid either exceeding flow control limits or deadlocking.
-On receipt of a RESET_STREAM frame, an endpoint will tear down state for the matching stream and ignore further data arriving on that stream. Without the offset included in RESET_STREAM, the two endpoints could disagree on the number of bytes that count towards connection flow control.
-To remedy this issue, a RESET_STREAM frame (Section 19.4) includes the final size of data sent on the stream. On receiving a RESET_STREAM frame, a receiver definitively knows how many bytes were sent on that stream before the RESET_STREAM frame, and the receiver MUST use the final size of the stream to account for all bytes sent on the stream in its connection level flow controller.
-RESET_STREAM terminates one direction of a stream abruptly. For a bidirectional stream, RESET_STREAM has no effect on data flow in the opposite direction. Both endpoints MUST maintain flow control state for the stream in the unterminated direction until that direction enters a terminal state, or until one of the endpoints sends CONNECTION_CLOSE.
-The final size is the amount of flow control credit that is consumed by a stream. Assuming that every contiguous byte on the stream was sent once, the final size is the number of bytes sent. More generally, this is one higher than the offset of the byte with the largest offset sent on the stream, or zero if no bytes were sent.
-For a stream that is reset, the final size is carried explicitly in a RESET_STREAM frame. Otherwise, the final size is the offset plus the length of a STREAM frame marked with a FIN flag, or 0 in the case of incoming unidirectional streams.
-An endpoint will know the final size for a stream when the receiving part of the stream enters the “Size Known” or “Reset Recvd” state (Section 3).
-An endpoint MUST NOT send data on a stream at or beyond the final size.
-Once a final size for a stream is known, it cannot change. If a RESET_STREAM or STREAM frame is received indicating a change in the final size for the stream, an endpoint SHOULD respond with a FINAL_SIZE_ERROR error (see Section 11). A receiver SHOULD treat receipt of data at or beyond the final size as a FINAL_SIZE_ERROR error, even after a stream is closed. Generating these errors is not mandatory, but only because requiring that an endpoint generate these errors also means that the endpoint needs to maintain the final size state for closed streams, which could mean a significant state commitment.
-An endpoint limits the cumulative number of incoming streams a peer can open. Only streams with a stream ID less than (max_stream * 4 + initial_stream_id_for_type) can be opened (see Table 5). Initial limits are set in the transport parameters (see Section 18.2) and subsequently limits are advertised using MAX_STREAMS frames (Section 19.11). Separate limits apply to unidirectional and bidirectional streams.
-If a max_streams transport parameter or MAX_STREAMS frame is received with a value greater than 2^60, this would allow a maximum stream ID that cannot be expressed as a variable-length integer (see Section 16). If either is received, the connection MUST be closed immediately with a connection error of type STREAM_LIMIT_ERROR (see Section 10.3).
-Endpoints MUST NOT exceed the limit set by their peer. An endpoint that receives a frame with a stream ID exceeding the limit it has sent MUST treat this as a connection error of type STREAM_LIMIT_ERROR (Section 11).
-Once a receiver advertises a stream limit using the MAX_STREAMS frame, advertising a smaller limit has no effect. A receiver MUST ignore any MAX_STREAMS frame that does not increase the stream limit.
-As with stream and connection flow control, this document leaves when and how many streams to advertise to a peer via MAX_STREAMS to implementations. Implementations might choose to increase limits as streams close to keep the number of streams available to peers roughly consistent.
-An endpoint that is unable to open a new stream due to the peer’s limits SHOULD send a STREAMS_BLOCKED frame (Section 19.14). This signal is considered useful for debugging. An endpoint MUST NOT wait to receive this signal before advertising additional credit, since doing so will mean that the peer will be blocked for at least an entire round trip, and potentially for longer if the peer chooses to not send STREAMS_BLOCKED frames.
-QUIC’s connection establishment combines version negotiation with the cryptographic and transport handshakes to reduce connection establishment latency, as described in Section 7. Once established, a connection may migrate to a different IP or port at either endpoint as described in Section 9. Finally, a connection may be terminated by either endpoint, as described in Section 10.
-Each connection possesses a set of connection identifiers, or connection IDs, each of which can identify the connection. Connection IDs are independently selected by endpoints; each endpoint selects the connection IDs that its peer uses.
-The primary function of a connection ID is to ensure that changes in addressing at lower protocol layers (UDP, IP) don’t cause packets for a QUIC connection to be delivered to the wrong endpoint. Each endpoint selects connection IDs using an implementation-specific (and perhaps deployment-specific) method which will allow packets with that connection ID to be routed back to the endpoint and identified by the endpoint upon receipt.
-Connection IDs MUST NOT contain any information that can be used by an external observer (that is, one that does not cooperate with the issuer) to correlate them with other connection IDs for the same connection. As a trivial example, this means the same connection ID MUST NOT be issued more than once on the same connection.
-Packets with long headers include Source Connection ID and Destination Connection ID fields. These fields are used to set the connection IDs for new connections; see Section 7.2 for details.
-Packets with short headers (Section 17.3) only include the Destination Connection ID and omit the explicit length. The length of the Destination Connection ID field is expected to be known to endpoints. Endpoints using a load balancer that routes based on connection ID could agree with the load balancer on a fixed length for connection IDs, or agree on an encoding scheme. A fixed portion could encode an explicit length, which allows the entire connection ID to vary in length and still be used by the load balancer.
-A Version Negotiation (Section 17.2.1) packet echoes the connection IDs selected by the client, both to ensure correct routing toward the client and to allow the client to validate that the packet is in response to an Initial packet.
-A zero-length connection ID can be used when a connection ID is not needed to route to the correct endpoint. However, multiplexing connections on the same local IP address and port while using zero-length connection IDs will cause failures in the presence of peer connection migration, NAT rebinding, and client port reuse; and therefore MUST NOT be done unless an endpoint is certain that those protocol features are not in use.
-When an endpoint has requested a non-zero-length connection ID, it needs to ensure that the peer has a supply of connection IDs from which to choose for packets sent to the endpoint. These connection IDs are supplied by the endpoint using the NEW_CONNECTION_ID frame (Section 19.15).
-Each Connection ID has an associated sequence number to assist in deduplicating messages. The initial connection ID issued by an endpoint is sent in the Source Connection ID field of the long packet header (Section 17.2) during the handshake. The sequence number of the initial connection ID is 0. If the preferred_address transport parameter is sent, the sequence number of the supplied connection ID is 1.
-Additional connection IDs are communicated to the peer using NEW_CONNECTION_ID frames (Section 19.15). The sequence number on each newly-issued connection ID MUST increase by 1. The connection ID randomly selected by the client in the Initial packet and any connection ID provided by a Retry packet are not assigned sequence numbers unless a server opts to retain them as its initial connection ID.
-When an endpoint issues a connection ID, it MUST accept packets that carry this connection ID for the duration of the connection or until its peer invalidates the connection ID via a RETIRE_CONNECTION_ID frame (Section 19.16). Connection IDs that are issued and not retired are considered active; any active connection ID can be used.
-An endpoint SHOULD ensure that its peer has a sufficient number of available and unused connection IDs. Endpoints store received connection IDs for future use and advertise the number of connection IDs they are willing to store with the active_connection_id_limit transport parameter. An endpoint SHOULD NOT provide more connection IDs than the peer’s limit.
-An endpoint SHOULD supply a new connection ID when it receives a packet with a previously unused connection ID or when the peer retires one, unless providing the new connection ID would exceed the peer’s limit. An endpoint MAY limit the frequency or the total number of connection IDs issued for each connection to avoid the risk of running out of connection IDs; see Section 10.4.2.
-An endpoint that initiates migration and requires non-zero-length connection IDs SHOULD ensure that the pool of connection IDs available to its peer allows the peer to use a new connection ID on migration, as the peer will close the connection if the pool is exhausted.
-An endpoint can change the connection ID it uses for a peer to another available one at any time during the connection. An endpoint consumes connection IDs in response to a migrating peer; see Section 9.5 for more.
-An endpoint maintains a set of connection IDs received from its peer, any of which it can use when sending packets. When the endpoint wishes to remove a connection ID from use, it sends a RETIRE_CONNECTION_ID frame to its peer. Sending a RETIRE_CONNECTION_ID frame indicates that the connection ID will not be used again and requests that the peer replace it with a new connection ID using a NEW_CONNECTION_ID frame.
-As discussed in Section 9.5, each connection ID MUST be used on packets sent from only one local address. An endpoint that migrates away from a local address SHOULD retire all connection IDs used on that address once it no longer plans to use that address.
-An endpoint can request that its peer retire connection IDs by sending a NEW_CONNECTION_ID frame with an increased Retire Prior To field. Upon receipt, the peer SHOULD retire the corresponding connection IDs and send the corresponding RETIRE_CONNECTION_ID frames in a timely manner. Failing to do so can cause packets to be delayed, lost, or cause the original endpoint to send a stateless reset in response to a connection ID it can no longer route correctly.
-An endpoint MAY discard a connection ID for which retirement has been requested once an interval of no less than 3 PTO has elapsed since an acknowledgement is received for the NEW_CONNECTION_ID frame requesting that retirement. Subsequent incoming packets using that connection ID could elicit a response with the corresponding stateless reset token.
-Incoming packets are classified on receipt. Packets can either be associated with an existing connection, or - for servers - potentially create a new connection.
-Hosts try to associate a packet with an existing connection. If the packet has a non-zero-length Destination Connection ID corresponding to an existing connection, QUIC processes that packet accordingly. Note that more than one connection ID can be associated with a connection; see Section 5.1.
-If the Destination Connection ID is zero length and the packet matches the local address and port of a connection where the host used zero-length connection IDs, QUIC processes the packet as part of that connection.
-Endpoints can send a Stateless Reset (Section 10.4) for any packets that cannot be attributed to an existing connection. A stateless reset allows a peer to more quickly identify when a connection becomes unusable.
-Packets that are matched to an existing connection are discarded if the packets are inconsistent with the state of that connection. For example, packets are discarded if they indicate a different protocol version than that of the connection, or if the removal of packet protection is unsuccessful once the expected keys are available.
-Invalid packets without packet protection, such as Initial, Retry, or Version Negotiation, MAY be discarded. An endpoint MUST generate a connection error if it commits changes to state before discovering an error.
-Valid packets sent to clients always include a Destination Connection ID that matches a value the client selects. Clients that choose to receive zero-length connection IDs can use the local address and port to identify a connection. Packets that don’t match an existing connection are discarded.
-Due to packet reordering or loss, a client might receive packets for a connection that are encrypted with a key it has not yet computed. The client MAY drop these packets, or MAY buffer them in anticipation of later packets that allow it to compute the key.
-If a client receives a packet that has an unsupported version, it MUST discard that packet.
-If a server receives a packet that has an unsupported version, but the packet is sufficiently large to initiate a new connection for any version supported by the server, it SHOULD send a Version Negotiation packet as described in Section 6.1. Servers MAY rate control these packets to avoid storms of Version Negotiation packets. Otherwise, servers MUST drop packets that specify unsupported versions.
-The first packet for an unsupported version can use different semantics and encodings for any version-specific field. In particular, different packet protection keys might be used for different versions. Servers that do not support a particular version are unlikely to be able to decrypt the payload of the packet. Servers SHOULD NOT attempt to decode or decrypt a packet from an unknown version, but instead send a Version Negotiation packet, provided that the packet is sufficiently long.
-Packets with a supported version, or no version field, are matched to a connection using the connection ID or - for packets with zero-length connection IDs - the local address and port. If the packet doesn’t match an existing connection, the server continues below.
-If the packet is an Initial packet fully conforming with the specification, the server proceeds with the handshake (Section 7). This commits the server to the version that the client selected.
-If a server isn’t currently accepting any new connections, it SHOULD send an Initial packet containing a CONNECTION_CLOSE frame with error code SERVER_BUSY.
-If the packet is a 0-RTT packet, the server MAY buffer a limited number of these packets in anticipation of a late-arriving Initial packet. Clients are not able to send Handshake packets prior to receiving a server response, so servers SHOULD ignore any such packets.
-Servers MUST drop incoming packets under all other circumstances.
-TBD.
-There are certain operations which an application MUST be able to perform when interacting with the QUIC transport. This document does not specify an API, but any implementation of this version of QUIC MUST expose the ability to perform the operations described in this section on a QUIC connection.
-When implementing the client role, applications need to be able to:
- - -When implementing the server role, applications need to be able to:
- - -In either role, applications need to be able to:
- - -Version negotiation ensures that client and server agree to a QUIC version that is mutually supported. A server sends a Version Negotiation packet in response to each packet that might initiate a new connection; see Section 5.2 for details.
-The size of the first packet sent by a client will determine whether a server sends a Version Negotiation packet. Clients that support multiple QUIC versions SHOULD pad the first packet they send to the largest of the minimum packet sizes across all versions they support. This ensures that the server responds if there is a mutually supported version.
-If the version selected by the client is not acceptable to the server, the server responds with a Version Negotiation packet (see Section 17.2.1). This includes a list of versions that the server will accept. An endpoint MUST NOT send a Version Negotiation packet in response to receiving a Version Negotiation packet.
-This system allows a server to process packets with unsupported versions without retaining state. Though either the Initial packet or the Version Negotiation packet that is sent in response could be lost, the client will send new packets until it successfully receives a response or it abandons the connection attempt. As a result, the client discards all state for the connection and does not send any more packets on the connection.
-A server MAY limit the number of Version Negotiation packets it sends. For instance, a server that is able to recognize packets as 0-RTT might choose not to send Version Negotiation packets in response to 0-RTT packets with the expectation that it will eventually receive an Initial packet.
-When a client receives a Version Negotiation packet, it MUST abandon the current connection attempt. Version Negotiation packets are designed to allow future versions of QUIC to negotiate the version in use between endpoints. Future versions of QUIC might change how implementations that support multiple versions of QUIC react to Version Negotiation packets when attempting to establish a connection using this version. How to perform version negotiation is left as future work defined by future versions of QUIC. In particular, that future work will need to ensure robustness against version downgrade attacks Section 21.10.
-[[RFC editor: please remove this section before publication.]]
-When a draft implementation receives a Version Negotiation packet, it MAY use it to attempt a new connection with one of the versions listed in the packet, instead of abandoning the current connection attempt Section 6.2.
-The client MUST check that the Destination and Source Connection ID fields match the Source and Destination Connection ID fields in a packet that the client sent. If this check fails, the packet MUST be discarded.
-Once the Version Negotiation packet is determined to be valid, the client then selects an acceptable protocol version from the list provided by the server. The client then attempts to create a new connection using that version. The new connection MUST use a new random Destination Connection ID different from the one it had previously sent.
-Note that this mechanism does not protect against downgrade attacks and MUST NOT be used outside of draft implementations.
-For a server to use a new version in the future, clients need to correctly handle unsupported versions. To help ensure this, a server SHOULD include a version that is reserved for forcing version negotiation (0x?a?a?a?a as defined in Section 15) when generating a Version Negotiation packet.
-The design of version negotiation permits a server to avoid maintaining state for packets that it rejects in this fashion.
-A client MAY send a packet using a version that is reserved for forcing version negotiation. This can be used to solicit a list of supported versions from a server.
-QUIC relies on a combined cryptographic and transport handshake to minimize connection establishment latency. QUIC uses the CRYPTO frame Section 19.6 to transmit the cryptographic handshake. Version 0x00000001 of QUIC uses TLS as described in [QUIC-TLS]; a different QUIC version number could indicate that a different cryptographic handshake protocol is in use.
-QUIC provides reliable, ordered delivery of the cryptographic handshake data. QUIC packet protection is used to encrypt as much of the handshake protocol as possible. The cryptographic handshake MUST provide the following properties:
- - -The first CRYPTO frame from a client MUST be sent in a single packet. Any second attempt that is triggered by address validation (see Section 8.1) MUST also be sent within a single packet. This avoids having to reassemble a message from multiple packets.
-The first client packet of the cryptographic handshake protocol MUST fit within a 1232 byte QUIC packet payload. This includes overheads that reduce the space available to the cryptographic handshake protocol.
-An endpoint can verify support for Explicit Congestion Notification (ECN) in the first packets it sends, as described in Section 13.4.2.
-The CRYPTO frame can be sent in different packet number spaces. The sequence numbers used by CRYPTO frames to ensure ordered delivery of cryptographic handshake data start from zero in each packet number space.
-Endpoints MUST explicitly negotiate an application protocol. This avoids situations where there is a disagreement about the protocol that is in use.
-Details of how TLS is integrated with QUIC are provided in [QUIC-TLS], but some examples are provided here. An extension of this exchange to support client address validation is shown in Section 8.1.1.
-Once any address validation exchanges are complete, the cryptographic handshake is used to agree on cryptographic keys. The cryptographic handshake is carried in Initial (Section 17.2.2) and Handshake (Section 17.2.4) packets.
-Figure 3 provides an overview of the 1-RTT handshake. Each line shows a QUIC packet with the packet type and packet number shown first, followed by the frames that are typically contained in those packets. So, for instance the first packet is of type Initial, with packet number 0, and contains a CRYPTO frame carrying the ClientHello.
-Note that multiple QUIC packets – even of different encryption levels – may be coalesced into a single UDP datagram (see Section 12.2), and so this handshake may consist of as few as 4 UDP datagrams, or any number more. For instance, the server’s first flight contains packets from the Initial encryption level (obfuscation), the Handshake level, and “0.5-RTT data” from the server at the 1-RTT encryption level.
- - --Client Server - -Initial[0]: CRYPTO[CH] -> - - Initial[0]: CRYPTO[SH] ACK[0] - Handshake[0]: CRYPTO[EE, CERT, CV, FIN] - <- 1-RTT[0]: STREAM[1, "..."] - -Initial[1]: ACK[0] -Handshake[0]: CRYPTO[FIN], ACK[0] -1-RTT[0]: STREAM[0, "..."], ACK[0] -> - - 1-RTT[1]: STREAM[3, "..."], ACK[0] - <- Handshake[1]: ACK[0] --
Figure 3: Example 1-RTT Handshake
-Figure 4 shows an example of a connection with a 0-RTT handshake and a single packet of 0-RTT data. Note that as described in Section 12.3, the server acknowledges 0-RTT data at the 1-RTT encryption level, and the client sends 1-RTT packets in the same packet number space.
- - --Client Server - -Initial[0]: CRYPTO[CH] -0-RTT[0]: STREAM[0, "..."] -> - - Initial[0]: CRYPTO[SH] ACK[0] - Handshake[0] CRYPTO[EE, FIN] - <- 1-RTT[0]: STREAM[1, "..."] ACK[0] - -Initial[1]: ACK[0] -Handshake[0]: CRYPTO[FIN], ACK[0] -1-RTT[1]: STREAM[0, "..."] ACK[0] -> - - 1-RTT[1]: STREAM[3, "..."], ACK[1] - <- Handshake[1]: ACK[0] --
Figure 4: Example 0-RTT Handshake
-A connection ID is used to ensure consistent routing of packets, as described in Section 5.1. The long header contains two connection IDs: the Destination Connection ID is chosen by the recipient of the packet and is used to provide consistent routing; the Source Connection ID is used to set the Destination Connection ID used by the peer.
-During the handshake, packets with the long header (Section 17.2) are used to establish the connection ID that each endpoint uses. Each endpoint uses the Source Connection ID field to specify the connection ID that is used in the Destination Connection ID field of packets being sent to them. Upon receiving a packet, each endpoint sets the Destination Connection ID it sends to match the value of the Source Connection ID that they receive.
-When an Initial packet is sent by a client that has not previously received an Initial or Retry packet from the server, it populates the Destination Connection ID field with an unpredictable value. This MUST be at least 8 bytes in length. Until a packet is received from the server, the client MUST use the same value unless it abandons the connection attempt and starts a new one. The initial Destination Connection ID is used to determine packet protection keys for Initial packets.
-The client populates the Source Connection ID field with a value of its choosing and sets the SCID Len field to indicate the length.
-The first flight of 0-RTT packets use the same Destination and Source Connection ID values as the client’s first Initial.
-Upon first receiving an Initial or Retry packet from the server, the client uses the Source Connection ID supplied by the server as the Destination Connection ID for subsequent packets, including any subsequent 0-RTT packets. That means that a client might change the Destination Connection ID twice during connection establishment, once in response to a Retry and once in response to the first Initial packet from the server. Once a client has received an Initial packet from the server, it MUST discard any packet it receives with a different Source Connection ID.
-A client MUST only change the value it sends in the Destination Connection ID in response to the first packet of each type it receives from the server (Retry or Initial); a server MUST set its value based on the Initial packet. Any additional changes are not permitted; if subsequent packets of those types include a different Source Connection ID, they MUST be discarded. This avoids problems that might arise from stateless processing of multiple Initial packets producing different connection IDs.
-The connection ID can change over the lifetime of a connection, especially in response to connection migration (Section 9); see Section 5.1.1 for details.
-During connection establishment, both endpoints make authenticated declarations of their transport parameters. These declarations are made unilaterally by each endpoint. Endpoints are required to comply with the restrictions implied by these parameters; the description of each parameter includes rules for its handling.
-The encoding of the transport parameters is detailed in Section 18.
-QUIC includes the encoded transport parameters in the cryptographic handshake. Once the handshake completes, the transport parameters declared by the peer are available. Each endpoint validates the value provided by its peer.
-Definitions for each of the defined transport parameters are included in Section 18.2.
-An endpoint MUST treat receipt of a transport parameter with an invalid value as a connection error of type TRANSPORT_PARAMETER_ERROR.
-An endpoint MUST NOT send a parameter more than once in a given transport parameters extension. An endpoint SHOULD treat receipt of duplicate transport parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.
-A server MUST include the original_connection_id transport parameter (Section 18.2) if it sent a Retry packet to enable validation of the Retry, as described in Section 17.2.5.
-Both endpoints store the value of the server transport parameters from a connection and apply them to any 0-RTT packets that are sent in subsequent connections to that peer, except for transport parameters that are explicitly excluded. Remembered transport parameters apply to the new connection until the handshake completes and the client starts sending 1-RTT packets. Once the handshake completes, the client uses the transport parameters established in the handshake.
-The definition of new transport parameters (Section 7.3.2) MUST specify whether they MUST, MAY, or MUST NOT be stored for 0-RTT. A client need not store a transport parameter it cannot process.
-A client MUST NOT use remembered values for the following parameters: original_connection_id, preferred_address, stateless_reset_token, ack_delay_exponent and active_connection_id_limit. The client MUST use the server’s new values in the handshake instead, and absent new values from the server, the default value.
-A client that attempts to send 0-RTT data MUST remember all other transport parameters used by the server. The server can remember these transport parameters, or store an integrity-protected copy of the values in the ticket and recover the information when accepting 0-RTT data. A server uses the transport parameters in determining whether to accept 0-RTT data.
-If 0-RTT data is accepted by the server, the server MUST NOT reduce any limits or alter any values that might be violated by the client with its 0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT set values for the following parameters (Section 18.2) that are smaller than the remembered value of the parameters.
- - -Omitting or setting a zero value for certain transport parameters can result in 0-RTT data being enabled, but not usable. The applicable subset of transport parameters that permit sending of application data SHOULD be set to non-zero values for 0-RTT. This includes initial_max_data and either initial_max_streams_bidi and initial_max_stream_data_bidi_remote, or initial_max_streams_uni and initial_max_stream_data_uni.
-A server MUST either reject 0-RTT data or abort a handshake if the implied values for transport parameters cannot be supported.
-When sending frames in 0-RTT packets, a client MUST only use remembered transport parameters; importantly, it MUST NOT use updated values that it learns from the server’s updated transport parameters or from frames received in 1-RTT packets. Updated values of transport parameters from the handshake apply only to 1-RTT packets. For instance, flow control limits from remembered transport parameters apply to all 0-RTT packets even if those values are increased by the handshake or by frames sent in 1-RTT packets. A server MAY treat use of updated transport parameters in 0-RTT as a connection error of type PROTOCOL_VIOLATION.
-New transport parameters can be used to negotiate new protocol behavior. An endpoint MUST ignore transport parameters that it does not support. Absence of a transport parameter therefore disables any optional protocol feature that is negotiated using the parameter. As described in Section 18.1, some identifiers are reserved in order to exercise this requirement.
-New transport parameters can be registered according to the rules in Section 22.1.
-Implementations need to maintain a buffer of CRYPTO data received out of order. Because there is no flow control of CRYPTO frames, an endpoint could potentially force its peer to buffer an unbounded amount of data.
-Implementations MUST support buffering at least 4096 bytes of data received in CRYPTO frames out of order. Endpoints MAY choose to allow more data to be buffered during the handshake. A larger limit during the handshake could allow for larger keys or credentials to be exchanged. An endpoint’s buffer size does not need to remain constant during the life of the connection.
-Being unable to buffer CRYPTO frames during the handshake can lead to a connection failure. If an endpoint’s buffer is exceeded during the handshake, it can expand its buffer temporarily to complete the handshake. If an endpoint does not expand its buffer, it MUST close the connection with a CRYPTO_BUFFER_EXCEEDED error code.
-Once the handshake completes, if an endpoint is unable to buffer all data in a CRYPTO frame, it MAY discard that CRYPTO frame and all CRYPTO frames received in the future, or it MAY close the connection with a CRYPTO_BUFFER_EXCEEDED error code. Packets containing discarded CRYPTO frames MUST be acknowledged because the packet has been received and processed by the transport even though the CRYPTO frame was discarded.
-Address validation is used by QUIC to avoid being used for a traffic amplification attack. In such an attack, a packet is sent to a server with spoofed source address information that identifies a victim. If a server generates more or larger packets in response to that packet, the attacker can use the server to send more data toward the victim than it would be able to send on its own.
-The primary defense against amplification attack is verifying that an endpoint is able to receive packets at the transport address that it claims. Address validation is performed both during connection establishment (see Section 8.1) and during connection migration (see Section 8.2).
-Connection establishment implicitly provides address validation for both endpoints. In particular, receipt of a packet protected with Handshake keys confirms that the client received the Initial packet from the server. Once the server has successfully processed a Handshake packet from the client, it can consider the client address to have been validated.
-Prior to validating the client address, servers MUST NOT send more than three times as many bytes as the number of bytes they have received. This limits the magnitude of any amplification attack that can be mounted using spoofed source addresses. In determining this limit, servers only count the size of successfully processed packets.
-Clients MUST ensure that UDP datagrams containing only Initial packets are sized to at least 1200 bytes, adding padding to packets in the datagram as necessary. Sending padded datagrams ensures that the server is not overly constrained by the amplification restriction.
-Packet loss, in particular loss of a Handshake packet from the server, can cause a situation in which the server cannot send when the client has no data to send and the anti-amplification limit is reached. In order to avoid this causing a handshake deadlock, clients SHOULD send a packet upon a crypto retransmission timeout, as described in [QUIC-RECOVERY]. If the client has no data to retransmit and does not have Handshake keys, it SHOULD send an Initial packet in a UDP datagram of at least 1200 bytes. If the client has Handshake keys, it SHOULD send a Handshake packet.
-A server might wish to validate the client address before starting the cryptographic handshake. QUIC uses a token in the Initial packet to provide address validation prior to completing the handshake. This token is delivered to the client during connection establishment with a Retry packet (see Section 8.1.1) or in a previous connection using the NEW_TOKEN frame (see Section 8.1.2).
-In addition to sending limits imposed prior to address validation, servers are also constrained in what they can send by the limits set by the congestion controller. Clients are only constrained by the congestion controller.
-Upon receiving the client’s Initial packet, the server can request address validation by sending a Retry packet (Section 17.2.5) containing a token. This token MUST be repeated by the client in all Initial packets it sends for that connection after it receives the Retry packet. In response to processing an Initial containing a token, a server can either abort the connection or permit it to proceed.
-As long as it is not possible for an attacker to generate a valid token for its own address (see Section 8.1.3) and the client is able to return that token, it proves to the server that it received the token.
-A server can also use a Retry packet to defer the state and processing costs of connection establishment. By giving the client a different connection ID to use, a server can cause the connection to be routed to a server instance with more resources available for new connections.
-A flow showing the use of a Retry packet is shown in Figure 5.
- - --Client Server - -Initial[0]: CRYPTO[CH] -> - - <- Retry+Token - -Initial+Token[1]: CRYPTO[CH] -> - - Initial[0]: CRYPTO[SH] ACK[1] - Handshake[0]: CRYPTO[EE, CERT, CV, FIN] - <- 1-RTT[0]: STREAM[1, "..."] --
Figure 5: Example Handshake with Retry
-A server MAY provide clients with an address validation token during one connection that can be used on a subsequent connection. Address validation is especially important with 0-RTT because a server potentially sends a significant amount of data to a client in response to 0-RTT data.
-The server uses the NEW_TOKEN frame Section 19.7 to provide the client with an address validation token that can be used to validate future connections. The client includes this token in Initial packets to provide address validation in a future connection. The client MUST include the token in all Initial packets it sends, unless a Retry replaces the token with a newer one. The client MUST NOT use the token provided in a Retry for future connections. Servers MAY discard any Initial packet that does not carry the expected token.
-A token SHOULD be constructed in a way that allows the server to distinguish it from tokens that are sent in Retry packets as they are carried in the same field.
-The token MUST NOT include information that would allow it to be linked by an on-path observer to the connection on which it was issued. For example, it cannot include the connection ID or addressing information unless the values are encrypted.
-Unlike the token that is created for a Retry packet, there might be some time between when the token is created and when the token is subsequently used. Thus, a token SHOULD have an expiration time, which could be either an explicit expiration time or an issued timestamp that can be used to dynamically calculate the expiration time. A server can store the expiration time or include it in an encrypted form in the token.
-It is unlikely that the client port number is the same on two different connections; validating the port is therefore unlikely to be successful.
-If the client has a token received in a NEW_TOKEN frame on a previous connection to what it believes to be the same server, it SHOULD include that value in the Token field of its Initial packet. Including a token might allow the server to validate the client address without an additional round trip.
-A token allows a server to correlate activity between the connection where the token was issued and any connection where it is used. Clients that want to break continuity of identity with a server MAY discard tokens provided using the NEW_TOKEN frame. A token obtained in a Retry packet MUST be used immediately during the connection attempt and cannot be used in subsequent connection attempts.
-A client SHOULD NOT reuse a token in different connections. Reusing a token allows connections to be linked by entities on the network path; see Section 9.5. A client MUST NOT reuse a token if it believes that its point of network attachment has changed since the token was last used; that is, if there is a change in its local IP address or network interface. A client needs to start the connection process over if there is any change in its local address prior to completing the handshake.
-Clients might receive multiple tokens on a single connection. Aside from preventing linkability, any token can be used in any connection attempt. Servers can send additional tokens to either enable address validation for multiple connection attempts or to replace older tokens that might become invalid. For a client, this ambiguity means that sending the most recent unused token is most likely to be effective. Though saving and using older tokens has no negative consequences, clients can regard older tokens as being less likely be useful to the server for address validation.
-When a server receives an Initial packet with an address validation token, it MUST attempt to validate the token, unless it has already completed address validation. If the token is invalid then the server SHOULD proceed as if the client did not have a validated address, including potentially sending a Retry. If the validation succeeds, the server SHOULD then allow the handshake to proceed.
- - -In a stateless design, a server can use encrypted and authenticated tokens to pass information to clients that the server can later recover and use to validate a client address. Tokens are not integrated into the cryptographic handshake and so they are not authenticated. For instance, a client might be able to reuse a token. To avoid attacks that exploit this property, a server can limit its use of tokens to only the information needed to validate client addresses.
-Attackers could replay tokens to use servers as amplifiers in DDoS attacks. To protect against such attacks, servers SHOULD ensure that tokens sent in Retry packets are only accepted for a short time. Tokens that are provided in NEW_TOKEN frames (see Section 19.7) need to be valid for longer, but SHOULD NOT be accepted multiple times in a short period. Servers are encouraged to allow tokens to be used only once, if possible.
-An address validation token MUST be difficult to guess. Including a large enough random value in the token would be sufficient, but this depends on the server remembering the value it sends to clients.
-A token-based scheme allows the server to offload any state associated with validation to the client. For this design to work, the token MUST be covered by integrity protection against modification or falsification by clients. Without integrity protection, malicious clients could generate or guess values for tokens that would be accepted by the server. Only the server requires access to the integrity protection key for tokens.
-There is no need for a single well-defined format for the token because the server that generates the token also consumes it. A token could include information about the claimed client address (IP and port), a timestamp, and any other supplementary information the server will need to validate the token in the future.
-Path validation is used during connection migration (see Section 9 and Section 9.6) by the migrating endpoint to verify reachability of a peer from a new local address. In path validation, endpoints test reachability between a specific local address and a specific peer address, where an address is the two-tuple of IP address and port.
-Path validation tests that packets (PATH_CHALLENGE) can be both sent to and received (PATH_RESPONSE) from a peer on the path. Importantly, it validates that the packets received from the migrating endpoint do not carry a spoofed source address.
-Path validation can be used at any time by either endpoint. For instance, an endpoint might check that a peer is still in possession of its address after a period of quiescence.
-Path validation is not designed as a NAT traversal mechanism. Though the mechanism described here might be effective for the creation of NAT bindings that support NAT traversal, the expectation is that one or other peer is able to receive packets without first having sent a packet on that path. Effective NAT traversal needs additional synchronization mechanisms that are not provided here.
-An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that are used for path validation with other frames. In particular, an endpoint may pad a packet carrying a PATH_CHALLENGE for PMTU discovery, or an endpoint may bundle a PATH_RESPONSE with its own PATH_CHALLENGE.
-When probing a new path, an endpoint might want to ensure that its peer has an unused connection ID available for responses. The endpoint can send NEW_CONNECTION_ID and PATH_CHALLENGE frames in the same packet. This ensures that an unused connection ID will be available to the peer when sending a response.
-To initiate path validation, an endpoint sends a PATH_CHALLENGE frame containing a random payload on the path to be validated.
-An endpoint MAY send multiple PATH_CHALLENGE frames to guard against packet loss, however an endpoint SHOULD NOT send multiple PATH_CHALLENGE frames in a single packet. An endpoint SHOULD NOT send a PATH_CHALLENGE more frequently than it would an Initial packet, ensuring that connection migration is no more load on a new path than establishing a new connection.
-The endpoint MUST use unpredictable data in every PATH_CHALLENGE frame so that it can associate the peer’s response with the corresponding PATH_CHALLENGE.
-On receiving a PATH_CHALLENGE frame, an endpoint MUST respond immediately by echoing the data contained in the PATH_CHALLENGE frame in a PATH_RESPONSE frame.
-An endpoint MUST NOT send more than one PATH_RESPONSE frame in response to one PATH_CHALLENGE frame (see Section 13.3). The peer is expected to send more PATH_CHALLENGE frames as necessary to evoke additional PATH_RESPONSE frames.
-A new address is considered valid when a PATH_RESPONSE frame is received that contains the data that was sent in a previous PATH_CHALLENGE. Receipt of an acknowledgment for a packet containing a PATH_CHALLENGE frame is not adequate validation, since the acknowledgment can be spoofed by a malicious peer.
-Note that receipt on a different local address does not result in path validation failure, as it might be a result of a forwarded packet (see Section 9.3.3) or misrouting. It is possible that a valid PATH_RESPONSE might be received in the future.
-Path validation only fails when the endpoint attempting to validate the path abandons its attempt to validate the path.
-Endpoints SHOULD abandon path validation based on a timer. When setting this timer, implementations are cautioned that the new path could have a longer round-trip time than the original. A value of three times the larger of the current Probe Timeout (PTO) or the initial timeout (that is, 2*kInitialRtt) as defined in [QUIC-RECOVERY] is RECOMMENDED. That is:
-- validation_timeout = max(3*PTO, 6*kInitialRtt) --
Note that the endpoint might receive packets containing other frames on the new path, but a PATH_RESPONSE frame with appropriate data is required for path validation to succeed.
-When an endpoint abandons path validation, it determines that the path is unusable. This does not necessarily imply a failure of the connection - endpoints can continue sending packets over other paths as appropriate. If no paths are available, an endpoint can wait for a new path to become available or close the connection.
-A path validation might be abandoned for other reasons besides failure. Primarily, this happens if a connection migration to a new path is initiated while a path validation on the old path is in progress.
-The use of a connection ID allows connections to survive changes to endpoint addresses (IP address and port), such as those caused by an endpoint migrating to a new network. This section describes the process by which an endpoint migrates to a new address.
-The design of QUIC relies on endpoints retaining a stable address for the duration of the handshake. An endpoint MUST NOT initiate connection migration before the handshake is confirmed, as defined in section 4.1.2 of [QUIC-TLS].
-An endpoint also MUST NOT send packets from a different local address, actively initiating migration, if the peer sent the disable_active_migration transport parameter during the handshake. An endpoint which has sent this transport parameter, but detects that a peer has nonetheless migrated to a different network MUST either drop the incoming packets on that path without generating a stateless reset or proceed with path validation and allow the peer to migrate. Generating a stateless reset or closing the connection would allow third parties in the network to cause connections to close by spoofing or otherwise manipulating observed traffic.
-Not all changes of peer address are intentional, or active, migrations. The peer could experience NAT rebinding: a change of address due to a middlebox, usually a NAT, allocating a new outgoing port or even a new outgoing IP address for a flow. An endpoint MUST perform path validation (Section 8.2) if it detects any change to a peer’s address, unless it has previously validated that address.
-When an endpoint has no validated path on which to send packets, it MAY discard connection state. An endpoint capable of connection migration MAY wait for a new path to become available before discarding connection state.
-This document limits migration of connections to new client addresses, except as described in Section 9.6. Clients are responsible for initiating all migrations. Servers do not send non-probing packets (see Section 9.1) toward a client address until they see a non-probing packet from that address. If a client receives packets from an unknown server address, the client MUST discard these packets.
-An endpoint MAY probe for peer reachability from a new local address using path validation Section 8.2 prior to migrating the connection to the new local address. Failure of path validation simply means that the new path is not usable for this connection. Failure to validate a path does not cause the connection to end unless there are no valid alternative paths available.
-An endpoint uses a new connection ID for probes sent from a new local address, see Section 9.5 for further discussion. An endpoint that uses a new local address needs to ensure that at least one new connection ID is available at the peer. That can be achieved by including a NEW_CONNECTION_ID frame in the probe.
-Receiving a PATH_CHALLENGE frame from a peer indicates that the peer is probing for reachability on a path. An endpoint sends a PATH_RESPONSE in response as per Section 8.2.
-PATH_CHALLENGE, PATH_RESPONSE, NEW_CONNECTION_ID, and PADDING frames are “probing frames”, and all other frames are “non-probing frames”. A packet containing only probing frames is a “probing packet”, and a packet containing any other frame is a “non-probing packet”.
-An endpoint can migrate a connection to a new local address by sending packets containing non-probing frames from that address.
-Each endpoint validates its peer’s address during connection establishment. Therefore, a migrating endpoint can send to its peer knowing that the peer is willing to receive at the peer’s current address. Thus an endpoint can migrate to a new local address without first validating the peer’s address.
-When migrating, the new path might not support the endpoint’s current sending rate. Therefore, the endpoint resets its congestion controller, as described in Section 9.4.
-The new path might not have the same ECN capability. Therefore, the endpoint verifies ECN capability as described in Section 13.4.
-Receiving acknowledgments for data sent on the new path serves as proof of the peer’s reachability from the new address. Note that since acknowledgments may be received on any path, return reachability on the new path is not established. To establish return reachability on the new path, an endpoint MAY concurrently initiate path validation Section 8.2 on the new path.
-Receiving a packet from a new peer address containing a non-probing frame indicates that the peer has migrated to that address.
-In response to such a packet, an endpoint MUST start sending subsequent packets to the new peer address and MUST initiate path validation (Section 8.2) to verify the peer’s ownership of the unvalidated address.
-An endpoint MAY send data to an unvalidated peer address, but it MUST protect against potential attacks as described in Section 9.3.1 and Section 9.3.2. An endpoint MAY skip validation of a peer address if that address has been seen recently.
-An endpoint only changes the address that it sends packets to in response to the highest-numbered non-probing packet. This ensures that an endpoint does not send packets to an old peer address in the case that it receives reordered packets.
-After changing the address to which it sends non-probing packets, an endpoint could abandon any path validation for other addresses.
-Receiving a packet from a new peer address might be the result of a NAT rebinding at the peer.
-After verifying a new client address, the server SHOULD send new address validation tokens (Section 8) to the client.
-It is possible that a peer is spoofing its source address to cause an endpoint to send excessive amounts of data to an unwilling host. If the endpoint sends significantly more data than the spoofing peer, connection migration might be used to amplify the volume of data that an attacker can generate toward a victim.
-As described in Section 9.3, an endpoint is required to validate a peer’s new address to confirm the peer’s possession of the new address. Until a peer’s address is deemed valid, an endpoint MUST limit the rate at which it sends data to this address. The endpoint MUST NOT send more than a minimum congestion window’s worth of data per estimated round-trip time (kMinimumWindow, as defined in [QUIC-RECOVERY]). In the absence of this limit, an endpoint risks being used for a denial of service attack against an unsuspecting victim. Note that since the endpoint will not have any round-trip time measurements to this address, the estimate SHOULD be the default initial value (see [QUIC-RECOVERY]).
-If an endpoint skips validation of a peer address as described in Section 9.3, it does not need to limit its sending rate.
-An on-path attacker could cause a spurious connection migration by copying and forwarding a packet with a spoofed address such that it arrives before the original packet. The packet with the spoofed address will be seen to come from a migrating connection, and the original packet will be seen as a duplicate and dropped. After a spurious migration, validation of the source address will fail because the entity at the source address does not have the necessary cryptographic keys to read or respond to the PATH_CHALLENGE frame that is sent to it even if it wanted to.
-To protect the connection from failing due to such a spurious migration, an endpoint MUST revert to using the last validated peer address when validation of a new peer address fails.
-If an endpoint has no state about the last validated peer address, it MUST close the connection silently by discarding all connection state. This results in new packets on the connection being handled generically. For instance, an endpoint MAY send a stateless reset in response to any further incoming packets.
-Note that receipt of packets with higher packet numbers from the legitimate peer address will trigger another connection migration. This will cause the validation of the address of the spurious migration to be abandoned.
-An off-path attacker that can observe packets might forward copies of genuine packets to endpoints. If the copied packet arrives before the genuine packet, this will appear as a NAT rebinding. Any genuine packet will be discarded as a duplicate. If the attacker is able to continue forwarding packets, it might be able to cause migration to a path via the attacker. This places the attacker on path, giving it the ability to observe or drop all subsequent packets.
-Unlike the attack described in Section 9.3.2, the attacker can ensure that the new path is successfully validated.
-This style of attack relies on the attacker using a path that is approximately as fast as the direct path between endpoints. The attack is more reliable if relatively few packets are sent or if packet loss coincides with the attempted attack.
-A non-probing packet received on the original path that increases the maximum received packet number will cause the endpoint to move back to that path. Eliciting packets on this path increases the likelihood that the attack is unsuccessful. Therefore, mitigation of this attack relies on triggering the exchange of packets.
-In response to an apparent migration, endpoints MUST validate the previously active path using a PATH_CHALLENGE frame. This induces the sending of new packets on that path. If the path is no longer viable, the validation attempt will time out and fail; if the path is viable, but no longer desired, the validation will succeed, but only results in probing packets being sent on the path.
-An endpoint that receives a PATH_CHALLENGE on an active path SHOULD send a non-probing packet in response. If the non-probing packet arrives before any copy made by an attacker, this results in the connection being migrated back to the original path. Any subsequent migration to another path restarts this entire process.
-This defense is imperfect, but this is not considered a serious problem. If the path via the attack is reliably faster than the original path despite multiple attempts to use that original path, it is not possible to distinguish between attack and an improvement in routing.
-An endpoint could also use heuristics to improve detection of this style of attack. For instance, NAT rebinding is improbable if packets were recently received on the old path, similarly rebinding is rare on IPv6 paths. Endpoints can also look for duplicated packets. Conversely, a change in connection ID is more likely to indicate an intentional migration rather than an attack.
-The capacity available on the new path might not be the same as the old path. Packets sent on the old path SHOULD NOT contribute to congestion control or RTT estimation for the new path.
-On confirming a peer’s ownership of its new address, an endpoint MUST immediately reset the congestion controller and round-trip time estimator for the new path to initial values (see Sections A.3 and B.3 in [QUIC-RECOVERY]) unless it has knowledge that a previous send rate or round-trip time estimate is valid for the new path. For instance, an endpoint might infer that a change in only the client’s port number is indicative of a NAT rebinding, meaning that the new path is likely to have similar bandwidth and round-trip time. However, this determination will be imperfect. If the determination is incorrect, the congestion controller and the RTT estimator are expected to adapt to the new path. Generally, implementations are advised to be cautious when using previous values on a new path.
-There may be apparent reordering at the receiver when an endpoint sends data and probes from/to multiple addresses during the migration period, since the two resulting paths may have different round-trip times. A receiver of packets on multiple paths will still send ACK frames covering all received packets.
-While multiple paths might be used during connection migration, a single congestion control context and a single loss recovery context (as described in [QUIC-RECOVERY]) may be adequate. For instance, an endpoint might delay switching to a new congestion control context until it is confirmed that an old path is no longer needed (such as the case in Section 9.3.3).
-A sender can make exceptions for probe packets so that their loss detection is independent and does not unduly cause the congestion controller to reduce its sending rate. An endpoint might set a separate timer when a PATH_CHALLENGE is sent, which is cancelled when the corresponding PATH_RESPONSE is received. If the timer fires before the PATH_RESPONSE is received, the endpoint might send a new PATH_CHALLENGE, and restart the timer for a longer period of time.
-Using a stable connection ID on multiple network paths allows a passive observer to correlate activity between those paths. An endpoint that moves between networks might not wish to have their activity correlated by any entity other than their peer, so different connection IDs are used when sending from different local addresses, as discussed in Section 5.1. For this to be effective endpoints need to ensure that connection IDs they provide cannot be linked by any other entity.
-At any time, endpoints MAY change the Destination Connection ID they send to a value that has not been used on another path.
-An endpoint MUST use a new connection ID if it initiates connection migration as described in Section 9.2 or probes a new network path as described in Section 9.1. An endpoint MUST use a new connection ID in response to a change in the address of a peer if the packet with the new peer address uses an active connection ID that has not been previously used by the peer.
-Using different connection IDs for packets sent in both directions on each new network path eliminates the use of the connection ID for linking packets from the same connection across different network paths. Header protection ensures that packet numbers cannot be used to correlate activity. This does not prevent other properties of packets, such as timing and size, from being used to correlate activity.
-Unintentional changes in path without a change in connection ID are possible. For example, after a period of network inactivity, NAT rebinding might cause packets to be sent on a new path when the client resumes sending.
-A client might wish to reduce linkability by employing a new connection ID and source UDP port when sending traffic after a period of inactivity. Changing the UDP port from which it sends packets at the same time might cause the packet to appear as a connection migration. This ensures that the mechanisms that support migration are exercised even for clients that don’t experience NAT rebindings or genuine migrations. Changing port number can cause a peer to reset its congestion state (see Section 9.4), so the port SHOULD only be changed infrequently.
-An endpoint that exhausts available connection IDs cannot probe new paths or initiate migration, nor can it respond to probes or attempts by its peer to migrate. To ensure that migration is possible and packets sent on different paths cannot be correlated, endpoints SHOULD provide new connection IDs before peers migrate; see Section 5.1.1. If a peer might have exhausted available connection IDs, a migrating endpoint could include a NEW_CONNECTION_ID frame in all packets sent on a new network path.
-QUIC allows servers to accept connections on one IP address and attempt to transfer these connections to a more preferred address shortly after the handshake. This is particularly useful when clients initially connect to an address shared by multiple servers but would prefer to use a unicast address to ensure connection stability. This section describes the protocol for migrating a connection to a preferred server address.
-Migrating a connection to a new server address mid-connection is left for future work. If a client receives packets from a new server address not indicated by the preferred_address transport parameter, the client SHOULD discard these packets.
-A server conveys a preferred address by including the preferred_address transport parameter in the TLS handshake.
-Servers MAY communicate a preferred address of each address family (IPv4 and IPv6) to allow clients to pick the one most suited to their network attachment.
-Once the handshake is finished, the client SHOULD select one of the two server’s preferred addresses and initiate path validation (see Section 8.2) of that address using the connection ID provided in the preferred_address transport parameter.
-If path validation succeeds, the client SHOULD immediately begin sending all future packets to the new server address using the new connection ID and discontinue use of the old server address. If path validation fails, the client MUST continue sending all future packets to the server’s original IP address.
-A server might receive a packet addressed to its preferred IP address at any time after it accepts a connection. If this packet contains a PATH_CHALLENGE frame, the server sends a PATH_RESPONSE frame as per Section 8.2. The server MUST send other non-probing frames from its original address until it receives a non-probing packet from the client at its preferred address and until the server has validated the new path.
-The server MUST probe on the path toward the client from its preferred address. This helps to guard against spurious migration initiated by an attacker.
-Once the server has completed its path validation and has received a non-probing packet with a new largest packet number on its preferred address, the server begins sending non-probing packets to the client exclusively from its preferred IP address. It SHOULD drop packets for this connection received on the old IP address, but MAY continue to process delayed packets.
-A client might need to perform a connection migration before it has migrated to the server’s preferred address. In this case, the client SHOULD perform path validation to both the original and preferred server address from the client’s new address concurrently.
-If path validation of the server’s preferred address succeeds, the client MUST abandon validation of the original address and migrate to using the server’s preferred address. If path validation of the server’s preferred address fails but validation of the server’s original address succeeds, the client MAY migrate to its new address and continue sending to the server’s original address.
-If the connection to the server’s preferred address is not from the same client address, the server MUST protect against potential attacks as described in Section 9.3.1 and Section 9.3.2. In addition to intentional simultaneous migration, this might also occur because the client’s access network used a different NAT binding for the server’s preferred address.
-Servers SHOULD initiate path validation to the client’s new address upon receiving a probe packet from a different address. Servers MUST NOT send more than a minimum congestion window’s worth of non-probing packets to the new address before path validation is complete.
-A client that migrates to a new address SHOULD use a preferred address from the same address family for the server.
-Endpoints that send data using IPv6 SHOULD apply an IPv6 flow label in compliance with [RFC6437], unless the local API does not allow setting IPv6 flow labels.
-The IPv6 flow label SHOULD be a pseudo-random function of the source and destination addresses, source and destination UDP ports, and the destination CID. The flow label generation MUST be designed to minimize the chances of linkability with a previously used flow label, as this would enable correlating activity on multiple paths (see Section 9.5).
-A possible implementation is to compute the flow label as a cryptographic hash function of the source and destination addresses, source and destination UDP ports, destination CID, and a local secret.
-An established QUIC connection can be terminated in one of three ways:
- - -An endpoint MAY discard connection state if it does not have a validated path on which it can send packets (see Section 8.2).
-The closing and draining connection states exist to ensure that connections close cleanly and that delayed or reordered packets are properly discarded. These states SHOULD persist for at least three times the current Probe Timeout (PTO) interval as defined in [QUIC-RECOVERY].
-An endpoint enters a closing period after initiating an immediate close (Section 10.3). While closing, an endpoint MUST NOT send packets unless they contain a CONNECTION_CLOSE frame (see Section 10.3 for details). An endpoint retains only enough information to generate a packet containing a CONNECTION_CLOSE frame and to identify packets as belonging to the connection. The endpoint’s selected connection ID and the QUIC version are sufficient information to identify packets for a closing connection; an endpoint can discard all other connection state. An endpoint MAY retain packet protection keys for incoming packets to allow it to read and process a CONNECTION_CLOSE frame.
-The draining state is entered once an endpoint receives a signal that its peer is closing or draining. While otherwise identical to the closing state, an endpoint in the draining state MUST NOT send any packets. Retaining packet protection keys is unnecessary once a connection is in the draining state.
-An endpoint MAY transition from the closing period to the draining period if it receives a CONNECTION_CLOSE frame or stateless reset, both of which indicate that the peer is also closing or draining. The draining period SHOULD end when the closing period would have ended. In other words, the endpoint can use the same end time, but cease retransmission of the closing packet.
-Disposing of connection state prior to the end of the closing or draining period could cause delayed or reordered packets to generate an unnecessary stateless reset. Endpoints that have some alternative means to ensure that late-arriving packets on the connection do not induce a response, such as those that are able to close the UDP socket, MAY use an abbreviated draining period which can allow for faster resource recovery. Servers that retain an open socket for accepting new connections SHOULD NOT exit the closing or draining period early.
-Once the closing or draining period has ended, an endpoint SHOULD discard all connection state. This results in new packets on the connection being handled generically. For instance, an endpoint MAY send a stateless reset in response to any further incoming packets.
-The draining and closing periods do not apply when a stateless reset (Section 10.4) is sent.
-An endpoint is not expected to handle key updates when it is closing or draining. A key update might prevent the endpoint from moving from the closing state to draining, but it otherwise has no impact.
-While in the closing period, an endpoint could receive packets from a new source address, indicating a connection migration (Section 9). An endpoint in the closing state MUST strictly limit the number of packets it sends to this new address until the address is validated (see Section 8.2). A server in the closing state MAY instead choose to discard packets received from a new source address.
-If the idle timeout is enabled, a connection is silently closed and the state is discarded when it remains idle for longer than both the advertised idle timeout (see Section 18.2) and three times the current Probe Timeout (PTO).
-Each endpoint advertises its own idle timeout to its peer. An endpoint restarts any timer it maintains when a packet from its peer is received and processed successfully. The timer is also restarted when sending a packet containing frames other than ACK or PADDING (an ack-eliciting packet; see [QUIC-RECOVERY]), but only if no other ack-eliciting packets have been sent since last receiving a packet. Restarting when sending packets ensures that connections do not prematurely time out when initiating new activity.
-The value for an idle timeout can be asymmetric. The value advertised by an endpoint is only used to determine whether the connection is live at that endpoint. An endpoint that sends packets near the end of the idle timeout period of a peer risks having those packets discarded if its peer enters the draining state before the packets arrive. If a peer could timeout within a Probe Timeout (PTO; see Section 6.3 of [QUIC-RECOVERY]), it is advisable to test for liveness before sending any data that cannot be retried safely. Note that it is likely that only applications or application protocols will know what information can be retried.
-An endpoint sends a CONNECTION_CLOSE frame (Section 19.19) to terminate the connection immediately. A CONNECTION_CLOSE frame causes all streams to immediately become closed; open streams can be assumed to be implicitly reset.
-After sending a CONNECTION_CLOSE frame, endpoints immediately enter the closing state. During the closing period, an endpoint that sends a CONNECTION_CLOSE frame SHOULD respond to any packet that it receives with another packet containing a CONNECTION_CLOSE frame. To minimize the state that an endpoint maintains for a closing connection, endpoints MAY send the exact same packet. However, endpoints SHOULD limit the number of packets they generate containing a CONNECTION_CLOSE frame. For instance, an endpoint could progressively increase the number of packets that it receives before sending additional packets or increase the time between packets.
- - -New packets from unverified addresses could be used to create an amplification attack (see Section 8). To avoid this, endpoints MUST either limit transmission of CONNECTION_CLOSE frames to validated addresses or drop packets without response if the response would be more than three times larger than the received packet.
-After receiving a CONNECTION_CLOSE frame, endpoints enter the draining state. An endpoint that receives a CONNECTION_CLOSE frame MAY send a single packet containing a CONNECTION_CLOSE frame before entering the draining state, using a CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An endpoint MUST NOT send further packets, which could result in a constant exchange of CONNECTION_CLOSE frames until the closing period on either peer ended.
-An immediate close can be used after an application protocol has arranged to close a connection. This might be after the application protocols negotiates a graceful shutdown. The application protocol exchanges whatever messages that are needed to cause both endpoints to agree to close the connection, after which the application requests that the connection be closed. The application protocol can use a CONNECTION_CLOSE frame with an appropriate error code to signal closure.
-When sending CONNECTION_CLOSE, the goal is to ensure that the peer will process the frame. Generally, this means sending the frame in a packet with the highest level of packet protection to avoid the packet being discarded. However, during the handshake, it is possible that more advanced packet protection keys are not available to the peer, so the frame MAY be replicated in a packet that uses a lower packet protection level.
-After the handshake is confirmed, an endpoint MUST send any CONNECTION_CLOSE frames in a 1-RTT packet. Prior to handshake confirmation, the peer might not have 1-RTT keys, so the endpoint SHOULD send CONNECTION_CLOSE frames in a Handshake packet. If the endpoint does not have Handshake keys, it SHOULD send CONNECTION_CLOSE frames in an Initial packet.
-A client will always know whether the server has Handshake keys (see Section 17.2.2.1), but it is possible that a server does not know whether the client has Handshake keys. Under these circumstances, a server SHOULD send a CONNECTION_CLOSE frame in both Handshake and Initial packets to ensure that at least one of them is processable by the client. These packets can be coalesced into a single UDP datagram (see Section 12.2).
-A stateless reset is provided as an option of last resort for an endpoint that does not have access to the state of a connection. A crash or outage might result in peers continuing to send data to an endpoint that is unable to properly continue the connection. An endpoint MAY send a stateless reset in response to receiving a packet that it cannot associate with an active connection.
-A stateless reset is not appropriate for signaling error conditions. An endpoint that wishes to communicate a fatal connection error MUST use a CONNECTION_CLOSE frame if it has sufficient state to do so.
-To support this process, a token is sent by endpoints. The token is carried in the Stateless Reset Token field of a NEW_CONNECTION_ID frame. Servers can also specify a stateless_reset_token transport parameter during the handshake that applies to the connection ID that it selected during the handshake; clients cannot use this transport parameter because their transport parameters don’t have confidentiality protection. These tokens are protected by encryption, so only client and server know their value. Tokens are invalidated when their associated connection ID is retired via a RETIRE_CONNECTION_ID frame (Section 19.16).
-An endpoint that receives packets that it cannot process sends a packet in the following layout:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -|0|1| Unpredictable Bits (38 ..) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| | -+ + -| | -+ Stateless Reset Token (128) + -| | -+ + -| | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 6: Stateless Reset Packet
-This design ensures that a stateless reset packet is - to the extent possible - indistinguishable from a regular packet with a short header.
-A stateless reset uses an entire UDP datagram, starting with the first two bits of the packet header. The remainder of the first byte and an arbitrary number of bytes following it that are set to unpredictable values. The last 16 bytes of the datagram contain a Stateless Reset Token.
-To entities other than its intended recipient, a stateless reset will appear to be a packet with a short header. For the stateless reset to appear as a valid QUIC packet, the Unpredictable Bits field needs to include at least 38 bits of data (or 5 bytes, less the two fixed bits).
-A minimum size of 21 bytes does not guarantee that a stateless reset is difficult to distinguish from other packets if the recipient requires the use of a connection ID. To prevent a resulting stateless reset from being trivially distinguishable from a valid packet, all packets sent by an endpoint SHOULD be padded to at least 22 bytes longer than the minimum connection ID that the endpoint might use. An endpoint that sends a stateless reset in response to packet that is 43 bytes or less in length SHOULD send a stateless reset that is one byte shorter than the packet it responds to.
-These values assume that the Stateless Reset Token is the same as the minimum expansion of the packet protection AEAD. Additional unpredictable bytes are necessary if the endpoint could have negotiated a packet protection scheme with a larger minimum expansion.
-An endpoint MUST NOT send a stateless reset that is three times or more larger than the packet it receives to avoid being used for amplification. Section 10.4.3 describes additional limits on stateless reset size.
-Endpoints MUST discard packets that are too small to be valid QUIC packets. With the set of AEAD functions defined in [QUIC-TLS], packets that are smaller than 21 bytes are never valid.
-Endpoints MUST send stateless reset packets formatted as a packet with a short header. However, endpoints MUST treat any packet ending in a valid stateless reset token as a stateless reset, as other QUIC versions might allow the use of a long header.
-An endpoint MAY send a stateless reset in response to a packet with a long header. Sending a stateless reset is not effective prior to the stateless reset token being available to a peer. In this QUIC version, packets with a long header are only used during connection establishment. Because the stateless reset token is not available until connection establishment is complete or near completion, ignoring an unknown packet with a long header might be as effective as sending a stateless reset.
-An endpoint cannot determine the Source Connection ID from a packet with a short header, therefore it cannot set the Destination Connection ID in the stateless reset packet. The Destination Connection ID will therefore differ from the value used in previous packets. A random Destination Connection ID makes the connection ID appear to be the result of moving to a new connection ID that was provided using a NEW_CONNECTION_ID frame (Section 19.15).
-Using a randomized connection ID results in two problems:
- - -This stateless reset design is specific to QUIC version 1. An endpoint that supports multiple versions of QUIC needs to generate a stateless reset that will be accepted by peers that support any version that the endpoint might support (or might have supported prior to losing state). Designers of new versions of QUIC need to be aware of this and either reuse this design, or use a portion of the packet other than the last 16 bytes for carrying data.
-An endpoint detects a potential stateless reset when an incoming packet either cannot be associated with a connection, cannot be decrypted, or is marked as a duplicate packet. The endpoint MUST then compare the last 16 bytes of the packet with all Stateless Reset Tokens corresponding to active connection IDs that the endpoint has used for sending packets to the IP address and port on which the datagram is received. This includes Stateless Reset Tokens from NEW_CONNECTION_ID frames and the server’s transport parameters. An endpoint MUST NOT check for any Stateless Reset Tokens associated with connection IDs it has not used or for connection IDs that have been retired.
-If the last 16 bytes of the packet values are identical to a Stateless Reset Token, the endpoint MUST enter the draining period and not send any further packets on this connection. If the comparison fails, the packet can be discarded.
-The stateless reset token MUST be difficult to guess. In order to create a Stateless Reset Token, an endpoint could randomly generate [RFC4086] a secret for every connection that it creates. However, this presents a coordination problem when there are multiple instances in a cluster or a storage problem for an endpoint that might lose state. Stateless reset specifically exists to handle the case where state is lost, so this approach is suboptimal.
-A single static key can be used across all connections to the same endpoint by generating the proof using a second iteration of a preimage-resistant function that takes a static key and the connection ID chosen by the endpoint (see Section 5.1) as input. An endpoint could use HMAC [RFC2104] (for example, HMAC(static_key, connection_id)) or HKDF [RFC5869] (for example, using the static key as input keying material, with the connection ID as salt). The output of this function is truncated to 16 bytes to produce the Stateless Reset Token for that connection.
-An endpoint that loses state can use the same method to generate a valid Stateless Reset Token. The connection ID comes from the packet that the endpoint receives.
-This design relies on the peer always sending a connection ID in its packets so that the endpoint can use the connection ID from a packet to reset the connection. An endpoint that uses this design MUST either use the same connection ID length for all connections or encode the length of the connection ID such that it can be recovered without state. In addition, it cannot provide a zero-length connection ID.
-Revealing the Stateless Reset Token allows any entity to terminate the connection, so a value can only be used once. This method for choosing the Stateless Reset Token means that the combination of connection ID and static key MUST NOT be used for another connection. A denial of service attack is possible if the same connection ID is used by instances that share a static key, or if an attacker can cause a packet to be routed to an instance that has no state but the same static key; see Section 21.9. A connection ID from a connection that is reset by revealing the Stateless Reset Token MUST NOT be reused for new connections at nodes that share a static key.
-The same Stateless Reset Token MUST NOT be used for multiple connection IDs. Endpoints are not required to compare new values against all previous values, but a duplicate value MAY be treated as a connection error of type PROTOCOL_VIOLATION.
-Note that Stateless Reset packets do not have any cryptographic protection.
-The design of a Stateless Reset is such that without knowing the stateless reset token it is indistinguishable from a valid packet. For instance, if a server sends a Stateless Reset to another server it might receive another Stateless Reset in response, which could lead to an infinite exchange.
-An endpoint MUST ensure that every Stateless Reset that it sends is smaller than the packet which triggered it, unless it maintains state sufficient to prevent looping. In the event of a loop, this results in packets eventually being too small to trigger a response.
-An endpoint can remember the number of Stateless Reset packets that it has sent and stop generating new Stateless Reset packets once a limit is reached. Using separate limits for different remote addresses will ensure that Stateless Reset packets can be used to close connections when other peers or connections have exhausted limits.
-Reducing the size of a Stateless Reset below 41 bytes means that the packet could reveal to an observer that it is a Stateless Reset, depending upon the length of the peer’s connection IDs. Conversely, refusing to send a Stateless Reset in response to a small packet might result in Stateless Reset not being useful in detecting cases of broken connections where only very small packets are sent; such failures might only be detected by other means, such as timers.
-An endpoint that detects an error SHOULD signal the existence of that error to its peer. Both transport-level and application-level errors can affect an entire connection (see Section 11.1), while only application-level errors can be isolated to a single stream (see Section 11.2).
-The most appropriate error code (Section 20) SHOULD be included in the frame that signals the error. Where this specification identifies error conditions, it also identifies the error code that is used; though these are worded as requirements, different implementation strategies might lead to different errors being reported. In particular, an endpoint MAY use any applicable error code when it detects an error condition; a generic error code (such as PROTOCOL_VIOLATION or INTERNAL_ERROR) can always be used in place of specific error codes.
-A stateless reset (Section 10.4) is not suitable for any error that can be signaled with a CONNECTION_CLOSE or RESET_STREAM frame. A stateless reset MUST NOT be used by an endpoint that has the state necessary to send a frame on the connection.
-Errors that result in the connection being unusable, such as an obvious violation of protocol semantics or corruption of state that affects an entire connection, MUST be signaled using a CONNECTION_CLOSE frame (Section 19.19). An endpoint MAY close the connection in this manner even if the error only affects a single stream.
-Application protocols can signal application-specific protocol errors using the application-specific variant of the CONNECTION_CLOSE frame. Errors that are specific to the transport, including all those described in this document, are carried in the QUIC-specific variant of the CONNECTION_CLOSE frame.
-A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if it receives more packets on a terminated connection. Limiting the number of retransmissions and the time over which this final packet is sent limits the effort expended on terminated connections.
-An endpoint that chooses not to retransmit packets containing a CONNECTION_CLOSE frame risks a peer missing the first such packet. The only mechanism available to an endpoint that continues to receive data for a terminated connection is to use the stateless reset process (Section 10.4).
-An endpoint that receives an invalid CONNECTION_CLOSE frame MUST NOT signal the existence of the error to its peer.
-If an application-level error affects a single stream, but otherwise leaves the connection in a recoverable state, the endpoint can send a RESET_STREAM frame (Section 19.4) with an appropriate error code to terminate just the affected stream.
-RESET_STREAM MUST be instigated by the protocol using QUIC. RESET_STREAM carries an application error code. Only the application protocol is able to cause a stream to be terminated. A local instance of the application protocol uses a direct API call and a remote instance uses the STOP_SENDING frame, which triggers an automatic RESET_STREAM.
-Resetting a stream without knowledge of the application protocol could cause the protocol to enter an unrecoverable state. Application protocols might require certain streams to be reliably delivered in order to guarantee consistent state between endpoints. Application protocols SHOULD define rules for handling streams that are prematurely cancelled by either endpoint.
-QUIC endpoints communicate by exchanging packets. Packets have confidentiality and integrity protection (see Section 12.1) and are carried in UDP datagrams (see Section 12.2).
-This version of QUIC uses the long packet header (see Section 17.2) during connection establishment. Packets with the long header are Initial (Section 17.2.2), 0-RTT (Section 17.2.3), Handshake (Section 17.2.4), and Retry (Section 17.2.5). Version negotiation uses a version-independent packet with a long header (see Section 17.2.1).
-Packets with the short header (Section 17.3) are designed for minimal overhead and are used after a connection is established and 1-RTT keys are available.
-All QUIC packets except Version Negotiation and Retry packets use authenticated encryption with additional data (AEAD) [RFC5116] to provide confidentiality and integrity protection. Details of packet protection are found in [QUIC-TLS]; this section includes an overview of the process.
-Initial packets are protected using keys that are statically derived. This packet protection is not effective confidentiality protection. Initial protection only exists to ensure that the sender of the packet is on the network path. Any entity that receives the Initial packet from a client can recover the keys necessary to remove packet protection or to generate packets that will be successfully authenticated.
-All other packets are protected with keys derived from the cryptographic handshake. The type of the packet from the long header or key phase from the short header are used to identify which encryption level - and therefore the keys - that are used. Packets protected with 0-RTT and 1-RTT keys are expected to have confidentiality and data origin authentication; the cryptographic handshake ensures that only the communicating endpoints receive the corresponding keys.
-The packet number field contains a packet number, which has additional confidentiality protection that is applied after packet protection is applied (see [QUIC-TLS] for details). The underlying packet number increases with each packet sent in a given packet number space; see Section 12.3 for details.
-Initial (Section 17.2.2), 0-RTT (Section 17.2.3), and Handshake (Section 17.2.4) packets contain a Length field, which determines the end of the packet. The length includes both the Packet Number and Payload fields, both of which are confidentiality protected and initially of unknown length. The length of the Payload field is learned once header protection is removed.
-Using the Length field, a sender can coalesce multiple QUIC packets into one UDP datagram. This can reduce the number of UDP datagrams needed to complete the cryptographic handshake and start sending data. This can also be used to construct PMTU probes (see Section 14.3.1). Receivers MUST be able to process coalesced packets.
-Coalescing packets in order of increasing encryption levels (Initial, 0-RTT, Handshake, 1-RTT) makes it more likely the receiver will be able to process all the packets in a single pass. A packet with a short header does not include a length, so it can only be the last packet included in a UDP datagram. An endpoint SHOULD NOT coalesce multiple packets at the same encryption level.
-Senders MUST NOT coalesce QUIC packets for different connections into a single UDP datagram. Receivers SHOULD ignore any subsequent packets with a different Destination Connection ID than the first packet in the datagram.
-Every QUIC packet that is coalesced into a single UDP datagram is separate and complete. The receiver of coalesced QUIC packets MUST individually process each QUIC packet and separately acknowledge them, as if they were received as the payload of different UDP datagrams. For example, if decryption fails (because the keys are not available or any other reason), the receiver MAY either discard or buffer the packet for later processing and MUST attempt to process the remaining packets.
-Retry packets (Section 17.2.5), Version Negotiation packets (Section 17.2.1), and packets with a short header (Section 17.3) do not contain a Length field and so cannot be followed by other packets in the same UDP datagram. Note also that there is no situation where a Retry or Version Negotiation packet is coalesced with another packet.
-The packet number is an integer in the range 0 to 2^62-1. This number is used in determining the cryptographic nonce for packet protection. Each endpoint maintains a separate packet number for sending and receiving.
-Packet numbers are limited to this range because they need to be representable in whole in the Largest Acknowledged field of an ACK frame (Section 19.3). When present in a long or short header however, packet numbers are reduced and encoded in 1 to 4 bytes (see Section 17.1).
-Version Negotiation (Section 17.2.1) and Retry (Section 17.2.5) packets do not include a packet number.
-Packet numbers are divided into 3 spaces in QUIC:
- - -As described in [QUIC-TLS], each packet type uses different protection keys.
-Conceptually, a packet number space is the context in which a packet can be processed and acknowledged. Initial packets can only be sent with Initial packet protection keys and acknowledged in packets which are also Initial packets. Similarly, Handshake packets are sent at the Handshake encryption level and can only be acknowledged in Handshake packets.
-This enforces cryptographic separation between the data sent in the different packet sequence number spaces. Packet numbers in each space start at packet number 0. Subsequent packets sent in the same packet number space MUST increase the packet number by at least one.
-0-RTT and 1-RTT data exist in the same packet number space to make loss recovery algorithms easier to implement between the two packet types.
-A QUIC endpoint MUST NOT reuse a packet number within the same packet number space in one connection. If the packet number for sending reaches 2^62 - 1, the sender MUST close the connection without sending a CONNECTION_CLOSE frame or any further packets; an endpoint MAY send a Stateless Reset (Section 10.4) in response to further packets that it receives.
-A receiver MUST discard a newly unprotected packet unless it is certain that it has not processed another packet with the same packet number from the same packet number space. Duplicate suppression MUST happen after removing packet protection for the reasons described in Section 9.3 of [QUIC-TLS]. An efficient algorithm for duplicate suppression can be found in Section 3.4.3 of [RFC4303].
-Packet number encoding at a sender and decoding at a receiver are described in Section 17.1.
-The payload of QUIC packets, after removing packet protection, consists of a sequence of complete frames, as shown in Figure 7. Version Negotiation, Stateless Reset, and Retry packets do not contain frames.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Frame 1 (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Frame 2 (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Frame N (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 7: QUIC Payload
-The payload of a packet that contains frames MUST contain at least one frame, and MAY contain multiple frames and multiple frame types. Frames always fit within a single QUIC packet and cannot span multiple packets.
-Each frame begins with a Frame Type, indicating its type, followed by additional type-dependent fields:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Frame Type (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Type-Dependent Fields (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 8: Generic Frame Layout
-The frame types defined in this specification are listed in Table 3. The Frame Type in ACK, STREAM, MAX_STREAMS, STREAMS_BLOCKED, and CONNECTION_CLOSE frames is used to carry other frame-specific flags. For all other frames, the Frame Type field simply identifies the frame. These frames are explained in more detail in Section 19.
- - -| Type Value | -Frame Type Name | -Definition | -
|---|---|---|
| 0x00 | -PADDING | -Section 19.1 | -
| 0x01 | -PING | -Section 19.2 | -
| 0x02 - 0x03 | -ACK | -Section 19.3 | -
| 0x04 | -RESET_STREAM | -Section 19.4 | -
| 0x05 | -STOP_SENDING | -Section 19.5 | -
| 0x06 | -CRYPTO | -Section 19.6 | -
| 0x07 | -NEW_TOKEN | -Section 19.7 | -
| 0x08 - 0x0f | -STREAM | -Section 19.8 | -
| 0x10 | -MAX_DATA | -Section 19.9 | -
| 0x11 | -MAX_STREAM_DATA | -Section 19.10 | -
| 0x12 - 0x13 | -MAX_STREAMS | -Section 19.11 | -
| 0x14 | -DATA_BLOCKED | -Section 19.12 | -
| 0x15 | -STREAM_DATA_BLOCKED | -Section 19.13 | -
| 0x16 - 0x17 | -STREAMS_BLOCKED | -Section 19.14 | -
| 0x18 | -NEW_CONNECTION_ID | -Section 19.15 | -
| 0x19 | -RETIRE_CONNECTION_ID | -Section 19.16 | -
| 0x1a | -PATH_CHALLENGE | -Section 19.17 | -
| 0x1b | -PATH_RESPONSE | -Section 19.18 | -
| 0x1c - 0x1d | -CONNECTION_CLOSE | -Section 19.19 | -
An endpoint MUST treat the receipt of a frame of unknown type as a connection error of type FRAME_ENCODING_ERROR.
-All QUIC frames are idempotent in this version of QUIC. That is, a valid frame does not cause undesirable side effects or errors when received more than once.
-The Frame Type field uses a variable length integer encoding (see Section 16) with one exception. To ensure simple and efficient implementations of frame parsing, a frame type MUST use the shortest possible encoding. Though a two-, four- or eight-byte encoding of the frame types defined in this document is possible, the Frame Type field for these frames is encoded on a single byte. For instance, though 0x4001 is a legitimate two-byte encoding for a variable-length integer with a value of 1, PING frames are always encoded as a single byte with the value 0x01. An endpoint MAY treat the receipt of a frame type that uses a longer encoding than necessary as a connection error of type PROTOCOL_VIOLATION.
-A sender bundles one or more frames in a QUIC packet (see Section 12.4).
-A sender can minimize per-packet bandwidth and computational costs by bundling as many frames as possible within a QUIC packet. A sender MAY wait for a short period of time to bundle multiple frames before sending a packet that is not maximally packed, to avoid sending out large numbers of small packets. An implementation MAY use knowledge about application sending behavior or heuristics to determine whether and for how long to wait. This waiting period is an implementation decision, and an implementation should be careful to delay conservatively, since any delay is likely to increase application-visible latency.
-Stream multiplexing is achieved by interleaving STREAM frames from multiple streams into one or more QUIC packets. A single QUIC packet can include multiple STREAM frames from one or more streams.
-One of the benefits of QUIC is avoidance of head-of-line blocking across multiple streams. When a packet loss occurs, only streams with data in that packet are blocked waiting for a retransmission to be received, while other streams can continue making progress. Note that when data from multiple streams is bundled into a single QUIC packet, loss of that packet blocks all those streams from making progress. Implementations are advised to bundle as few streams as necessary in outgoing packets without losing transmission efficiency to underfilled packets.
-A packet MUST NOT be acknowledged until packet protection has been successfully removed and all frames contained in the packet have been processed. For STREAM frames, this means the data has been enqueued in preparation to be received by the application protocol, but it does not require that data is delivered and consumed.
-Once the packet has been fully processed, a receiver acknowledges receipt by sending one or more ACK frames containing the packet number of the received packet.
-Endpoints acknowledge all packets they receive and process. However, only ack-eliciting packets (see [QUIC-RECOVERY]) trigger the sending of an ACK frame. Packets that are not ack-eliciting are only acknowledged when an ACK frame is sent for other reasons.
-When sending a packet for any reason, an endpoint should attempt to bundle an ACK frame if one has not been sent recently. Doing so helps with timely loss detection at the peer.
-In general, frequent feedback from a receiver improves loss and congestion response, but this has to be balanced against excessive load generated by a receiver that sends an ACK frame in response to every ack-eliciting packet. The guidance offered below seeks to strike this balance.
-An ACK frame SHOULD be generated for at least every second ack-eliciting packet. This recommendation is in keeping with standard practice for TCP [RFC5681].
-An endpoint MUST NOT excessively delay acknowledgements of ack-eliciting packets. An endpoint commits to a maximum delay using the max_ack_delay transport parameter; see Section 18.2. max_ack_delay declares an explicit contract: an endpoint promises to never delay acknowledgments of an ack-eliciting packet by more than the indicated value. If it does, any excess accrues to the RTT estimate and could result in delayed retransmissions from the peer. For Initial and Handshake packets, a max_ack_delay of 0 is used. The sender uses the receiver’s max_ack_delay value in determining timeouts for timer-based retransmission, as detailed in Section 5.2.1 of [QUIC-RECOVERY].
-In order to assist loss detection at the sender, an endpoint SHOULD send an ACK frame immediately on receiving an ack-eliciting packet that is out of order. The endpoint MAY continue sending ACK frames immediately on each subsequently received packet, but the endpoint SHOULD return to acknowledging every other packet after a period of 1/8 x RTT, unless more ack-eliciting packets are received out of order. If every subsequent ack-eliciting packet arrives out of order, then an ACK frame SHOULD be sent immediately for every received ack-eliciting packet.
-Similarly, packets marked with the ECN Congestion Experienced (CE) codepoint in the IP header SHOULD be acknowledged immediately, to reduce the peer’s response time to congestion events.
-As an optimization, a receiver MAY process multiple packets before sending any ACK frames in response. In this case the receiver can determine whether an immediate or delayed acknowledgement should be generated after processing incoming packets.
-Packets containing PADDING frames are considered to be in flight for congestion control purposes [QUIC-RECOVERY]. Sending only PADDING frames might cause the sender to become limited by the congestion controller (as described in [QUIC-RECOVERY]) with no acknowledgments forthcoming from the receiver. Therefore, a sender SHOULD ensure that other frames are sent in addition to PADDING frames to elicit acknowledgments from the receiver.
-An endpoint that is only sending ACK frames will not receive acknowledgments from its peer unless those acknowledgements are included in packets with ack-eliciting frames. An endpoint SHOULD bundle ACK frames with other frames when there are new ack-eliciting packets to acknowledge. When only non-ack-eliciting packets need to be acknowledged, an endpoint MAY wait until an ack-eliciting packet has been received to bundle an ACK frame with outgoing frames.
-The algorithms in [QUIC-RECOVERY] are resilient to receivers that do not follow guidance offered above. However, an implementor should only deviate from these requirements after careful consideration of the performance implications of doing so.
-Packets containing only ACK frames are not congestion controlled, so there are limits on how frequently they can be sent. An endpoint MUST NOT send more than one ACK-frame-only packet in response to receiving an ack-eliciting packet (one containing frames other than ACK and/or PADDING). An endpoint MUST NOT send a packet containing only an ACK frame in response to a non-ack-eliciting packet (one containing only ACK and/or PADDING frames), even if there are packet gaps which precede the received packet. Limiting ACK frames avoids an infinite feedback loop of acknowledgements, which could prevent the connection from ever becoming idle. However, the endpoint acknowledges non-ack-eliciting packets when it sends an ACK frame.
-An endpoint SHOULD treat receipt of an acknowledgment for a packet it did not send as a connection error of type PROTOCOL_VIOLATION, if it is able to detect the condition.
-When an ACK frame is sent, one or more ranges of acknowledged packets are included. Including older packets reduces the chance of spurious retransmits caused by losing previously sent ACK frames, at the cost of larger ACK frames.
-ACK frames SHOULD always acknowledge the most recently received packets, and the more out-of-order the packets are, the more important it is to send an updated ACK frame quickly, to prevent the peer from declaring a packet as lost and spuriously retransmitting the frames it contains.
-Section 13.2.3 and Section 13.2.4 describe an exemplary approach for determining what packets to acknowledge in each ACK frame.
-When a packet containing an ACK frame is sent, the largest acknowledged in that frame may be saved. When a packet containing an ACK frame is acknowledged, the receiver can stop acknowledging packets less than or equal to the largest acknowledged in the sent ACK frame.
-In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT of reordering. In cases with ACK frame loss and reordering, this approach does not guarantee that every acknowledgement is seen by the sender before it is no longer included in the ACK frame. Packets could be received out of order and all subsequent ACK frames containing them could be lost. In this case, the loss recovery algorithm could cause spurious retransmits, but the sender will continue making forward progress.
-To limit ACK Ranges (see Section 19.3.1) to those that have not yet been received by the sender, the receiver SHOULD track which ACK frames have been acknowledged by its peer. The receiver SHOULD exclude already acknowledged packets from future ACK frames whenever these packets would unnecessarily contribute to the ACK frame size. When the receiver is only sending non-ack-eliciting packets, it can bundle a PING or other small ack-eliciting frame with a fraction of them, such as once per round trip, to enable dropping unnecessary ACK ranges and any state for previously sent packets. The receiver MUST NOT bundle an ack-eliciting frame, such as a PING, with all packets that would otherwise be non-ack-eliciting, in order to avoid an infinite feedback loop of acknowledgements.
-To limit receiver state or the size of ACK frames, a receiver MAY limit the number of ACK Ranges it sends. A receiver can do this even without receiving acknowledgment of its ACK frames, with the knowledge this could cause the sender to unnecessarily retransmit some data. Standard QUIC algorithms ([QUIC-RECOVERY]) declare packets lost after sufficiently newer packets are acknowledged. Therefore, the receiver SHOULD repeatedly acknowledge newly received packets in preference to packets received in the past.
-An endpoint measures the delays intentionally introduced between when an ack-eliciting packet is received and the corresponding acknowledgment is sent. The endpoint encodes this delay for the largest acknowledged packet in the Ack Delay field of an ACK frame (see Section 19.3). This allows the receiver of the ACK to adjust for any intentional delays, which is important for getting a better estimate of the path RTT when acknowledgments are delayed. A packet might be held in the OS kernel or elsewhere on the host before being processed. An endpoint MUST NOT include delays that is does not control when populating the Ack Delay field in an ACK frame.
-ACK frames MUST only be carried in a packet that has the same packet number space as the packet being ACKed (see Section 12.1). For instance, packets that are protected with 1-RTT keys MUST be acknowledged in packets that are also protected with 1-RTT keys.
-Packets that a client sends with 0-RTT packet protection MUST be acknowledged by the server in packets protected by 1-RTT keys. This can mean that the client is unable to use these acknowledgments if the server cryptographic handshake messages are delayed or lost. Note that the same limitation applies to other data sent by the server protected by the 1-RTT keys.
-QUIC packets that are determined to be lost are not retransmitted whole. The same applies to the frames that are contained within lost packets. Instead, the information that might be carried in frames is sent again in new frames as needed.
-New frames and packets are used to carry information that is determined to have been lost. In general, information is sent again when a packet containing that information is determined to be lost and sending ceases when a packet containing that information is acknowledged.
- - -Endpoints SHOULD prioritize retransmission of data over sending new data, unless priorities specified by the application indicate otherwise (see Section 2.3).
-Even though a sender is encouraged to assemble frames containing up-to-date information every time it sends a packet, it is not forbidden to retransmit copies of frames from lost packets. A receiver MUST accept packets containing an outdated frame, such as a MAX_DATA frame carrying a smaller maximum data than one found in an older packet.
-Upon detecting losses, a sender MUST take appropriate congestion control action. The details of loss detection and congestion control are described in [QUIC-RECOVERY].
-QUIC endpoints can use Explicit Congestion Notification (ECN) [RFC3168] to detect and respond to network congestion. ECN allows a network node to indicate congestion in the network by setting a codepoint in the IP header of a packet instead of dropping it. Endpoints react to congestion by reducing their sending rate in response, as described in [QUIC-RECOVERY].
-To use ECN, QUIC endpoints first determine whether a path supports ECN marking and the peer is able to access the ECN codepoint in the IP header. A network path does not support ECN if ECN marked packets get dropped or ECN markings are rewritten on the path. An endpoint validates the use of ECN on the path, both during connection establishment and when migrating to a new path (Section 9).
-On receiving a QUIC packet with an ECT or CE codepoint, an ECN-enabled endpoint that can access the ECN codepoints from the enclosing IP packet increases the corresponding ECT(0), ECT(1), or CE count, and includes these counts in subsequent ACK frames (see Section 13.2 and Section 19.3). Note that this requires being able to read the ECN codepoints from the enclosing IP packet, which is not possible on all platforms.
-A packet detected by a receiver as a duplicate does not affect the receiver’s local ECN codepoint counts; see (Section 21.8) for relevant security concerns.
-If an endpoint receives a QUIC packet without an ECT or CE codepoint in the IP packet header, it responds per Section 13.2 with an ACK frame without increasing any ECN counts. If an endpoint does not implement ECN support or does not have access to received ECN codepoints, it does not increase ECN counts.
-Coalesced packets (see Section 12.2) mean that several packets can share the same IP header. The ECN counter for the ECN codepoint received in the associated IP header are incremented once for each QUIC packet, not per enclosing IP packet or UDP datagram.
-Each packet number space maintains separate acknowledgement state and separate ECN counts. For example, if one each of an Initial, 0-RTT, Handshake, and 1-RTT QUIC packet are coalesced, the corresponding counts for the Initial and Handshake packet number space will be incremented by one and the counts for the 1-RTT packet number space will be increased by two.
-It is possible for faulty network devices to corrupt or erroneously drop packets with ECN markings. To provide robust connectivity in the presence of such devices, each endpoint independently validates ECN counts and disables ECN if errors are detected.
-Endpoints validate ECN for packets sent on each network path independently. An endpoint thus validates ECN on new connection establishment, when switching to a new server preferred address, and on active connection migration to a new path.
-Even if an endpoint does not use ECN markings on packets it transmits, the endpoint MUST provide feedback about ECN markings received from the peer if they are accessible. Failing to report ECN counts will cause the peer to disable ECN marking.
-To start ECN validation, an endpoint SHOULD do the following when sending packets on a new path to a peer:
- - -To reduce the chances of misinterpreting congestive loss as packets dropped by a faulty network element, an endpoint could set the ECT(0) codepoint in the first ten outgoing packets on a path, or for a period of three RTTs, whichever occurs first.
-Implementations MAY experiment with and use other strategies for use of ECN. Other methods of probing paths for ECN support are possible, as are different marking strategies. Implementations can also use the ECT(1) codepoint, as specified in [RFC8311].
-An endpoint that sets ECT(0) or ECT(1) codepoints on packets it transmits MUST use the following steps on receiving an ACK frame to validate ECN.
- - -Processing ECN counts out of order can result in validation failure. An endpoint SHOULD NOT perform this validation if this ACK frame does not advance the largest packet number acknowledged in this connection.
-An endpoint could miss acknowledgements for a packet when ACK frames are lost. It is therefore possible for the total increase in ECT(0), ECT(1), and CE counts to be greater than the number of packets acknowledged in an ACK frame. When this happens, and if validation succeeds, the local reference counts MUST be increased to match the counts in the ACK frame.
-If validation fails, then the endpoint stops sending ECN markings in subsequent IP packets with the expectation that either the network path or the peer does not support ECN.
-Upon successful validation, an endpoint can continue to set ECT codepoints in subsequent packets with the expectation that the path is ECN-capable. Network routing and path elements can change mid-connection however; an endpoint MUST disable ECN if validation fails at any point in the connection.
-Even if validation fails, an endpoint MAY revalidate ECN on the same path at any later time in the connection.
-The QUIC packet size includes the QUIC header and protected payload, but not the UDP or IP header.
-Clients MUST ensure they send the first Initial packet in a single IP packet. Similarly, the first Initial packet sent after receiving a Retry packet MUST be sent in a single IP packet.
-The payload of a UDP datagram carrying the first Initial packet MUST be expanded to at least 1200 bytes, by adding PADDING frames to the Initial packet and/or by coalescing the Initial packet (see Section 12.2). Sending a UDP datagram of this size ensures that the network path supports a reasonable Maximum Transmission Unit (MTU), and helps reduce the amplitude of amplification attacks caused by server responses toward an unverified client address; see Section 8.
-The datagram containing the first Initial packet from a client MAY exceed 1200 bytes if the client believes that the Path Maximum Transmission Unit (PMTU) supports the size that it chooses.
-A server MAY send a CONNECTION_CLOSE frame with error code PROTOCOL_VIOLATION in response to the first Initial packet it receives from a client if the UDP datagram is smaller than 1200 bytes. It MUST NOT send any other frame type in response, or otherwise behave as if any part of the offending packet was processed as valid.
-The server MUST also limit the number of bytes it sends before validating the address of the client; see Section 8.
-The PMTU is the maximum size of the entire IP packet including the IP header, UDP header, and UDP payload. The UDP payload includes the QUIC packet header, protected payload, and any authentication fields. The PMTU can depend upon the current path characteristics. Therefore, the current largest UDP payload an implementation will send is referred to as the QUIC maximum packet size.
-QUIC depends on a PMTU of at least 1280 bytes. This is the IPv6 minimum size [RFC8200] and is also supported by most modern IPv4 networks. All QUIC packets (except for PMTU probe packets) SHOULD be sized to fit within the maximum packet size to avoid the packet being fragmented or dropped [RFC8085].
-An endpoint SHOULD use Datagram Packetization Layer PMTU Discovery ([DPLPMTUD]) or implement Path MTU Discovery (PMTUD) [RFC1191] [RFC8201] to determine whether the path to a destination will support a desired message size without fragmentation.
-In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets larger than 1280 bytes. Assuming the minimum IP header size, this results in a QUIC maximum packet size of 1232 bytes for IPv6 and 1252 bytes for IPv4. A QUIC implementation MAY be more conservative in computing the QUIC maximum packet size to allow for unknown tunnel overheads or IP header options/extensions.
-Each pair of local and remote addresses could have a different PMTU. QUIC implementations that implement any kind of PMTU discovery therefore SHOULD maintain a maximum packet size for each combination of local and remote IP addresses.
-If a QUIC endpoint determines that the PMTU between any pair of local and remote IP addresses has fallen below the size needed to support the smallest allowed maximum packet size, it MUST immediately cease sending QUIC packets, except for PMTU probe packets, on the affected path. An endpoint MAY terminate the connection if an alternative path cannot be found.
-PMTU discovery [RFC1191] [RFC8201] relies on reception of ICMP messages (e.g., IPv6 Packet Too Big messages) that indicate when a packet is dropped because it is larger than the local router MTU. DPLPMTUD can also optionally use these messages. This use of ICMP messages is potentially vulnerable to off-path attacks that successfully guess the addresses used on the path and reduce the PMTU to a bandwidth-inefficient value.
-An endpoint MUST ignore an ICMP message that claims the PMTU has decreased below 1280 bytes.
-The requirements for generating ICMP ([RFC1812], [RFC4443]) state that the quoted packet should contain as much of the original packet as possible without exceeding the minimum MTU for the IP version. The size of the quoted packet can actually be smaller, or the information unintelligible, as described in Section 1.1 of [DPLPMTUD].
-QUIC endpoints SHOULD validate ICMP messages to protect from off-path injection as specified in [RFC8201] and Section 5.2 of [RFC8085]. This validation SHOULD use the quoted packet supplied in the payload of an ICMP message to associate the message with a corresponding transport connection [DPLPMTUD].
-ICMP message validation MUST include matching IP addresses and UDP ports [RFC8085] and, when possible, connection IDs to an active QUIC session.
-Further validation can also be provided:
- - -The endpoint SHOULD ignore all ICMP messages that fail validation.
-An endpoint MUST NOT increase PMTU based on ICMP messages. Any reduction in the QUIC maximum packet size MAY be provisional until QUIC’s loss detection algorithm determines that the quoted packet has actually been lost.
-Section 6.4 of [DPLPMTUD] provides considerations for implementing Datagram Packetization Layer PMTUD (DPLPMTUD) with QUIC.
-When implementing the algorithm in Section 5.3 of [DPLPMTUD], the initial value of BASE_PMTU SHOULD be consistent with the minimum QUIC packet size (1232 bytes for IPv6 and 1252 bytes for IPv4).
-PING and PADDING frames can be used to generate PMTU probe packets. These frames might not be retransmitted if a probe packet containing them is lost. However, these frames do consume congestion window, which could delay the transmission of subsequent application data.
-A PING frame can be included in a PMTU probe to ensure that a valid probe is acknowledged.
-The considerations for processing ICMP messages in the previous section also apply if these messages are used by DPLPMTUD.
-Endpoints that rely on the destination connection ID for routing QUIC packets are likely to require that the connection ID be included in PMTU probe packets to route any resulting ICMP messages (Section 14.2) back to the correct endpoint. However, only long header packets (Section 17.2) contain source connection IDs, and long header packets are not decrypted or acknowledged by the peer once the handshake is complete. One way to construct a PMTU probe is to coalesce (see Section 12.2) a Handshake packet (Section 17.2.4) with a short header packet in a single UDP datagram. If the UDP datagram reaches the endpoint, the Handshake packet will be ignored, but the short header packet will be acknowledged. If the UDP datagram elicits an ICMP message, that message will likely contain the source connection ID within the quoted portion of the UDP datagram.
-QUIC versions are identified using a 32-bit unsigned number.
-The version 0x00000000 is reserved to represent version negotiation. This version of the specification is identified by the number 0x00000001.
-Other versions of QUIC might have different properties to this version. The properties of QUIC that are guaranteed to be consistent across all versions of the protocol are described in [QUIC-INVARIANTS].
-Version 0x00000001 of QUIC uses TLS as a cryptographic handshake protocol, as described in [QUIC-TLS].
-Versions with the most significant 16 bits of the version number cleared are reserved for use in future IETF consensus documents.
-Versions that follow the pattern 0x?a?a?a?a are reserved for use in forcing version negotiation to be exercised. That is, any version number where the low four bits of all bytes is 1010 (in binary). A client or server MAY advertise support for any of these reserved versions.
-Reserved version numbers will probably never represent a real protocol; a client MAY use one of these version numbers with the expectation that the server will initiate version negotiation; a server MAY advertise support for one of these versions and can expect that clients ignore the value.
-[[RFC editor: please remove the remainder of this section before publication.]]
-The version number for the final version of this specification (0x00000001), is reserved for the version of the protocol that is published as an RFC.
-Version numbers used to identify IETF drafts are created by adding the draft number to 0xff000000. For example, draft-ietf-quic-transport-13 would be identified as 0xff00000D.
-Implementors are encouraged to register version numbers of QUIC that they are using for private experimentation on the GitHub wiki at <https://github.com/quicwg/base-drafts/wiki/QUIC-Versions>.
-QUIC packets and frames commonly use a variable-length encoding for non-negative integer values. This encoding ensures that smaller integer values need fewer bytes to encode.
-The QUIC variable-length integer encoding reserves the two most significant bits of the first byte to encode the base 2 logarithm of the integer encoding length in bytes. The integer value is encoded on the remaining bits, in network byte order.
-This means that integers are encoded on 1, 2, 4, or 8 bytes and can encode 6, 14, 30, or 62 bit values respectively. Table 4 summarizes the encoding properties.
- - -| 2Bit | -Length | -Usable Bits | -Range | -
|---|---|---|---|
| 00 | -1 | -6 | -0-63 | -
| 01 | -2 | -14 | -0-16383 | -
| 10 | -4 | -30 | -0-1073741823 | -
| 11 | -8 | -62 | -0-4611686018427387903 | -
For example, the eight byte sequence c2 19 7c 5e ff 14 e8 8c (in hexadecimal) decodes to the decimal value 151288809941952652; the four byte sequence 9d 7f 3e 7d decodes to 494878333; the two byte sequence 7b bd decodes to 15293; and the single byte 25 decodes to 37 (as does the two byte sequence 40 25).
-Error codes (Section 20) and versions (Section 15) are described using integers, but do not use this encoding.
-All numeric values are encoded in network byte order (that is, big-endian) and all field sizes are in bits. Hexadecimal notation is used for describing the value of fields.
-Packet numbers are integers in the range 0 to 2^62-1 (Section 12.3). When present in long or short packet headers, they are encoded in 1 to 4 bytes. The number of bits required to represent the packet number is reduced by including the least significant bits of the packet number.
-The encoded packet number is protected as described in Section 5.4 of [QUIC-TLS].
-The sender MUST use a packet number size able to represent more than twice as large a range than the difference between the largest acknowledged packet and packet number being sent. A peer receiving the packet will then correctly decode the packet number, unless the packet is delayed in transit such that it arrives after many higher-numbered packets have been received. An endpoint SHOULD use a large enough packet number encoding to allow the packet number to be recovered even if the packet arrives after packets that are sent afterwards.
-As a result, the size of the packet number encoding is at least one bit more than the base-2 logarithm of the number of contiguous unacknowledged packet numbers, including the new packet.
-For example, if an endpoint has received an acknowledgment for packet 0xabe8bc, sending a packet with a number of 0xac5c02 requires a packet number encoding with 16 bits or more; whereas the 24-bit packet number encoding is needed to send a packet with a number of 0xace8fe.
-At a receiver, protection of the packet number is removed prior to recovering the full packet number. The full packet number is then reconstructed based on the number of significant bits present, the value of those bits, and the largest packet number received on a successfully authenticated packet. Recovering the full packet number is necessary to successfully remove packet protection.
-Once header protection is removed, the packet number is decoded by finding the packet number value that is closest to the next expected packet. The next expected packet is the highest received packet number plus one. For example, if the highest successfully authenticated packet had a packet number of 0xa82f30ea, then a packet containing a 16-bit value of 0x9b32 will be decoded as 0xa82f9b32. Example pseudo-code for packet number decoding can be found in Appendix A.
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|1|1|T T|X X X X| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 9: Long Header Packet Format
-Long headers are used for packets that are sent prior to the establishment of 1-RTT keys. Once both conditions are met, a sender switches to sending packets using the short header (Section 17.3). The long form allows for special packets - such as the Version Negotiation packet - to be represented in this uniform fixed-length packet format. Packets that use the long header contain the following fields:
- - -In this version of QUIC, the following packet types with the long header are defined:
- - -| Type | -Name | -Section | -
|---|---|---|
| 0x0 | -Initial | -Section 17.2.2 | -
| 0x1 | -0-RTT | -Section 17.2.3 | -
| 0x2 | -Handshake | -Section 17.2.4 | -
| 0x3 | -Retry | -Section 17.2.5 | -
The header form bit, connection ID lengths byte, Destination and Source Connection ID fields, and Version fields of a long header packet are version-independent. The other fields in the first byte are version-specific. See [QUIC-INVARIANTS] for details on how packets from different versions of QUIC are interpreted.
-The interpretation of the fields and the payload are specific to a version and packet type. While type-specific semantics for this version are described in the following sections, several long-header packets in this version of QUIC contain these additional fields:
- - -A Version Negotiation packet is inherently not version-specific. Upon receipt by a client, it will be identified as a Version Negotiation packet based on the Version field having a value of 0.
-The Version Negotiation packet is a response to a client packet that contains a version that is not supported by the server, and is only sent by servers.
-The layout of a Version Negotiation packet is:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|1| Unused (7) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..2040) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Supported Version 1 (32) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Supported Version 2 (32)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Supported Version N (32)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 10: Version Negotiation Packet
-The value in the Unused field is selected randomly by the server. Clients MUST ignore the value of this field. Servers SHOULD set the most significant bit of this field (0x40) to 1 so that Version Negotiation packets appear to have the Fixed Bit field.
-The Version field of a Version Negotiation packet MUST be set to 0x00000000.
-The server MUST include the value from the Source Connection ID field of the packet it receives in the Destination Connection ID field. The value for Source Connection ID MUST be copied from the Destination Connection ID of the received packet, which is initially randomly selected by a client. Echoing both connection IDs gives clients some assurance that the server received the packet and that the Version Negotiation packet was not generated by an off-path attacker.
-As future versions of QUIC may support Connection IDs larger than the version 1 limit, Version Negotiation packets could carry Connection IDs that are longer than 20 bytes.
-The remainder of the Version Negotiation packet is a list of 32-bit versions which the server supports.
-A Version Negotiation packet cannot be explicitly acknowledged in an ACK frame by a client. Receiving another Initial packet implicitly acknowledges a Version Negotiation packet.
-The Version Negotiation packet does not include the Packet Number and Length fields present in other packets that use the long header form. Consequently, a Version Negotiation packet consumes an entire UDP datagram.
-A server MUST NOT send more than one Version Negotiation packet in response to a single UDP datagram.
-See Section 6 for a description of the version negotiation process.
-An Initial packet uses long headers with a type value of 0x0. It carries the first CRYPTO frames sent by the client and server to perform key exchange, and carries ACKs in either direction.
- - --+-+-+-+-+-+-+-+-+ -|1|1| 0 |R R|P P| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Token Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Token (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Packet Number (8/16/24/32) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 11: Initial Packet
-The Initial packet contains a long header as well as the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. Between the SCID and Length fields, there are two additional field specific to the Initial packet.
- - -In order to prevent tampering by version-unaware middleboxes, Initial packets are protected with connection- and version-specific keys (Initial keys) as described in [QUIC-TLS]. This protection does not provide confidentiality or integrity against on-path attackers, but provides some level of protection against off-path attackers.
-The client and server use the Initial packet type for any packet that contains an initial cryptographic handshake message. This includes all cases where a new packet containing the initial cryptographic message needs to be created, such as the packets sent after receiving a Retry packet (Section 17.2.5).
-A server sends its first Initial packet in response to a client Initial. A server may send multiple Initial packets. The cryptographic key exchange could require multiple round trips or retransmissions of this data.
-The payload of an Initial packet includes a CRYPTO frame (or frames) containing a cryptographic handshake message, ACK frames, or both. PADDING and CONNECTION_CLOSE frames are also permitted. An endpoint that receives an Initial packet containing other frames can either discard the packet as spurious or treat it as a connection error.
-The first packet sent by a client always includes a CRYPTO frame that contains the entirety of the first cryptographic handshake message. This packet, and the cryptographic handshake message, MUST fit in a single UDP datagram (see Section 7). The first CRYPTO frame sent always begins at an offset of 0 (see Section 7).
-Note that if the server sends a HelloRetryRequest, the client will send a second Initial packet. This Initial packet will continue the cryptographic handshake and will contain a CRYPTO frame with an offset matching the size of the CRYPTO frame sent in the first Initial packet. Cryptographic handshake messages subsequent to the first do not need to fit within a single UDP datagram.
-A client stops both sending and processing Initial packets when it sends its first Handshake packet. A server stops sending and processing Initial packets when it receives its first Handshake packet. Though packets might still be in flight or awaiting acknowledgment, no further Initial packets need to be exchanged beyond this point. Initial packet protection keys are discarded (see Section 4.9.1 of [QUIC-TLS]) along with any loss recovery and congestion control state (see Section 6.5 of [QUIC-RECOVERY]).
-Any data in CRYPTO frames is discarded - and no longer retransmitted - when Initial keys are discarded.
-A 0-RTT packet uses long headers with a type value of 0x1, followed by the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. It is used to carry “early” data from the client to the server as part of the first flight, prior to handshake completion. As part of the TLS handshake, the server can accept or reject this early data.
-See Section 2.3 of [TLS13] for a discussion of 0-RTT data and its limitations.
--+-+-+-+-+-+-+-+-+ -|1|1| 1 |R R|P P| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Packet Number (8/16/24/32) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
0-RTT Packet
-Packet numbers for 0-RTT protected packets use the same space as 1-RTT protected packets.
-After a client receives a Retry packet, 0-RTT packets are likely to have been lost or discarded by the server. A client SHOULD attempt to resend data in 0-RTT packets after it sends a new Initial packet.
-A client MUST NOT reset the packet number it uses for 0-RTT packets, since the keys used to protect 0-RTT packets will not change as a result of responding to a Retry packet. Sending packets with the same packet number in that case is likely to compromise the packet protection for all 0-RTT packets because the same key and nonce could be used to protect different content.
-A client only receives acknowledgments for its 0-RTT packets once the handshake is complete. Consequently, a server might expect 0-RTT packets to start with a packet number of 0. Therefore, in determining the length of the packet number encoding for 0-RTT packets, a client MUST assume that all packets up to the current packet number are in flight, starting from a packet number of 0. Thus, 0-RTT packets could need to use a longer packet number encoding.
-A client MUST NOT send 0-RTT packets once it starts processing 1-RTT packets from the server. This means that 0-RTT packets cannot contain any response to frames from 1-RTT packets. For instance, a client cannot send an ACK frame in a 0-RTT packet, because that can only acknowledge a 1-RTT packet. An acknowledgment for a 1-RTT packet MUST be carried in a 1-RTT packet.
-A server SHOULD treat a violation of remembered limits as a connection error of an appropriate type (for instance, a FLOW_CONTROL_ERROR for exceeding stream data limits).
-A Handshake packet uses long headers with a type value of 0x2, followed by the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. It is used to carry acknowledgments and cryptographic handshake messages from the server and client.
- - --+-+-+-+-+-+-+-+-+ -|1|1| 2 |R R|P P| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Packet Number (8/16/24/32) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 12: Handshake Protected Packet
-Once a client has received a Handshake packet from a server, it uses Handshake packets to send subsequent cryptographic handshake messages and acknowledgments to the server.
-The Destination Connection ID field in a Handshake packet contains a connection ID that is chosen by the recipient of the packet; the Source Connection ID includes the connection ID that the sender of the packet wishes to use (see Section 7.2).
-Handshake packets are their own packet number space, and thus the first Handshake packet sent by a server contains a packet number of 0.
-The payload of this packet contains CRYPTO frames and could contain PADDING, or ACK frames. Handshake packets MAY contain CONNECTION_CLOSE frames. Endpoints MUST treat receipt of Handshake packets with other frames as a connection error.
-Like Initial packets (see Section 17.2.2.1), data in CRYPTO frames at the Handshake encryption level is discarded - and no longer retransmitted - when Handshake protection keys are discarded.
-A Retry packet uses a long packet header with a type value of 0x3. It carries an address validation token created by the server. It is used by a server that wishes to perform a retry (see Section 8.1).
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|1|1| 3 | Unused| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Version (32) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| DCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| SCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Source Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ODCID Len (8) | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Original Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Retry Token (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 13: Retry Packet
-A Retry packet (shown in Figure 13) does not contain any protected fields. The value in the Unused field is selected randomly by the server. In addition to the long header, it contains these additional fields:
- - -The server populates the Destination Connection ID with the connection ID that the client included in the Source Connection ID of the Initial packet.
-The server includes a connection ID of its choice in the Source Connection ID field. This value MUST not be equal to the Destination Connection ID field of the packet sent by the client. The client MUST use this connection ID in the Destination Connection ID of subsequent packets that it sends.
-A server MAY send Retry packets in response to Initial and 0-RTT packets. A server can either discard or buffer 0-RTT packets that it receives. A server can send multiple Retry packets as it receives Initial or 0-RTT packets. A server MUST NOT send more than one Retry packet in response to a single UDP datagram.
-A client MUST accept and process at most one Retry packet for each connection attempt. After the client has received and processed an Initial or Retry packet from the server, it MUST discard any subsequent Retry packets that it receives.
-Clients MUST discard Retry packets that contain an Original Destination Connection ID field that does not match the Destination Connection ID from its Initial packet. This prevents an off-path attacker from injecting a Retry packet.
-The client responds to a Retry packet with an Initial packet that includes the provided Retry Token to continue connection establishment.
-A client sets the Destination Connection ID field of this Initial packet to the value from the Source Connection ID in the Retry packet. Changing Destination Connection ID also results in a change to the keys used to protect the Initial packet. It also sets the Token field to the token provided in the Retry. The client MUST NOT change the Source Connection ID because the server could include the connection ID as part of its token validation logic (see Section 8.1.3).
-The next Initial packet from the client uses the connection ID and token values from the Retry packet (see Section 7.2). Aside from this, the Initial packet sent by the client is subject to the same restrictions as the first Initial packet. A client MUST use the same cryptographic handshake message it includes in this packet. A server MAY treat a packet that contains a different cryptographic handshake message as a connection error or discard it.
-A client MAY attempt 0-RTT after receiving a Retry packet by sending 0-RTT packets to the connection ID provided by the server. A client MUST NOT change the cryptographic handshake message it sends in response to receiving a Retry.
-A client MUST NOT reset the packet number for any packet number space after processing a Retry packet; Section 17.2.3 contains more information on this.
-A server acknowledges the use of a Retry packet for a connection using the original_connection_id transport parameter (see Section 18.2). If the server sends a Retry packet, it MUST include the value of the Original Destination Connection ID field of the Retry packet (that is, the Destination Connection ID field from the client’s first Initial packet) in the transport parameter.
-If the client received and processed a Retry packet, it MUST validate that the original_connection_id transport parameter is present and correct; otherwise, it MUST validate that the transport parameter is absent. A client MUST treat a failed validation as a connection error of type TRANSPORT_PARAMETER_ERROR.
-A Retry packet does not include a packet number and cannot be explicitly acknowledged by a client.
-This version of QUIC defines a single packet type which uses the short packet header.
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+ -|0|1|S|R|R|K|P P| -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Destination Connection ID (0..160) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Packet Number (8/16/24/32) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Protected Payload (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 14: Short Header Packet Format
-The short header can be used after the version and 1-RTT keys are negotiated. Packets that use the short header contain the following fields:
- - -The header form bit and the connection ID field of a short header packet are version-independent. The remaining fields are specific to the selected QUIC version. See [QUIC-INVARIANTS] for details on how packets from different versions of QUIC are interpreted.
-The latency spin bit enables passive latency monitoring from observation points on the network path throughout the duration of a connection. The spin bit is only present in the short packet header, since it is possible to measure the initial RTT of a connection by observing the handshake. Therefore, the spin bit is available after version negotiation and connection establishment are completed. On-path measurement and use of the latency spin bit is further discussed in [QUIC-MANAGEABILITY].
-The spin bit is an OPTIONAL feature of QUIC. A QUIC stack that chooses to support the spin bit MUST implement it as specified in this section.
-Each endpoint unilaterally decides if the spin bit is enabled or disabled for a connection. Implementations MUST allow administrators of clients and servers to disable the spin bit either globally or on a per-connection basis. Even when the spin bit is not disabled by the administrator, implementations MUST disable the spin bit for a given connection with a certain likelihood. The random selection process SHOULD be designed such that on average the spin bit is disabled for at least one eighth of network paths. The selection process performed at the beginning of the connection SHOULD be applied for all paths used by the connection.
-When the spin bit is disabled, endpoints MAY set the spin bit to any value, and MUST ignore any incoming value. It is RECOMMENDED that endpoints set the spin bit to a random value either chosen independently for each packet or chosen independently for each connection ID.
-If the spin bit is enabled for the connection, the endpoint maintains a spin value and sets the spin bit in the short header to the currently stored value when a packet with a short header is sent out. The spin value is initialized to 0 in the endpoint at connection start. Each endpoint also remembers the highest packet number seen from its peer on the connection.
-When a server receives a short header packet that increments the highest packet number seen by the server from the client, it sets the spin value to be equal to the spin bit in the received packet.
-When a client receives a short header packet that increments the highest packet number seen by the client from the server, it sets the spin value to the inverse of the spin bit in the received packet.
-An endpoint resets its spin value to zero when sending the first packet of a given connection with a new connection ID. This reduces the risk that transient spin bit state can be used to link flows across connection migration or ID change.
-With this mechanism, the server reflects the spin value received, while the client ‘spins’ it after one RTT. On-path observers can measure the time between two spin bit toggle events to estimate the end-to-end RTT of a connection.
-The format of the transport parameters is the TransportParameters struct from Figure 15. This is described using the presentation language from Section 3 of [TLS13].
- - -
- enum {
- original_connection_id(0),
- idle_timeout(1),
- stateless_reset_token(2),
- max_packet_size(3),
- initial_max_data(4),
- initial_max_stream_data_bidi_local(5),
- initial_max_stream_data_bidi_remote(6),
- initial_max_stream_data_uni(7),
- initial_max_streams_bidi(8),
- initial_max_streams_uni(9),
- ack_delay_exponent(10),
- max_ack_delay(11),
- disable_active_migration(12),
- preferred_address(13),
- active_connection_id_limit(14),
- (65535)
- } TransportParameterId;
-
- struct {
- TransportParameterId parameter;
- opaque value<0..2^16-1>;
- } TransportParameter;
-
- TransportParameter TransportParameters<0..2^16-1>;
-
-Figure 15: Definition of TransportParameters
-The extension_data field of the quic_transport_parameters extension defined in [QUIC-TLS] contains a TransportParameters value. TLS encoding rules are therefore used to describe the encoding of transport parameters.
-QUIC encodes transport parameters into a sequence of bytes, which are then included in the cryptographic handshake.
-Transport parameters with an identifier of the form 31 * N + 27 for integer values of N are reserved to exercise the requirement that unknown transport parameters be ignored. These transport parameters have no semantics, and may carry arbitrary values.
-This section details the transport parameters defined in this document.
-Many transport parameters listed here have integer values. Those transport parameters that are identified as integers use a variable-length integer encoding (see Section 16) and have a default value of 0 if the transport parameter is absent, unless otherwise stated.
-The following transport parameters are defined:
- - -
- struct {
- opaque ipv4Address[4];
- uint16 ipv4Port;
- opaque ipv6Address[16];
- uint16 ipv6Port;
- opaque connectionId<0..20>;
- opaque statelessResetToken[16];
- } PreferredAddress;
-
-Figure 16: Preferred Address format
- - -If present, transport parameters that set initial flow control limits (initial_max_stream_data_bidi_local, initial_max_stream_data_bidi_remote, and initial_max_stream_data_uni) are equivalent to sending a MAX_STREAM_DATA frame (Section 19.10) on every stream of the corresponding type immediately after opening. If the transport parameter is absent, streams of that type start with a flow control limit of 0.
-A client MUST NOT include an original connection ID, a stateless reset token, or a preferred address. A server MUST treat receipt of any of these transport parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.
-As described in Section 12.4, packets contain one or more frames. This section describes the format and semantics of the core QUIC frame types.
-The PADDING frame (type=0x00) has no semantic value. PADDING frames can be used to increase the size of a packet. Padding can be used to increase an initial client packet to the minimum required size, or to provide protection against traffic analysis for protected packets.
-A PADDING frame has no content. That is, a PADDING frame consists of the single byte that identifies the frame as a PADDING frame.
-Endpoints can use PING frames (type=0x01) to verify that their peers are still alive or to check reachability to the peer. The PING frame contains no additional fields.
-The receiver of a PING frame simply needs to acknowledge the packet containing this frame.
-The PING frame can be used to keep a connection alive when an application or application protocol wishes to prevent the connection from timing out. An application protocol SHOULD provide guidance about the conditions under which generating a PING is recommended. This guidance SHOULD indicate whether it is the client or the server that is expected to send the PING. Having both endpoints send PING frames without coordination can produce an excessive number of packets and poor performance.
-A connection will time out if no packets are sent or received for a period longer than the time specified in the idle_timeout transport parameter (see Section 10). However, state in middleboxes might time out earlier than that. Though REQ-5 in [RFC4787] recommends a 2 minute timeout interval, experience shows that sending packets every 15 to 30 seconds is necessary to prevent the majority of middleboxes from losing state for UDP flows.
-Receivers send ACK frames (types 0x02 and 0x03) to inform senders of packets they have received and processed. The ACK frame contains one or more ACK Ranges. ACK Ranges identify acknowledged packets. If the frame type is 0x03, ACK frames also contain the sum of QUIC packets with associated ECN marks received on the connection up until this point. QUIC implementations MUST properly handle both types and, if they have enabled ECN for packets they send, they SHOULD use the information in the ECN section to manage their congestion state.
-QUIC acknowledgements are irrevocable. Once acknowledged, a packet remains acknowledged, even if it does not appear in a future ACK frame. This is unlike TCP SACKs ([RFC2018]).
-It is expected that a sender will reuse the same packet number across different packet number spaces. ACK frames only acknowledge the packet numbers that were transmitted by the sender in the same packet number space of the packet that the ACK was received in.
-Version Negotiation and Retry packets cannot be acknowledged because they do not contain a packet number. Rather than relying on ACK frames, these packets are implicitly acknowledged by the next Initial packet sent by the client.
-An ACK frame is as follows:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Largest Acknowledged (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ACK Delay (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ACK Range Count (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| First ACK Range (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ACK Ranges (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [ECN Counts] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 17: ACK Frame Format
-ACK frames contain the following fields:
- - -The ACK Ranges field consists of alternating Gap and ACK Range values in descending packet number order. The number of Gap and ACK Range values is determined by the ACK Range Count field; one of each value is present for each value in the ACK Range Count field.
-ACK Ranges are structured as follows:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Gap (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [ACK Range (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Gap (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [ACK Range (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Gap (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [ACK Range (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 18: ACK Ranges
-The fields that form the ACK Ranges are:
- - -Gap and ACK Range value use a relative integer encoding for efficiency. Though each encoded value is positive, the values are subtracted, so that each ACK Range describes progressively lower-numbered packets.
-Each ACK Range acknowledges a contiguous range of packets by indicating the number of acknowledged packets that precede the largest packet number in that range. A value of zero indicates that only the largest packet number is acknowledged. Larger ACK Range values indicate a larger range, with corresponding lower values for the smallest packet number in the range. Thus, given a largest packet number for the range, the smallest value is determined by the formula:
-- smallest = largest - ack_range --
An ACK Range acknowledges all packets between the smallest packet number and the largest, inclusive.
-The largest value for an ACK Range is determined by cumulatively subtracting the size of all preceding ACK Ranges and Gaps.
-Each Gap indicates a range of packets that are not being acknowledged. The number of packets in the gap is one higher than the encoded value of the Gap field.
-The value of the Gap field establishes the largest packet number value for the subsequent ACK Range using the following formula:
-- largest = previous_smallest - gap - 2 --
If any computed packet number is negative, an endpoint MUST generate a connection error of type FRAME_ENCODING_ERROR indicating an error in an ACK frame.
-The ACK frame uses the least significant bit (that is, type 0x03) to indicate ECN feedback and report receipt of QUIC packets with associated ECN codepoints of ECT(0), ECT(1), or CE in the packet’s IP header. ECN Counts are only present when the ACK frame type is 0x03.
-ECN Counts are only parsed when the ACK frame type is 0x03. There are 3 ECN counts, as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ECT(0) Count (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ECT(1) Count (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| ECN-CE Count (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
The three ECN Counts are:
- - -ECN counts are maintained separately for each packet number space.
-An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly terminate the sending part of a stream.
-After sending a RESET_STREAM, an endpoint ceases transmission and retransmission of STREAM frames on the identified stream. A receiver of RESET_STREAM can discard any data that it already received on that stream.
-An endpoint that receives a RESET_STREAM frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The RESET_STREAM frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Application Error Code (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Final Size (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
RESET_STREAM frames contain the following fields:
- - -An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that incoming data is being discarded on receipt at application request. STOP_SENDING requests that a peer cease transmission on a stream.
-A STOP_SENDING frame can be sent for streams in the Recv or Size Known states (see Section 3.1). Receiving a STOP_SENDING frame for a locally-initiated stream that has not yet been created MUST be treated as a connection error of type STREAM_STATE_ERROR. An endpoint that receives a STOP_SENDING frame for a receive-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STOP_SENDING frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Application Error Code (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
STOP_SENDING frames contain the following fields:
- - -The CRYPTO frame (type=0x06) is used to transmit cryptographic handshake messages. It can be sent in all packet types except 0-RTT. The CRYPTO frame offers the cryptographic protocol an in-order stream of bytes. CRYPTO frames are functionally identical to STREAM frames, except that they do not bear a stream identifier; they are not flow controlled; and they do not carry markers for optional offset, optional length, and the end of the stream.
-The CRYPTO frame is as follows:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Offset (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Crypto Data (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 19: CRYPTO Frame Format
-CRYPTO frames contain the following fields:
- - -There is a separate flow of cryptographic handshake data in each encryption level, each of which starts at an offset of 0. This implies that each encryption level is treated as a separate CRYPTO stream of data.
-Unlike STREAM frames, which include a Stream ID indicating to which stream the data belongs, the CRYPTO frame carries data for a single stream per encryption level. The stream does not have an explicit end, so CRYPTO frames do not have a FIN bit.
-A server sends a NEW_TOKEN frame (type=0x07) to provide the client with a token to send in the header of an Initial packet for a future connection.
-The NEW_TOKEN frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Token Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Token (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
NEW_TOKEN frames contain the following fields:
- - -An endpoint might receive multiple NEW_TOKEN frames that contain the same token value. Endpoints are responsible for discarding duplicate values, which might be used to link connection attempts; see Section 8.1.2.
-Clients MUST NOT send NEW_TOKEN frames. Servers MUST treat receipt of a NEW_TOKEN frame as a connection error of type PROTOCOL_VIOLATION.
-STREAM frames implicitly create a stream and carry stream data. The STREAM frame takes the form 0b00001XXX (or the set of values from 0x08 to 0x0f). The value of the three low-order bits of the frame type determines the fields that are present in the frame.
- - -An endpoint that receives a STREAM frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STREAM frames are as follows:
- - -- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Offset (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [Length (i)] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream Data (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
Figure 20: STREAM Frame Format
-STREAM frames contain the following fields:
- - -When a Stream Data field has a length of 0, the offset in the STREAM frame is the offset of the next byte that would be sent.
-The first byte in the stream has an offset of 0. The largest offset delivered on a stream - the sum of the offset and data length - cannot exceed 2^62-1, as it is not possible to provide flow control credit for that data. Receipt of a frame that exceeds this limit will be treated as a connection error of type FLOW_CONTROL_ERROR.
-The MAX_DATA frame (type=0x10) is used in flow control to inform the peer of the maximum amount of data that can be sent on the connection as a whole.
-The MAX_DATA frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Maximum Data (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
MAX_DATA frames contain the following fields:
- - -All data sent in STREAM frames counts toward this limit. The sum of the largest received offsets on all streams - including streams in terminal states - MUST NOT exceed the value advertised by a receiver. An endpoint MUST terminate a connection with a FLOW_CONTROL_ERROR error if it receives more data than the maximum data value that it has sent, unless this is a result of a change in the initial limits (see Section 7.3.1).
-The MAX_STREAM_DATA frame (type=0x11) is used in flow control to inform a peer of the maximum amount of data that can be sent on a stream.
-A MAX_STREAM_DATA frame can be sent for streams in the Recv state (see Section 3.1). Receiving a MAX_STREAM_DATA frame for a locally-initiated stream that has not yet been created MUST be treated as a connection error of type STREAM_STATE_ERROR. An endpoint that receives a MAX_STREAM_DATA frame for a receive-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The MAX_STREAM_DATA frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Maximum Stream Data (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
MAX_STREAM_DATA frames contain the following fields:
- - -When counting data toward this limit, an endpoint accounts for the largest received offset of data that is sent or received on the stream. Loss or reordering can mean that the largest received offset on a stream can be greater than the total size of data received on that stream. Receiving STREAM frames might not increase the largest received offset.
-The data sent on a stream MUST NOT exceed the largest maximum stream data value advertised by the receiver. An endpoint MUST terminate a connection with a FLOW_CONTROL_ERROR error if it receives more data than the largest maximum stream data that it has sent for the affected stream, unless this is a result of a change in the initial limits (see Section 7.3.1).
-The MAX_STREAMS frames (type=0x12 and 0x13) inform the peer of the cumulative number of streams of a given type it is permitted to open. A MAX_STREAMS frame with a type of 0x12 applies to bidirectional streams, and a MAX_STREAMS frame with a type of 0x13 applies to unidirectional streams.
-The MAX_STREAMS frames are as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Maximum Streams (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
MAX_STREAMS frames contain the following fields:
- - -Loss or reordering can cause a MAX_STREAMS frame to be received which states a lower stream limit than an endpoint has previously received. MAX_STREAMS frames which do not increase the stream limit MUST be ignored.
-An endpoint MUST NOT open more streams than permitted by the current stream limit set by its peer. For instance, a server that receives a unidirectional stream limit of 3 is permitted to open stream 3, 7, and 11, but not stream 15. An endpoint MUST terminate a connection with a STREAM_LIMIT_ERROR error if a peer opens more streams than was permitted.
-Note that these frames (and the corresponding transport parameters) do not describe the number of streams that can be opened concurrently. The limit includes streams that have been closed as well as those that are open.
-A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes to send data, but is unable to due to connection-level flow control (see Section 4). DATA_BLOCKED frames can be used as input to tuning of flow control algorithms (see Section 4.2).
-The DATA_BLOCKED frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Data Limit (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
DATA_BLOCKED frames contain the following fields:
- - -A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it wishes to send data, but is unable to due to stream-level flow control. This frame is analogous to DATA_BLOCKED (Section 19.12).
-An endpoint that receives a STREAM_DATA_BLOCKED frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STREAM_DATA_BLOCKED frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream ID (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream Data Limit (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
STREAM_DATA_BLOCKED frames contain the following fields:
- - -A sender SHOULD send a STREAMS_BLOCKED frame (type=0x16 or 0x17) when it wishes to open a stream, but is unable to due to the maximum stream limit set by its peer (see Section 19.11). A STREAMS_BLOCKED frame of type 0x16 is used to indicate reaching the bidirectional stream limit, and a STREAMS_BLOCKED frame of type 0x17 indicates reaching the unidirectional stream limit.
-A STREAMS_BLOCKED frame does not open the stream, but informs the peer that a new stream was needed and the stream limit prevented the creation of the stream.
-The STREAMS_BLOCKED frames are as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Stream Limit (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
STREAMS_BLOCKED frames contain the following fields:
- - -An endpoint sends a NEW_CONNECTION_ID frame (type=0x18) to provide its peer with alternative connection IDs that can be used to break linkability when migrating connections (see Section 9.5).
-The NEW_CONNECTION_ID frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Sequence Number (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Retire Prior To (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Length (8) | | -+-+-+-+-+-+-+-+-+ Connection ID (8..160) + -| ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| | -+ + -| | -+ Stateless Reset Token (128) + -| | -+ + -| | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
NEW_CONNECTION_ID frames contain the following fields:
- - -An endpoint MUST NOT send this frame if it currently requires that its peer send packets with a zero-length Destination Connection ID. Changing the length of a connection ID to or from zero-length makes it difficult to identify when the value of the connection ID changed. An endpoint that is sending packets with a zero-length Destination Connection ID MUST treat receipt of a NEW_CONNECTION_ID frame as a connection error of type PROTOCOL_VIOLATION.
-Transmission errors, timeouts and retransmissions might cause the same NEW_CONNECTION_ID frame to be received multiple times. Receipt of the same frame multiple times MUST NOT be treated as a connection error. A receiver can use the sequence number supplied in the NEW_CONNECTION_ID frame to identify new connection IDs from old ones.
-If an endpoint receives a NEW_CONNECTION_ID frame that repeats a previously issued connection ID with a different Stateless Reset Token or a different sequence number, or if a sequence number is used for different connection IDs, the endpoint MAY treat that receipt as a connection error of type PROTOCOL_VIOLATION.
-The Retire Prior To field is a request for the peer to retire all connection IDs with a sequence number less than the specified value. This includes the initial and preferred_address transport parameter connection IDs. The peer SHOULD retire the corresponding connection IDs and send the corresponding RETIRE_CONNECTION_ID frames in a timely manner.
-The Retire Prior To field MUST be less than or equal to the Sequence Number field. Receiving a value greater than the Sequence Number MUST be treated as a connection error of type PROTOCOL_VIOLATION.
-Once a sender indicates a Retire Prior To value, smaller values sent in subsequent NEW_CONNECTION_ID frames have no effect. A receiver MUST ignore any Retire Prior To fields that do not increase the largest received Retire Prior To value.
-An endpoint sends a RETIRE_CONNECTION_ID frame (type=0x19) to indicate that it will no longer use a connection ID that was issued by its peer. This may include the connection ID provided during the handshake. Sending a RETIRE_CONNECTION_ID frame also serves as a request to the peer to send additional connection IDs for future use (see Section 5.1). New connection IDs can be delivered to a peer using the NEW_CONNECTION_ID frame (Section 19.15).
-Retiring a connection ID invalidates the stateless reset token associated with that connection ID.
-The RETIRE_CONNECTION_ID frame is as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Sequence Number (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
RETIRE_CONNECTION_ID frames contain the following fields:
- - -Receipt of a RETIRE_CONNECTION_ID frame containing a sequence number greater than any previously sent to the peer MAY be treated as a connection error of type PROTOCOL_VIOLATION.
-The sequence number specified in a RETIRE_CONNECTION_ID frame MUST NOT refer to the Destination Connection ID field of the packet in which the frame is contained. The peer MAY treat this as a connection error of type PROTOCOL_VIOLATION.
-An endpoint cannot send this frame if it was provided with a zero-length connection ID by its peer. An endpoint that provides a zero-length connection ID MUST treat receipt of a RETIRE_CONNECTION_ID frame as a connection error of type PROTOCOL_VIOLATION.
-Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check reachability to the peer and for path validation during connection migration.
-The PATH_CHALLENGE frames are as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| | -+ Data (64) + -| | -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
PATH_CHALLENGE frames contain the following fields:
- - -A PATH_CHALLENGE frame containing 8 bytes that are hard to guess is sufficient to ensure that it is easier to receive the packet than it is to guess the value correctly.
-The recipient of this frame MUST generate a PATH_RESPONSE frame (Section 19.18) containing the same Data.
-The PATH_RESPONSE frame (type=0x1b) is sent in response to a PATH_CHALLENGE frame. Its format is identical to the PATH_CHALLENGE frame (Section 19.17).
-If the content of a PATH_RESPONSE frame does not match the content of a PATH_CHALLENGE frame previously sent by the endpoint, the endpoint MAY generate a connection error of type PROTOCOL_VIOLATION.
-An endpoint sends a CONNECTION_CLOSE frame (type=0x1c or 0x1d) to notify its peer that the connection is being closed. The CONNECTION_CLOSE with a frame type of 0x1c is used to signal errors at only the QUIC layer, or the absence of errors (with the NO_ERROR code). The CONNECTION_CLOSE frame with a type of 0x1d is used to signal an error with the application that uses QUIC.
-If there are open streams that haven’t been explicitly closed, they are implicitly closed when the connection is closed.
-The CONNECTION_CLOSE frames are as follows:
-- 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Error Code (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| [ Frame Type (i) ] ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Reason Phrase Length (i) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -| Reason Phrase (*) ... -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ --
CONNECTION_CLOSE frames contain the following fields:
- - -QUIC frames do not use a self-describing encoding. An endpoint therefore needs to understand the syntax of all frames before it can successfully process a packet. This allows for efficient encoding of frames, but it means that an endpoint cannot send a frame of a type that is unknown to its peer.
-An extension to QUIC that wishes to use a new type of frame MUST first ensure that a peer is able to understand the frame. An endpoint can use a transport parameter to signal its willingness to receive one or more extension frame types with the one transport parameter.
-Extension frames MUST be congestion controlled and MUST cause an ACK frame to be sent. The exception is extension frames that replace or supplement the ACK frame. Extension frames are not included in flow control unless specified in the extension.
-An IANA registry is used to manage the assignment of frame types; see Section 22.2.
-QUIC error codes are 62-bit unsigned integers.
-This section lists the defined QUIC transport error codes that may be used in a CONNECTION_CLOSE frame. These errors apply to the entire connection.
- - -See Section 22.3 for details of registering new error codes.
-In defining these error codes, several principles are applied. Error conditions that might require specific action on the part of a recipient are given unique codes. Errors that represent common conditions are given specific codes. Absent either of these conditions, error codes are used to identify a general function of the stack, like flow control or transport parameter handling. Finally, generic errors are provided for conditions where implementations are unable or unwilling to use more specific codes.
-Application protocol error codes are 62-bit unsigned integers, but the management of application error codes is left to application protocols. Application protocol error codes are used for the RESET_STREAM frame (Section 19.4), the STOP_SENDING frame (Section 19.5), and the CONNECTION_CLOSE frame with a type of 0x1d (Section 19.19).
-As an encrypted and authenticated transport QUIC provides a range of protections against denial of service. Once the cryptographic handshake is complete, QUIC endpoints discard most packets that are not authenticated, greatly limiting the ability of an attacker to interfere with existing connections.
-Once a connection is established QUIC endpoints might accept some unauthenticated ICMP packets (see Section 14.2), but the use of these packets is extremely limited. The only other type of packet that an endpoint might accept is a stateless reset (Section 10.4) which relies on the token being kept secret until it is used.
-During the creation of a connection, QUIC only provides protection against attack from off the network path. All QUIC packets contain proof that the recipient saw a preceding packet from its peer.
-The first mechanism used is the source and destination connection IDs, which are required to match those set by a peer. Except for an Initial and stateless reset packets, an endpoint only accepts packets that include a destination connection that matches a connection ID the endpoint previously chose. This is the only protection offered for Version Negotiation packets.
-The destination connection ID in an Initial packet is selected by a client to be unpredictable, which serves an additional purpose. The packets that carry the cryptographic handshake are protected with a key that is derived from this connection ID and salt specific to the QUIC version. This allows endpoints to use the same process for authenticating packets that they receive as they use after the cryptographic handshake completes. Packets that cannot be authenticated are discarded. Protecting packets in this fashion provides a strong assurance that the sender of the packet saw the Initial packet and understood it.
-These protections are not intended to be effective against an attacker that is able to receive QUIC packets prior to the connection being established. Such an attacker can potentially send packets that will be accepted by QUIC endpoints. This version of QUIC attempts to detect this sort of attack, but it expects that endpoints will fail to establish a connection rather than recovering. For the most part, the cryptographic handshake protocol [QUIC-TLS] is responsible for detecting tampering during the handshake.
-Endpoints are permitted to use other methods to detect and attempt to recover from interference with the handshake. Invalid packets may be identified and discarded using other methods, but no specific method is mandated in this document.
-An attacker might be able to receive an address validation token (Section 8) from a server and then release the IP address it used to acquire that token. At a later time, the attacker may initiate a 0-RTT connection with a server by spoofing this same address, which might now address a different (victim) endpoint. The attacker can thus potentially cause the server to send an initial congestion window’s worth of data towards the victim.
-Servers SHOULD provide mitigations for this attack by limiting the usage and lifetime of address validation tokens (see Section 8.1.2).
-An endpoint that acknowledges packets it has not received might cause a congestion controller to permit sending at rates beyond what the network supports. An endpoint MAY skip packet numbers when sending packets to detect this behavior. An endpoint can then immediately close the connection with a connection error of type PROTOCOL_VIOLATION (see Section 10.3).
-The attacks commonly known as Slowloris [SLOWLORIS] try to keep many connections to the target endpoint open and hold them open as long as possible. These attacks can be executed against a QUIC endpoint by generating the minimum amount of activity necessary to avoid being closed for inactivity. This might involve sending small amounts of data, gradually opening flow control windows in order to control the sender rate, or manufacturing ACK frames that simulate a high loss rate.
-QUIC deployments SHOULD provide mitigations for the Slowloris attacks, such as increasing the maximum number of clients the server will allow, limiting the number of connections a single IP address is allowed to make, imposing restrictions on the minimum transfer speed a connection is allowed to have, and restricting the length of time an endpoint is allowed to stay connected.
-An adversarial sender might intentionally send fragments of stream data in order to cause disproportionate receive buffer memory commitment and/or creation of a large and inefficient data structure.
-An adversarial receiver might intentionally not acknowledge packets containing stream data in order to force the sender to store the unacknowledged stream data for retransmission.
-The attack on receivers is mitigated if flow control windows correspond to available memory. However, some receivers will over-commit memory and advertise flow control offsets in the aggregate that exceed actual available memory. The over-commitment strategy can lead to better performance when endpoints are well behaved, but renders endpoints vulnerable to the stream fragmentation attack.
-QUIC deployments SHOULD provide mitigations against stream fragmentation attacks. Mitigations could consist of avoiding over-committing memory, limiting the size of tracking data structures, delaying reassembly of STREAM frames, implementing heuristics based on the age and duration of reassembly holes, or some combination.
-An adversarial endpoint can open lots of streams, exhausting state on an endpoint. The adversarial endpoint could repeat the process on a large number of connections, in a manner similar to SYN flooding attacks in TCP.
-Normally, clients will open streams sequentially, as explained in Section 2.1. However, when several streams are initiated at short intervals, loss or reordering may cause STREAM frames that open streams to be received out of sequence. On receiving a higher-numbered stream ID, a receiver is required to open all intervening streams of the same type (see Section 3.2). Thus, on a new connection, opening stream 4000000 opens 1 million and 1 client-initiated bidirectional streams.
-The number of active streams is limited by the initial_max_streams_bidi and initial_max_streams_uni transport parameters, as explained in Section 4.5. If chosen judiciously, these limits mitigate the effect of the stream commitment attack. However, setting the limit too low could affect performance when applications expect to open large number of streams.
-QUIC and TLS both contain messages that have legitimate uses in some contexts, but that can be abused to cause a peer to expend processing resources without having any observable impact on the state of the connection.
-Messages can also be used to change and revert state in small or inconsequential ways, such as by sending small increments to flow control limits.
-If processing costs are disproportionately large in comparison to bandwidth consumption or effect on state, then this could allow a malicious peer to exhaust processing capacity.
-While there are legitimate uses for all messages, implementations SHOULD track cost of processing relative to progress and treat excessive quantities of any non-productive packets as indicative of an attack. Endpoints MAY respond to this condition with a connection error, or by dropping packets.
-An on-path attacker could manipulate the value of ECN codepoints in the IP header to influence the sender’s rate. [RFC3168] discusses manipulations and their effects in more detail.
-An on-the-side attacker can duplicate and send packets with modified ECN codepoints to affect the sender’s rate. If duplicate packets are discarded by a receiver, an off-path attacker will need to race the duplicate packet against the original to be successful in this attack. Therefore, QUIC endpoints ignore the ECN codepoint field on an IP packet unless at least one QUIC packet in that IP packet is successfully processed; see Section 13.4.
-Stateless resets create a possible denial of service attack analogous to a TCP reset injection. This attack is possible if an attacker is able to cause a stateless reset token to be generated for a connection with a selected connection ID. An attacker that can cause this token to be generated can reset an active connection with the same connection ID.
-If a packet can be routed to different instances that share a static key, for example by changing an IP address or port, then an attacker can cause the server to send a stateless reset. To defend against this style of denial service, endpoints that share a static key for stateless reset (see Section 10.4.2) MUST be arranged so that packets with a given connection ID always arrive at an instance that has connection state, unless that connection is no longer active.
-In the case of a cluster that uses dynamic load balancing, it’s possible that a change in load balancer configuration could happen while an active instance retains connection state; even if an instance retains connection state, the change in routing and resulting stateless reset will result in the connection being terminated. If there is no chance in the packet being routed to the correct instance, it is better to send a stateless reset than wait for connections to time out. However, this is acceptable only if the routing cannot be influenced by an attacker.
-This document defines QUIC Version Negotiation packets Section 6, which can be used to negotiate the QUIC version used between two endpoints. However, this document does not specify how this negotiation will be performed between this version and subsequent future versions. In particular, Version Negotiation packets do not contain any mechanism to prevent version downgrade attacks. Future versions of QUIC that use Version Negotiation packets MUST define a mechanism that is robust against version downgrade attacks.
-Deployments should limit the ability of an attacker to target a new connection to a particular server instance. This means that client-controlled fields, such as the initial Destination Connection ID used on Initial and 0-RTT packets SHOULD NOT be used by themselves to make routing decisions. Ideally, routing decisions are made independently of client-selected values; a Source Connection ID can be selected to route later packets to the same server.
-IANA [SHALL add/has added] a registry for “QUIC Transport Parameters” under a “QUIC Protocol” heading.
-The “QUIC Transport Parameters” registry governs a 16-bit space. This space is split into two spaces that are governed by different policies. Values with the first byte in the range 0x00 to 0xfe (in hexadecimal) are assigned via the Specification Required policy [RFC8126]. Values with the first byte 0xff are reserved for Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -The nominated expert(s) verify that a specification exists and is readily accessible. Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are shown in Table 6.
- - -| Value | -Parameter Name | -Specification | -
|---|---|---|
| 0x0000 | -original_connection_id | -Section 18.2 | -
| 0x0001 | -idle_timeout | -Section 18.2 | -
| 0x0002 | -stateless_reset_token | -Section 18.2 | -
| 0x0003 | -max_packet_size | -Section 18.2 | -
| 0x0004 | -initial_max_data | -Section 18.2 | -
| 0x0005 | -initial_max_stream_data_bidi_local | -Section 18.2 | -
| 0x0006 | -initial_max_stream_data_bidi_remote | -Section 18.2 | -
| 0x0007 | -initial_max_stream_data_uni | -Section 18.2 | -
| 0x0008 | -initial_max_streams_bidi | -Section 18.2 | -
| 0x0009 | -initial_max_streams_uni | -Section 18.2 | -
| 0x000a | -ack_delay_exponent | -Section 18.2 | -
| 0x000b | -max_ack_delay | -Section 18.2 | -
| 0x000c | -disable_active_migration | -Section 18.2 | -
| 0x000d | -preferred_address | -Section 18.2 | -
| 0x000e | -active_connection_id_limit | -Section 18.2 | -
Additionally, each value of the format 31 * N + 27 for integer values of N (that is, 27, 58, 89, …) MUST NOT be assigned by IANA.
-IANA [SHALL add/has added] a registry for “QUIC Frame Types” under a “QUIC Protocol” heading.
-The “QUIC Frame Types” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -The nominated expert(s) verify that a specification exists and is readily accessible. Specifications for new registrations need to describe the means by which an endpoint might determine that it can send the identified type of frame. An accompanying transport parameter registration (see Section 22.1) is expected for most registrations. The specification needs to describe the format and assigned semantics of any fields in the frame.
-Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are tabulated in Table 3.
-IANA [SHALL add/has added] a registry for “QUIC Transport Error Codes” under a “QUIC Protocol” heading.
-The “QUIC Transport Error Codes” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -The nominated expert(s) verify that a specification exists and is readily accessible. Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are shown in Table 7.
- - -| Value | -Error | -Description | -Specification | -
|---|---|---|---|
| 0x0 | -NO_ERROR | -No error | -Section 20 | -
| 0x1 | -INTERNAL_ERROR | -Implementation error | -Section 20 | -
| 0x2 | -SERVER_BUSY | -Server currently busy | -Section 20 | -
| 0x3 | -FLOW_CONTROL_ERROR | -Flow control error | -Section 20 | -
| 0x4 | -STREAM_LIMIT_ERROR | -Too many streams opened | -Section 20 | -
| 0x5 | -STREAM_STATE_ERROR | -Frame received in invalid stream state | -Section 20 | -
| 0x6 | -FINAL_SIZE_ERROR | -Change to final size | -Section 20 | -
| 0x7 | -FRAME_ENCODING_ERROR | -Frame encoding error | -Section 20 | -
| 0x8 | -TRANSPORT_PARAMETER_ERROR | -Error in transport parameters | -Section 20 | -
| 0xA | -PROTOCOL_VIOLATION | -Generic protocol violation | -Section 20 | -
| 0xD | -CRYPTO_BUFFER_EXCEEDED | -CRYPTO data buffer overflowed | -Section 20 | -
| [DPLPMTUD] | --Fairhurst, G., Jones, T., Tuexen, M., Ruengeler, I. and T. Voelker, "Packetization Layer Path MTU Discovery for Datagram Transports", Internet-Draft draft-ietf-tsvwg-datagram-plpmtud-08, June 2019. | -
| [QUIC-RECOVERY] | --Iyengar, J. and I. Swett, "QUIC Loss Detection and Congestion Control", Internet-Draft draft-ietf-quic-recovery, October 2019. | -
| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using Transport Layer Security (TLS) to Secure QUIC", Internet-Draft draft-ietf-quic-tls, October 2019. | -
| [RFC1191] | --Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, DOI 10.17487/RFC1191, November 1990. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC3168] | --Ramakrishnan, K., Floyd, S. and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, DOI 10.17487/RFC3168, September 2001. | -
| [RFC3629] | --Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November 2003. | -
| [RFC4086] | --Eastlake 3rd, D., Schiller, J. and S. Crocker, "Randomness Requirements for Security", BCP 106, RFC 4086, DOI 10.17487/RFC4086, June 2005. | -
| [RFC5116] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
| [RFC6437] | --Amante, S., Carpenter, B., Jiang, S. and J. Rajahalme, "IPv6 Flow Label Specification", RFC 6437, DOI 10.17487/RFC6437, November 2011. | -
| [RFC8085] | --Eggert, L., Fairhurst, G. and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, March 2017. | -
| [RFC8126] | --Cotton, M., Leiba, B. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 8126, DOI 10.17487/RFC8126, June 2017. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC8201] | --McCann, J., Deering, S., Mogul, J. and R. Hinden, "Path MTU Discovery for IP version 6", STD 87, RFC 8201, DOI 10.17487/RFC8201, July 2017. | -
| [RFC8311] | --Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018. | -
| [TLS13] | --Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018. | -
| [EARLY-DESIGN] | --Roskind, J., "QUIC: Multiplexed Transport Over UDP", December 2013. | -
| [HTTP2] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| [QUIC-INVARIANTS] | --Thomson, M., "Version-Independent Properties of QUIC", Internet-Draft draft-ietf-quic-invariants, October 2019. | -
| [QUIC-MANAGEABILITY] | --Kuehlewind, M. and B. Trammell, "Manageability of the QUIC Transport Protocol", Internet-Draft draft-ietf-quic-manageability-05, July 2019. | -
| [RFC1812] | --Baker, F., "Requirements for IP Version 4 Routers", RFC 1812, DOI 10.17487/RFC1812, June 1995. | -
| [RFC2018] | --Mathis, M., Mahdavi, J., Floyd, S. and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, DOI 10.17487/RFC2018, October 1996. | -
| [RFC2104] | --Krawczyk, H., Bellare, M. and R. Canetti, "HMAC: Keyed-Hashing for Message Authentication", RFC 2104, DOI 10.17487/RFC2104, February 1997. | -
| [RFC2360] | --Scott, G., "Guide for Internet Standards Writers", BCP 22, RFC 2360, DOI 10.17487/RFC2360, June 1998. | -
| [RFC4303] | --Kent, S., "IP Encapsulating Security Payload (ESP)", RFC 4303, DOI 10.17487/RFC4303, December 2005. | -
| [RFC4443] | --Conta, A., Deering, S. and M. Gupta, "Internet Control Message Protocol (ICMPv6) for the Internet Protocol Version 6 (IPv6) Specification", STD 89, RFC 4443, DOI 10.17487/RFC4443, March 2006. | -
| [RFC4787] | --Audet, F. and C. Jennings, "Network Address Translation (NAT) Behavioral Requirements for Unicast UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January 2007. | -
| [RFC5681] | --Allman, M., Paxson, V. and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009. | -
| [RFC5869] | --Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand Key Derivation Function (HKDF)", RFC 5869, DOI 10.17487/RFC5869, May 2010. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC8200] | --Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", STD 86, RFC 8200, DOI 10.17487/RFC8200, July 2017. | -
| [SLOWLORIS] | --RSnake Hansen, R., "Welcome to Slowloris...", June 2009. | -
The following pseudo-code shows how an implementation can decode packet numbers after header protection has been removed.
--DecodePacketNumber(largest_pn, truncated_pn, pn_nbits): - expected_pn = largest_pn + 1 - pn_win = 1 << pn_nbits - pn_hwin = pn_win / 2 - pn_mask = pn_win - 1 - // The incoming packet number should be greater than - // expected_pn - pn_hwin and less than or equal to - // expected_pn + pn_hwin - // - // This means we can't just strip the trailing bits from - // expected_pn and add the truncated_pn because that might - // yield a value outside the window. - // - // The following code calculates a candidate value and - // makes sure it's within the packet number window. - candidate_pn = (expected_pn & ~pn_mask) | truncated_pn - if candidate_pn <= expected_pn - pn_hwin: - return candidate_pn + pn_win - // Note the extra check for underflow when candidate_pn - // is near zero. - if candidate_pn > expected_pn + pn_hwin and - candidate_pn > pn_win: - return candidate_pn - pn_win - return candidate_pn --
Issue and pull request numbers are listed with a leading octothorp.
-Substantial editorial reorganization; no technical changes.
-Special thanks are due to the following for helping shape pre-IETF QUIC and its deployment: Chris Bentzel, Misha Efimov, Roberto Peon, Alistair Riddoch, Siddharth Vijayakrishnan, and Assar Westerlund.
-This document has benefited immensely from various private discussions and public ones on the quic@ietf.org and proto-quic@chromium.org mailing lists. Our thanks to all.
-The original authors of this specification were Ryan Hamilton, Jana Iyengar, Ian Swett, and Alyssa Wilk.
-The original design and rationale behind this protocol draw significantly from work by Jim Roskind [EARLY-DESIGN]. In alphabetical order, the contributors to the pre-IETF QUIC project at Google are: Britt Cyr, Jeremy Dorfman, Ryan Hamilton, Jana Iyengar, Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley, Jim Roskind, Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti, Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale Worley, Fan Yang, Dan Zhang, Daniel Ziegler.
-View saved issues, - or the latest GitHub issues - and pull requests.
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| draft-ietf-quic-invariants | -html | -plain text | -diff with master | -- diff with last submission | -
| draft-ietf-quic-qpack | -html | -plain text | -diff with master | -- diff with last submission | -
| draft-ietf-quic-recovery | -html | -plain text | -diff with master | -- diff with last submission | -
| draft-ietf-quic-tls | -html | -plain text | -diff with master | -- diff with last submission | -
| draft-ietf-quic-transport | -html | -plain text | -diff with master | -- diff with last submission | -
| QUIC | -M. Bishop, Ed. | -
| Internet-Draft | -Akamai | -
| Intended status: Standards Track | -November 12, 2019 | -
| Expires: May 15, 2020 | -- |
Hypertext Transfer Protocol Version 3 (HTTP/3)
- draft-ietf-quic-http-latest
The QUIC transport protocol has several features that are desirable in a transport for HTTP, such as stream multiplexing, per-stream flow control, and low-latency connection establishment. This document describes a mapping of HTTP semantics over QUIC. This document also identifies HTTP/2 features that are subsumed by QUIC, and describes how HTTP/2 extensions can be ported to HTTP/3.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-http.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
+/* Provide table alignment support. We can't use the alignX classes above + since they do unwanted things with caption and other styling. */ +table.right { + margin-left: auto; + margin-right: 0; +} +table.center { + margin-left: auto; + margin-right: auto; +} +table.left { + margin-left: 0; + margin-right: auto; +} - - -| Internet-Draft | +HTTP/3 | +January 2020 | +
| Bishop | +Expires 16 July 2020 | +[Page] | +
The QUIC transport protocol has several features that are desirable in a +transport for HTTP, such as stream multiplexing, per-stream flow control, and +low-latency connection establishment. This document describes a mapping of HTTP +semantics over QUIC. This document also identifies HTTP/2 features that are +subsumed by QUIC, and describes how HTTP/2 extensions can be ported to HTTP/3.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +https://mailarchive.ietf.org/arch/search/?email_list=quic.¶
+Working Group information can be found at https://github.com/quicwg; source +code and issues list for this draft can be found at +https://github.com/quicwg/base-drafts/labels/-http.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+HTTP semantics are used for a broad range of services on the Internet. These +semantics have commonly been used with two different TCP mappings, HTTP/1.1 and +HTTP/2. HTTP/3 supports the same semantics over a new transport protocol, QUIC.¶
+HTTP/1.1 is a TCP mapping which uses whitespace-delimited text fields to convey +HTTP messages. While these exchanges are human-readable, using whitespace for +message formatting leads to parsing difficulties and workarounds to be tolerant +of variant behavior. Because each connection can transfer only a single HTTP +request or response at a time in each direction, multiple parallel TCP +connections are often used, reducing the ability of the congestion controller to +accurately manage traffic between endpoints.¶
+HTTP/2 introduced a binary framing and multiplexing layer to improve latency +without modifying the transport layer. However, because the parallel nature of +HTTP/2's multiplexing is not visible to TCP's loss recovery mechanisms, a lost +or reordered packet causes all active transactions to experience a stall +regardless of whether that transaction was impacted by the lost packet.¶
+The QUIC transport protocol incorporates stream multiplexing and per-stream flow +control, similar to that provided by the HTTP/2 framing layer. By providing +reliability at the stream level and congestion control across the entire +connection, it has the capability to improve the performance of HTTP compared to +a TCP mapping. QUIC also incorporates TLS 1.3 at the transport layer, offering +comparable security to running TLS over TCP, with the improved connection setup +latency of TCP Fast Open [RFC7413].¶
+This document defines a mapping of HTTP semantics over the QUIC transport +protocol, drawing heavily on the design of HTTP/2. While delegating stream +lifetime and flow control issues to QUIC, a similar binary framing is used on +each stream. Some HTTP/2 features are subsumed by QUIC, while other features are +implemented atop QUIC.¶
+QUIC is described in [QUIC-TRANSPORT]. For a full description of HTTP/2, see +[HTTP2].¶
+HTTP/3 provides a transport for HTTP semantics using the QUIC transport protocol +and an internal framing layer similar to HTTP/2.¶
+Once a client knows that an HTTP/3 server exists at a certain endpoint, it opens +a QUIC connection. QUIC provides protocol negotiation, stream-based +multiplexing, and flow control. An HTTP/3 endpoint can be discovered using HTTP +Alternative Services; this process is described in greater detail in +Section 3.2.¶
+Within each stream, the basic unit of HTTP/3 communication is a frame +(Section 7.2). Each frame type serves a different purpose. For example, HEADERS +and DATA frames form the basis of HTTP requests and responses +(Section 4.1).¶
+Multiplexing of requests is performed using the QUIC stream abstraction, +described in Section 2 of [QUIC-TRANSPORT]. Each request-response pair +consumes a single QUIC stream. Streams are independent of each other, so one +stream that is blocked or suffers packet loss does not prevent progress on other +streams.¶
+Server push is an interaction mode introduced in HTTP/2 [HTTP2] which permits +a server to push a request-response exchange to a client in anticipation of the +client making the indicated request. This trades off network usage against a +potential latency gain. Several HTTP/3 frames are used to manage server push, +such as PUSH_PROMISE, DUPLICATE_PUSH, MAX_PUSH_ID, and CANCEL_PUSH.¶
+As in HTTP/2, request and response headers are compressed for transmission. +Because HPACK [HPACK] relies on in-order transmission of compressed +header blocks (a guarantee not provided by QUIC), HTTP/3 replaces HPACK with +QPACK [QPACK]. QPACK uses separate unidirectional streams to modify and track +header table state, while header blocks refer to the state of the table without +modifying it.¶
+The following sections provide a detailed overview of the connection lifecycle +and key concepts:¶
The details of the wire protocol and interactions with the transport are described in subsequent sections:
- - +The details of the wire protocol and interactions with the transport are +described in subsequent sections:¶
Additional resources are provided in the final sections:
- - +Additional resources are provided in the final sections:¶
The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Field definitions are given in Augmented Backus-Naur Form (ABNF), as defined in [RFC5234].
-This document uses the variable-length integer encoding from [QUIC-TRANSPORT].
-The following terms are used:
- +The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+Field definitions are given in Augmented Backus-Naur Form (ABNF), as defined in +[RFC5234].¶
+This document uses the variable-length integer encoding from +[QUIC-TRANSPORT].¶
+The following terms are used:¶
+The term “payload body” is defined in Section 3.3 of [RFC7230].
-Finally, the terms “gateway”, “intermediary”, “proxy”, and “tunnel” are defined in Section 2.3 of [RFC7230]. Intermediaries act as both client and server at different times.
-HTTP/3 uses the token “h3” to identify itself in ALPN and Alt-Svc. Only implementations of the final, published RFC can identify themselves as “h3”. Until such an RFC exists, implementations MUST NOT identify themselves using this string.
-Implementations of draft versions of the protocol MUST add the string “-“ and the corresponding draft number to the identifier. For example, draft-ietf-quic-http-01 is identified using the string “h3-01”.
-Non-compatible experiments that are based on these draft versions MUST append the string “-“ and an experiment name to the identifier. For example, an experimental implementation based on draft-ietf-quic-http-09 which reserves an extra stream for unsolicited transmission of 1980s pop music might identify itself as “h3-09-rickroll”. Note that any label MUST conform to the “token” syntax defined in Section 3.2.6 of [RFC7230]. Experimenters are encouraged to coordinate their experiments on the quic@ietf.org mailing list.
-An HTTP origin advertises the availability of an equivalent HTTP/3 endpoint via the Alt-Svc HTTP response header field or the HTTP/2 ALTSVC frame ([ALTSVC]), using the ALPN token defined in Section 3.3.
-For example, an origin could indicate in an HTTP response that HTTP/3 was available on UDP port 50781 at the same hostname by including the following header field:
+The term "payload body" is defined in Section 3.3 of [RFC7230].¶
+Finally, the terms "gateway", "intermediary", "proxy", and "tunnel" are defined +in Section 2.3 of [RFC7230]. Intermediaries act as both client and server at +different times.¶
+HTTP/3 uses the token "h3" to identify itself in ALPN and Alt-Svc. Only +implementations of the final, published RFC can identify themselves as "h3". +Until such an RFC exists, implementations MUST NOT identify themselves using +this string.¶
+Implementations of draft versions of the protocol MUST add the string "-" and +the corresponding draft number to the identifier. For example, +draft-ietf-quic-http-01 is identified using the string "h3-01".¶
+Non-compatible experiments that are based on these draft versions MUST append +the string "-" and an experiment name to the identifier. For example, an +experimental implementation based on draft-ietf-quic-http-09 which reserves an +extra stream for unsolicited transmission of 1980s pop music might identify +itself as "h3-09-rickroll". Note that any label MUST conform to the "token" +syntax defined in Section 3.2.6 of [RFC7230]. Experimenters are encouraged to +coordinate their experiments on the quic@ietf.org mailing list.¶
+An HTTP origin advertises the availability of an equivalent HTTP/3 endpoint via +the Alt-Svc HTTP response header field or the HTTP/2 ALTSVC frame +([ALTSVC]), using the ALPN token defined in +Section 3.3.¶
+For example, an origin could indicate in an HTTP response that HTTP/3 was +available on UDP port 50781 at the same hostname by including the following +header field:¶
+Alt-Svc: h3=":50781" --
On receipt of an Alt-Svc record indicating HTTP/3 support, a client MAY attempt to establish a QUIC connection to the indicated host and port and, if successful, send HTTP requests using the mapping described in this document.
-Connectivity problems (e.g. firewall blocking UDP) can result in QUIC connection establishment failure, in which case the client SHOULD continue using the existing connection or try another alternative endpoint offered by the origin.
-Servers MAY serve HTTP/3 on any UDP port, since an alternative always includes an explicit port.
-HTTP/3 relies on QUIC as the underlying transport. The QUIC version being used MUST use TLS version 1.3 or greater as its handshake protocol. HTTP/3 clients MUST indicate the target domain name during the TLS handshake. This may be done using the Server Name Indication (SNI) [RFC6066] extension to TLS or using some other mechanism.
-QUIC connections are established as described in [QUIC-TRANSPORT]. During connection establishment, HTTP/3 support is indicated by selecting the ALPN token “h3” in the TLS handshake. Support for other application-layer protocols MAY be offered in the same handshake.
-While connection-level options pertaining to the core QUIC protocol are set in the initial crypto handshake, HTTP/3-specific settings are conveyed in the SETTINGS frame. After the QUIC connection is established, a SETTINGS frame (Section 7.2.4) MUST be sent by each endpoint as the initial frame of their respective HTTP control stream (see Section 6.2.1).
-Once a connection exists to a server endpoint, this connection MAY be reused for requests with multiple different URI authority components. The client MAY send any requests for which the client considers the server authoritative.
-An authoritative HTTP/3 endpoint is typically discovered because the client has received an Alt-Svc record from the request’s origin which nominates the endpoint as a valid HTTP Alternative Service for that origin. As required by [RFC7838], clients MUST check that the nominated server can present a valid certificate for the origin before considering it authoritative. Clients MUST NOT assume that an HTTP/3 endpoint is authoritative for other origins without an explicit signal.
-Prior to making requests for an origin whose scheme is not “https,” the client MUST ensure the server is willing to serve that scheme. If the client intends to make requests for an origin whose scheme is “http”, this means that it MUST obtain a valid http-opportunistic response for the origin as described in [RFC8164] prior to making any such requests. Other schemes might define other mechanisms.
-A server that does not wish clients to reuse connections for a particular origin can indicate that it is not authoritative for a request by sending a 421 (Misdirected Request) status code in response to the request (see Section 9.1.2 of [HTTP2]).
-The considerations discussed in Section 9.1 of [HTTP2] also apply to the management of HTTP/3 connections.
-A client sends an HTTP request on a client-initiated bidirectional QUIC stream. A client MUST send only a single request on a given stream. A server sends zero or more non-final HTTP responses on the same stream as the request, followed by a single final HTTP response, as detailed below.
-An HTTP message (request or response) consists of:
- - -On receipt of an Alt-Svc record indicating HTTP/3 support, a client MAY attempt +to establish a QUIC connection to the indicated host and port and, if +successful, send HTTP requests using the mapping described in this document.¶
+Connectivity problems (e.g. firewall blocking UDP) can result in QUIC connection +establishment failure, in which case the client SHOULD continue using the +existing connection or try another alternative endpoint offered by the origin.¶
+Servers MAY serve HTTP/3 on any UDP port, since an alternative always includes +an explicit port.¶
+HTTP/3 relies on QUIC as the underlying transport. The QUIC version being used +MUST use TLS version 1.3 or greater as its handshake protocol. HTTP/3 clients +MUST indicate the target domain name during the TLS handshake. This may be done +using the Server Name Indication (SNI) [RFC6066] extension to TLS or using +some other mechanism.¶
+QUIC connections are established as described in [QUIC-TRANSPORT]. During +connection establishment, HTTP/3 support is indicated by selecting the ALPN +token "h3" in the TLS handshake. Support for other application-layer protocols +MAY be offered in the same handshake.¶
+While connection-level options pertaining to the core QUIC protocol are set in +the initial crypto handshake, HTTP/3-specific settings are conveyed in the +SETTINGS frame. After the QUIC connection is established, a SETTINGS frame +(Section 7.2.4) MUST be sent by each endpoint as the initial frame of their +respective HTTP control stream (see Section 6.2.1).¶
+Once a connection exists to a server endpoint, this connection MAY be reused for +requests with multiple different URI authority components. The client MAY send +any requests for which the client considers the server authoritative.¶
+An authoritative HTTP/3 endpoint is typically discovered because the client has +received an Alt-Svc record from the request's origin which nominates the +endpoint as a valid HTTP Alternative Service for that origin. As required by +[RFC7838], clients MUST check that the nominated server can present a valid +certificate for the origin before considering it authoritative. Clients MUST NOT +assume that an HTTP/3 endpoint is authoritative for other origins without an +explicit signal.¶
+Prior to making requests for an origin whose scheme is not "https," the client
+MUST ensure the server is willing to serve that scheme. If the client intends
+to make requests for an origin whose scheme is "http", this means that it MUST
+obtain a valid http-opportunistic response for the origin as described in
+[RFC8164] prior to making any such requests. Other schemes might define
+other mechanisms.¶
A server that does not wish clients to reuse connections for a particular origin +can indicate that it is not authoritative for a request by sending a 421 +(Misdirected Request) status code in response to the request (see Section 9.1.2 +of [HTTP2]).¶
+The considerations discussed in Section 9.1 of [HTTP2] also apply to the +management of HTTP/3 connections.¶
+A client sends an HTTP request on a client-initiated bidirectional QUIC stream. +A client MUST send only a single request on a given stream. A server sends zero +or more non-final HTTP responses on the same stream as the request, followed by +a single final HTTP response, as detailed below.¶
+An HTTP message (request or response) consists of:¶
+A server MAY send one or more PUSH_PROMISE frames (see Section 7.2.5) before, after, or interleaved with the frames of a response message. These PUSH_PROMISE frames are not part of the response; see Section 4.4 for more details.
-Frames of unknown types (Section 9), including reserved frames (Section 7.2.9) MAY be sent on a request or push stream before, after, or interleaved with other frames described in this section.
-The HEADERS and PUSH_PROMISE frames might reference updates to the QPACK dynamic table. While these updates are not directly part of the message exchange, they must be received and processed before the message can be consumed. See Section 4.1.1 for more details.
-The “chunked” transfer encoding defined in Section 4.1 of [RFC7230] MUST NOT be used.
-A response MAY consist of multiple messages when and only when one or more informational responses (1xx; see [RFC7231], Section 6.2) precede a final response to the same request. Non-final responses do not contain a payload body or trailers.
-If an endpoint receives an invalid sequence of frames on either a request or a push stream, it MUST respond with a connection error of type H3_FRAME_UNEXPECTED (Section 8). In particular, a DATA frame before any HEADERS frame, or a HEADERS or DATA frame after the trailing HEADERS frame is considered invalid.
-An HTTP request/response exchange fully consumes a bidirectional QUIC stream. After sending a request, a client MUST close the stream for sending. Unless using the CONNECT method (see Section 4.2), clients MUST NOT make stream closure dependent on receiving a response to their request. After sending a final response, the server MUST close the stream for sending. At this point, the QUIC stream is fully closed.
-When a stream is closed, this indicates the end of an HTTP message. Because some messages are large or unbounded, endpoints SHOULD begin processing partial HTTP messages once enough of the message has been received to make progress. If a client stream terminates without enough of the HTTP message to provide a complete response, the server SHOULD abort its response with the error code H3_REQUEST_INCOMPLETE.
-A server can send a complete response prior to the client sending an entire request if the response does not depend on any portion of the request that has not been sent and received. When the server does not need to receive the remainder of the request, it MAY abort reading the request stream with error code H3_EARLY_RESPONSE, send a complete response, and cleanly close the sending part of the stream. Clients MUST NOT discard complete responses as a result of having their request terminated abruptly, though clients can always discard responses at their discretion for other reasons. If the server sends a partial or complete response but does not abort reading, clients SHOULD continue sending the body of the request and close the stream normally.
-HTTP message headers carry information as a series of key-value pairs, called header fields. For a listing of registered HTTP header fields, see the “Message Header Field” registry maintained at https://www.iana.org/assignments/message-headers.
-Just as in previous versions of HTTP, header field names are strings of ASCII characters that are compared in a case-insensitive fashion. Properties of HTTP header field names and values are discussed in more detail in Section 3.2 of [RFC7230], though the wire rendering in HTTP/3 differs. As in HTTP/2, header field names MUST be converted to lowercase prior to their encoding. A request or response containing uppercase header field names MUST be treated as malformed (Section 4.1.3).
-As in HTTP/2, HTTP/3 uses special pseudo-header fields beginning with the ‘:’ character (ASCII 0x3a) to convey the target URI, the method of the request, and the status code for the response. These pseudo-header fields are defined in Section 8.1.2.3 and 8.1.2.4 of [HTTP2]. Pseudo-header fields are not HTTP header fields. Endpoints MUST NOT generate pseudo-header fields other than those defined in [HTTP2]. The restrictions on the use of pseudo-header fields in Section 8.1.2 of [HTTP2] also apply to HTTP/3. Messages which are considered malformed under these restrictions are handled as described in Section 4.1.3.
-HTTP/3 uses QPACK header compression as described in [QPACK], a variation of HPACK which allows the flexibility to avoid header-compression-induced head-of-line blocking. See that document for additional details.
-To allow for better compression efficiency, the cookie header field [RFC6265] MAY be split into separate header fields, each with one or more cookie-pairs, before compression. If a decompressed header list contains multiple cookie header fields, these MUST be concatenated before being passed into a non-HTTP/2, non-HTTP/3 context, as described in [HTTP2], Section 8.1.2.5.
-An HTTP/3 implementation MAY impose a limit on the maximum size of the message header it will accept on an individual HTTP message. A server that receives a larger header field list than it is willing to handle can send an HTTP 431 (Request Header Fields Too Large) status code [RFC6585]. A client can discard responses that it cannot process. The size of a header field list is calculated based on the uncompressed size of header fields, including the length of the name and value in bytes plus an overhead of 32 bytes for each header field.
-If an implementation wishes to advise its peer of this limit, it can be conveyed as a number of bytes in the SETTINGS_MAX_HEADER_LIST_SIZE parameter. An implementation which has received this parameter SHOULD NOT send an HTTP message header which exceeds the indicated size, as the peer will likely refuse to process it. However, because this limit is applied at each hop, messages below this limit are not guaranteed to be accepted.
-Clients can cancel requests by resetting and aborting the request stream with an error code of H3_REQUEST_CANCELLED (Section 8.1). When the client aborts reading a response, it indicates that this response is no longer of interest. Implementations SHOULD cancel requests by abruptly terminating any directions of a stream that are still open.
-When the server rejects a request without performing any application processing, it SHOULD abort its response stream with the error code H3_REQUEST_REJECTED. In this context, “processed” means that some data from the stream was passed to some higher layer of software that might have taken some action as a result. The client can treat requests rejected by the server as though they had never been sent at all, thereby allowing them to be retried later on a new connection. Servers MUST NOT use the H3_REQUEST_REJECTED error code for requests which were partially or fully processed. When a server abandons a response after partial processing, it SHOULD abort its response stream with the error code H3_REQUEST_CANCELLED.
-When a client resets a request with the error code H3_REQUEST_CANCELLED, a server MAY abruptly terminate the response using the error code H3_REQUEST_REJECTED if no processing was performed. Clients MUST NOT use the H3_REQUEST_REJECTED error code, except when a server has requested closure of the request stream with this error code.
-If a stream is cancelled after receiving a complete response, the client MAY ignore the cancellation and use the response. However, if a stream is cancelled after receiving a partial response, the response SHOULD NOT be used. Automatically retrying such requests is not possible, unless this is otherwise permitted (e.g., idempotent actions like GET, PUT, or DELETE).
-A malformed request or response is one that is an otherwise valid sequence of frames but is invalid due to the presence of extraneous frames, prohibited header fields, the absence of mandatory header fields, or the inclusion of uppercase header field names.
-A request or response that includes a payload body can include a content-length header field. A request or response is also malformed if the value of a content-length header field does not equal the sum of the DATA frame payload lengths that form the body. A response that is defined to have no payload, as described in Section 3.3.2 of [RFC7230] can have a non-zero content-length header field, even though no content is included in DATA frames.
-Intermediaries that process HTTP requests or responses (i.e., any intermediary not acting as a tunnel) MUST NOT forward a malformed request or response. Malformed requests or responses that are detected MUST be treated as a stream error (Section 8) of type H3_GENERAL_PROTOCOL_ERROR.
-For malformed requests, a server MAY send an HTTP response prior to closing or resetting the stream. Clients MUST NOT accept a malformed response. Note that these requirements are intended to protect against several types of common attacks against HTTP; they are deliberately strict because being permissive can expose implementations to these vulnerabilities.
-The pseudo-method CONNECT ([RFC7231], Section 4.3.6) is primarily used with HTTP proxies to establish a TLS session with an origin server for the purposes of interacting with “https” resources. In HTTP/1.x, CONNECT is used to convert an entire HTTP connection into a tunnel to a remote host. In HTTP/2, the CONNECT method is used to establish a tunnel over a single HTTP/2 stream to a remote host for similar purposes.
-A CONNECT request in HTTP/3 functions in the same manner as in HTTP/2. The request MUST be formatted as described in [HTTP2], Section 8.3. A CONNECT request that does not conform to these restrictions is malformed (see Section 4.1.3). The request stream MUST NOT be closed at the end of the request.
-A proxy that supports CONNECT establishes a TCP connection ([RFC0793]) to the server identified in the “:authority” pseudo-header field. Once this connection is successfully established, the proxy sends a HEADERS frame containing a 2xx series status code to the client, as defined in [RFC7231], Section 4.3.6.
-All DATA frames on the stream correspond to data sent or received on the TCP connection. Any DATA frame sent by the client is transmitted by the proxy to the TCP server; data received from the TCP server is packaged into DATA frames by the proxy. Note that the size and number of TCP segments is not guaranteed to map predictably to the size and number of HTTP DATA or QUIC STREAM frames.
-Once the CONNECT method has completed, only DATA frames are permitted to be sent on the stream. Extension frames MAY be used if specifically permitted by the definition of the extension. Receipt of any other frame type MUST be treated as a connection error of type H3_FRAME_UNEXPECTED.
-The TCP connection can be closed by either peer. When the client ends the request stream (that is, the receive stream at the proxy enters the “Data Recvd” state), the proxy will set the FIN bit on its connection to the TCP server. When the proxy receives a packet with the FIN bit set, it will terminate the send stream that it sends to the client. TCP connections which remain half-closed in a single direction are not invalid, but are often handled poorly by servers, so clients SHOULD NOT close a stream for sending while they still expect to receive data from the target of the CONNECT.
-A TCP connection error is signaled by abruptly terminating the stream. A proxy treats any error in the TCP connection, which includes receiving a TCP segment with the RST bit set, as a stream error of type H3_CONNECT_ERROR (Section 8.1). Correspondingly, if a proxy detects an error with the stream or the QUIC connection, it MUST close the TCP connection. If the underlying TCP implementation permits it, the proxy SHOULD send a TCP segment with the RST bit set.
-HTTP/3 does not support the HTTP Upgrade mechanism ([RFC7230], Section 6.7) or 101 (Switching Protocols) informational status code ([RFC7231], Section 6.2.2).
-Server push is an interaction mode introduced in HTTP/2 [HTTP2] which permits a server to push a request-response exchange to a client in anticipation of the client making the indicated request. This trades off network usage against a potential latency gain. HTTP/3 server push is similar to what is described in HTTP/2 [HTTP2], but uses different mechanisms.
-Each server push is identified by a unique Push ID. This Push ID is used in a single PUSH_PROMISE frame (see Section 7.2.5) which carries the request headers, possibly included in one or more DUPLICATE_PUSH frames (see Section 7.2.8), then included with the push stream which ultimately fulfills those promises.
-Server push is only enabled on a connection when a client sends a MAX_PUSH_ID frame (see Section 7.2.7). A server cannot use server push until it receives a MAX_PUSH_ID frame. A client sends additional MAX_PUSH_ID frames to control the number of pushes that a server can promise. A server SHOULD use Push IDs sequentially, starting at 0. A client MUST treat receipt of a push stream with a Push ID that is greater than the maximum Push ID as a connection error of type H3_ID_ERROR.
-The header of the request message is carried by a PUSH_PROMISE frame (see Section 7.2.5) on the request stream which generated the push. This allows the server push to be associated with a client request. Promised requests MUST conform to the requirements in Section 8.2 of [HTTP2].
-The same server push can be associated with additional client requests using a DUPLICATE_PUSH frame (see Section 7.2.8).
-Ordering of a PUSH_PROMISE or DUPLICATE_PUSH in relation to certain parts of the response is important. The server SHOULD send PUSH_PROMISE or DUPLICATE_PUSH frames prior to sending HEADERS or DATA frames that reference the promised responses. This reduces the chance that a client requests a resource that will be pushed by the server.
-When a server later fulfills a promise, the server push response is conveyed on a push stream (see Section 6.2.2). The push stream identifies the Push ID of the promise that it fulfills, then contains a response to the promised request using the same format described for responses in Section 4.1.
-Due to reordering, DUPLICATE_PUSH frames or push stream data can arrive before the corresponding PUSH_PROMISE frame. When a client receives a DUPLICATE_PUSH frame for an as-yet-unknown Push ID, the request headers of the push are not immediately available. The client can either delay generating new requests for content referenced following the DUPLICATE_PUSH frame until the request headers become available, or can initiate requests for discovered resources and cancel the requests if the requested resource is already being pushed. When a client receives a new push stream with an as-yet-unknown Push ID, both the associated client request and the pushed request headers are unknown. The client can buffer the stream data in expectation of the matching PUSH_PROMISE. The client can use stream flow control (see section 4.1 of [QUIC-TRANSPORT]) to limit the amount of data a server may commit to the pushed stream.
-If a promised server push is not needed by the client, the client SHOULD send a CANCEL_PUSH frame. If the push stream is already open or opens after sending the CANCEL_PUSH frame, the client can abort reading the stream with an error code of H3_REQUEST_CANCELLED. This asks the server not to transfer additional data and indicates that it will be discarded upon receipt.
-Once established, an HTTP/3 connection can be used for many requests and responses over time until the connection is closed. Connection closure can happen in any of several different ways.
-Each QUIC endpoint declares an idle timeout during the handshake. If the connection remains idle (no packets received) for longer than this duration, the peer will assume that the connection has been closed. HTTP/3 implementations will need to open a new connection for new requests if the existing connection has been idle for longer than the server’s advertised idle timeout, and SHOULD do so if approaching the idle timeout.
-HTTP clients are expected to request that the transport keep connections open while there are responses outstanding for requests or server pushes, as described in Section 19.2 of [QUIC-TRANSPORT]. If the client is not expecting a response from the server, allowing an idle connection to time out is preferred over expending effort maintaining a connection that might not be needed. A gateway MAY maintain connections in anticipation of need rather than incur the latency cost of connection establishment to servers. Servers SHOULD NOT actively keep connections open.
-Even when a connection is not idle, either endpoint can decide to stop using the connection and let the connection close gracefully. Since clients drive request generation, clients perform a connection shutdown by not sending additional requests on the connection; responses and pushed responses associated to previous requests will continue to completion. Servers perform the same function by communicating with clients.
-Servers initiate the shutdown of a connection by sending a GOAWAY frame (Section 7.2.6). The GOAWAY frame indicates that client-initiated requests on lower stream IDs were or might be processed in this connection, while requests on the indicated stream ID and greater were rejected. This enables client and server to agree on which requests were accepted prior to the connection shutdown. This identifier MAY be zero if no requests were processed. Servers SHOULD NOT permit additional QUIC streams after sending a GOAWAY frame.
-Clients MUST NOT send new requests on the connection after receiving GOAWAY; a new connection MAY be established to send additional requests.
-Some requests might already be in transit. If the client has already sent requests on streams with a Stream ID greater than or equal to that indicated in the GOAWAY frame, those requests will not be processed and MAY be retried by the client on a different connection. The client MAY cancel these requests. It is RECOMMENDED that the server explicitly reject such requests (see Section 4.1.2) in order to clean up transport state for the affected streams.
-Requests on Stream IDs less than the Stream ID in the GOAWAY frame might have been processed; their status cannot be known until a response is received, the stream is reset individually, or the connection terminates. Servers MAY reject individual requests on streams below the indicated ID if these requests were not processed.
-Servers SHOULD send a GOAWAY frame when the closing of a connection is known in advance, even if the advance notice is small, so that the remote peer can know whether a request has been partially processed or not. For example, if an HTTP client sends a POST at the same time that a server closes a QUIC connection, the client cannot know if the server started to process that POST request if the server does not send a GOAWAY frame to indicate what streams it might have acted on.
-A client that is unable to retry requests loses all requests that are in flight when the server closes the connection. A server MAY send multiple GOAWAY frames indicating different stream IDs, but MUST NOT increase the value they send in the last Stream ID, since clients might already have retried unprocessed requests on another connection. A server that is attempting to gracefully shut down a connection SHOULD send an initial GOAWAY frame with the last Stream ID set to the maximum value allowed by QUIC’s MAX_STREAMS and SHOULD NOT increase the MAX_STREAMS limit thereafter. This signals to the client that a shutdown is imminent and that initiating further requests is prohibited. After allowing time for any in-flight requests (at least one round-trip time), the server MAY send another GOAWAY frame with an updated last Stream ID. This ensures that a connection can be cleanly shut down without losing requests.
-Once all accepted requests have been processed, the server can permit the connection to become idle, or MAY initiate an immediate closure of the connection. An endpoint that completes a graceful shutdown SHOULD use the H3_NO_ERROR code when closing the connection.
-If a client has consumed all available bidirectional stream IDs with requests, the server need not send a GOAWAY frame, since the client is unable to make further requests.
-An HTTP/3 implementation can immediately close the QUIC connection at any time. This results in sending a QUIC CONNECTION_CLOSE frame to the peer; the error code in this frame indicates to the peer why the connection is being closed. See Section 8 for error codes which can be used when closing a connection.
-Before closing the connection, a GOAWAY MAY be sent to allow the client to retry some requests. Including the GOAWAY frame in the same packet as the QUIC CONNECTION_CLOSE frame improves the chances of the frame being received by clients.
-For various reasons, the QUIC transport could indicate to the application layer that the connection has terminated. This might be due to an explicit closure by the peer, a transport-level error, or a change in network topology which interrupts connectivity.
-If a connection terminates without a GOAWAY frame, clients MUST assume that any request which was sent, whether in whole or in part, might have been processed.
-A QUIC stream provides reliable in-order delivery of bytes, but makes no guarantees about order of delivery with regard to bytes on other streams. On the wire, data is framed into QUIC STREAM frames, but this framing is invisible to the HTTP framing layer. The transport layer buffers and orders received QUIC STREAM frames, exposing the data contained within as a reliable byte stream to the application. Although QUIC permits out-of-order delivery within a stream, HTTP/3 does not make use of this feature.
-QUIC streams can be either unidirectional, carrying data only from initiator to receiver, or bidirectional. Streams can be initiated by either the client or the server. For more detail on QUIC streams, see Section 2 of [QUIC-TRANSPORT].
-When HTTP headers and data are sent over QUIC, the QUIC layer handles most of the stream management. HTTP does not need to do any separate multiplexing when using QUIC - data sent over a QUIC stream always maps to a particular HTTP transaction or connection context.
-All client-initiated bidirectional streams are used for HTTP requests and responses. A bidirectional stream ensures that the response can be readily correlated with the request. This means that the client’s first request occurs on QUIC stream 0, with subsequent requests on stream 4, 8, and so on. In order to permit these streams to open, an HTTP/3 server SHOULD configure non-zero minimum values for the number of permitted streams and the initial stream flow control window. So as to not unnecessarily limit parallelism, at least 100 requests SHOULD be permitted at a time.
-HTTP/3 does not use server-initiated bidirectional streams, though an extension could define a use for these streams. Clients MUST treat receipt of a server-initiated bidirectional stream as a connection error of type H3_STREAM_CREATION_ERROR unless such an extension has been negotiated.
-Unidirectional streams, in either direction, are used for a range of purposes. The purpose is indicated by a stream type, which is sent as a variable-length integer at the start of the stream. The format and structure of data that follows this integer is determined by the stream type.
- - +A server MAY send one or more PUSH_PROMISE (see Section 7.2.5) or +DUPLICATE_PUSH (see Section 7.2.8) frames before, after, or +interleaved with the frames of a response message. These PUSH_PROMISE and +DUPLICATE_PUSH frames are not part of the response; see Section 4.4 for +more details.¶
+Frames of unknown types (Section 9), including reserved frames +(Section 7.2.9) MAY be sent on a request or push stream before, after, or +interleaved with other frames described in this section.¶
+The HEADERS and PUSH_PROMISE frames might reference updates to the QPACK dynamic +table. While these updates are not directly part of the message exchange, they +must be received and processed before the message can be consumed. See +Section 4.1.1 for more details.¶
+The "chunked" transfer encoding defined in Section 4.1 of [RFC7230] MUST NOT +be used.¶
+A response MAY consist of multiple messages when and only when one or more +informational responses (1xx; see [RFC7231], Section 6.2) precede a final +response to the same request. Non-final responses do not contain a payload body +or trailers.¶
+If an endpoint receives an invalid sequence of frames on either a request or +a push stream, it MUST respond with a connection error of type +H3_FRAME_UNEXPECTED (Section 8). In particular, a DATA frame before any +HEADERS frame, or a HEADERS or DATA frame after the trailing HEADERS frame is +considered invalid.¶
+An HTTP request/response exchange fully consumes a bidirectional QUIC stream. +After sending a request, a client MUST close the stream for sending. Unless +using the CONNECT method (see Section 4.2), clients MUST NOT make +stream closure dependent on receiving a response to their request. After sending +a final response, the server MUST close the stream for sending. At this point, +the QUIC stream is fully closed.¶
+When a stream is closed, this indicates the end of an HTTP message. Because some +messages are large or unbounded, endpoints SHOULD begin processing partial HTTP +messages once enough of the message has been received to make progress. If a +client stream terminates without enough of the HTTP message to provide a +complete response, the server SHOULD abort its response with the error code +H3_REQUEST_INCOMPLETE.¶
+A server can send a complete response prior to the client sending an entire +request if the response does not depend on any portion of the request that has +not been sent and received. When the server does not need to receive the +remainder of the request, it MAY abort reading the request stream, send a +complete response, and cleanly close the sending part of the stream. The error +code H3_NO_ERROR SHOULD be used when requesting that the client stop sending on +the request stream. Clients MUST NOT discard complete responses as a result of +having their request terminated abruptly, though clients can always discard +responses at their discretion for other reasons. If the server sends a partial +or complete response but does not abort reading, clients SHOULD continue sending +the body of the request and close the stream normally.¶
+HTTP message headers carry information as a series of key-value pairs, called +header fields. For a listing of registered HTTP header fields, see the "Message +Header Field" registry maintained at +https://www.iana.org/assignments/message-headers.¶
+Just as in previous versions of HTTP, header field names are strings of ASCII +characters that are compared in a case-insensitive fashion. Properties of HTTP +header field names and values are discussed in more detail in Section 3.2 of +[RFC7230], though the wire rendering in HTTP/3 differs. As in HTTP/2, header +field names MUST be converted to lowercase prior to their encoding. A request +or response containing uppercase header field names MUST be treated as +malformed (Section 4.1.3).¶
+As in HTTP/2, HTTP/3 uses special pseudo-header fields beginning with the ':' +character (ASCII 0x3a) to convey the target URI, the method of the request, and +the status code for the response. These pseudo-header fields are defined in +Section 8.1.2.3 and 8.1.2.4 of [HTTP2]. Pseudo-header fields are not HTTP +header fields. Endpoints MUST NOT generate pseudo-header fields other than +those defined in [HTTP2]. The restrictions on the use of pseudo-header +fields in Section 8.1.2 of [HTTP2] also apply to HTTP/3. Messages which +are considered malformed under these restrictions are handled as described in +Section 4.1.3.¶
+HTTP/3 uses QPACK header compression as described in [QPACK], a variation of +HPACK which allows the flexibility to avoid header-compression-induced +head-of-line blocking. See that document for additional details.¶
+To allow for better compression efficiency, the cookie header field [RFC6265] +MAY be split into separate header fields, each with one or more cookie-pairs, +before compression. If a decompressed header list contains multiple cookie +header fields, these MUST be concatenated before being passed into a non-HTTP/2, +non-HTTP/3 context, as described in [HTTP2], Section 8.1.2.5.¶
+An HTTP/3 implementation MAY impose a limit on the maximum size of the message +header it will accept on an individual HTTP message. A server that receives a +larger header field list than it is willing to handle can send an HTTP 431 +(Request Header Fields Too Large) status code [RFC6585]. A client can +discard responses that it cannot process. The size of a header field list is +calculated based on the uncompressed size of header fields, including the length +of the name and value in bytes plus an overhead of 32 bytes for each header +field.¶
+If an implementation wishes to advise its peer of this limit, it can be conveyed
+as a number of bytes in the SETTINGS_MAX_HEADER_LIST_SIZE parameter. An
+implementation which has received this parameter SHOULD NOT send an HTTP message
+header which exceeds the indicated size, as the peer will likely refuse to
+process it. However, because this limit is applied at each hop, messages below
+this limit are not guaranteed to be accepted.¶
Clients can cancel requests by resetting and aborting the request stream with an +error code of H3_REQUEST_CANCELLED (Section 8.1). When the client +aborts reading a response, it indicates that this response is no longer of +interest. Implementations SHOULD cancel requests by abruptly terminating any +directions of a stream that are still open.¶
+When the server rejects a request without performing any application processing, +it SHOULD abort its response stream with the error code H3_REQUEST_REJECTED. +In this context, "processed" means that some data from the stream was passed to +some higher layer of software that might have taken some action as a result. The +client can treat requests rejected by the server as though they had never been +sent at all, thereby allowing them to be retried later on a new connection. +Servers MUST NOT use the H3_REQUEST_REJECTED error code for requests which +were partially or fully processed. When a server abandons a response after +partial processing, it SHOULD abort its response stream with the error code +H3_REQUEST_CANCELLED.¶
+When a client resets a request with the error code H3_REQUEST_CANCELLED, a +server MAY abruptly terminate the response using the error code +H3_REQUEST_REJECTED if no processing was performed. Clients MUST NOT use the +H3_REQUEST_REJECTED error code, except when a server has requested closure of +the request stream with this error code.¶
+If a stream is cancelled after receiving a complete response, the client MAY +ignore the cancellation and use the response. However, if a stream is cancelled +after receiving a partial response, the response SHOULD NOT be used. +Automatically retrying such requests is not possible, unless this is otherwise +permitted (e.g., idempotent actions like GET, PUT, or DELETE).¶
+A malformed request or response is one that is an otherwise valid sequence of +frames but is invalid due to the presence of extraneous frames, prohibited +header fields, the absence of mandatory header fields, or the inclusion of +uppercase header field names.¶
+A request or response that includes a payload body can include a
+content-length header field. A request or response is also malformed if the
+value of a content-length header field does not equal the sum of the DATA frame
+payload lengths that form the body. A response that is defined to have no
+payload, as described in Section 3.3.2 of [RFC7230] can have a non-zero
+content-length header field, even though no content is included in DATA frames.¶
Intermediaries that process HTTP requests or responses (i.e., any intermediary +not acting as a tunnel) MUST NOT forward a malformed request or response. +Malformed requests or responses that are detected MUST be treated as a stream +error (Section 8) of type H3_GENERAL_PROTOCOL_ERROR.¶
+For malformed requests, a server MAY send an HTTP response prior to closing or +resetting the stream. Clients MUST NOT accept a malformed response. Note that +these requirements are intended to protect against several types of common +attacks against HTTP; they are deliberately strict because being permissive can +expose implementations to these vulnerabilities.¶
+The pseudo-method CONNECT ([RFC7231], Section 4.3.6) is primarily used with +HTTP proxies to establish a TLS session with an origin server for the purposes +of interacting with "https" resources. In HTTP/1.x, CONNECT is used to convert +an entire HTTP connection into a tunnel to a remote host. In HTTP/2, the CONNECT +method is used to establish a tunnel over a single HTTP/2 stream to a remote +host for similar purposes.¶
+A CONNECT request in HTTP/3 functions in the same manner as in HTTP/2. The +request MUST be formatted as described in [HTTP2], Section 8.3. A CONNECT +request that does not conform to these restrictions is malformed (see +Section 4.1.3). The request stream MUST NOT be closed at the end of the request.¶
+A proxy that supports CONNECT establishes a TCP connection ([RFC0793]) to the +server identified in the ":authority" pseudo-header field. Once this connection +is successfully established, the proxy sends a HEADERS frame containing a 2xx +series status code to the client, as defined in [RFC7231], Section 4.3.6.¶
+All DATA frames on the stream correspond to data sent or received on the TCP +connection. Any DATA frame sent by the client is transmitted by the proxy to the +TCP server; data received from the TCP server is packaged into DATA frames by +the proxy. Note that the size and number of TCP segments is not guaranteed to +map predictably to the size and number of HTTP DATA or QUIC STREAM frames.¶
+Once the CONNECT method has completed, only DATA frames are permitted +to be sent on the stream. Extension frames MAY be used if specifically +permitted by the definition of the extension. Receipt of any other frame type +MUST be treated as a connection error of type H3_FRAME_UNEXPECTED.¶
+The TCP connection can be closed by either peer. When the client ends the +request stream (that is, the receive stream at the proxy enters the "Data Recvd" +state), the proxy will set the FIN bit on its connection to the TCP server. When +the proxy receives a packet with the FIN bit set, it will terminate the send +stream that it sends to the client. TCP connections which remain half-closed in +a single direction are not invalid, but are often handled poorly by servers, so +clients SHOULD NOT close a stream for sending while they still expect to receive +data from the target of the CONNECT.¶
+A TCP connection error is signaled by abruptly terminating the stream. A proxy +treats any error in the TCP connection, which includes receiving a TCP segment +with the RST bit set, as a stream error of type H3_CONNECT_ERROR +(Section 8.1). Correspondingly, if a proxy detects an error with the +stream or the QUIC connection, it MUST close the TCP connection. If the +underlying TCP implementation permits it, the proxy SHOULD send a TCP segment +with the RST bit set.¶
+HTTP/3 does not support the HTTP Upgrade mechanism ([RFC7230], Section 6.7) or +101 (Switching Protocols) informational status code ([RFC7231], Section 6.2.2).¶
+Server push is an interaction mode introduced in HTTP/2 [HTTP2] which permits +a server to push a request-response exchange to a client in anticipation of the +client making the indicated request. This trades off network usage against a +potential latency gain. HTTP/3 server push is similar to what is described in +HTTP/2 [HTTP2], but uses different mechanisms.¶
+Each server push is identified by a unique Push ID. This Push ID is used in a +single PUSH_PROMISE frame (see Section 7.2.5) which carries the request +headers, possibly included in one or more DUPLICATE_PUSH frames (see +Section 7.2.8), then included with the push stream which ultimately +fulfills those promises.¶
+Server push is only enabled on a connection when a client sends a MAX_PUSH_ID +frame (see Section 7.2.7). A server cannot use server push until it +receives a MAX_PUSH_ID frame. A client sends additional MAX_PUSH_ID frames to +control the number of pushes that a server can promise. A server SHOULD use Push +IDs sequentially, starting at 0. A client MUST treat receipt of a push stream +with a Push ID that is greater than the maximum Push ID as a connection error of +type H3_ID_ERROR.¶
+The header of the request message is carried by a PUSH_PROMISE frame (see +Section 7.2.5) on the request stream which generated the push. This +allows the server push to be associated with a client request. Promised +requests MUST conform to the requirements in Section 8.2 of [HTTP2].¶
+The same server push can be associated with additional client requests using a +DUPLICATE_PUSH frame (see Section 7.2.8).¶
+Ordering of a PUSH_PROMISE or DUPLICATE_PUSH in relation to certain parts of the +response is important. The server SHOULD send PUSH_PROMISE or DUPLICATE_PUSH +frames prior to sending HEADERS or DATA frames that reference the promised +responses. This reduces the chance that a client requests a resource that will +be pushed by the server.¶
+When a server later fulfills a promise, the server push response is conveyed on +a push stream (see Section 6.2.2). The push stream identifies the Push ID of +the promise that it fulfills, then contains a response to the promised request +using the same format described for responses in Section 4.1.¶
+Due to reordering, DUPLICATE_PUSH frames or push stream data can arrive before +the corresponding PUSH_PROMISE frame. When a client receives a DUPLICATE_PUSH +frame for an as-yet-unknown Push ID, the request headers of the push are not +immediately available. The client can either delay generating new requests for +content referenced following the DUPLICATE_PUSH frame until the request headers +become available, or can initiate requests for discovered resources and cancel +the requests if the requested resource is already being pushed. When a client +receives a new push stream with an as-yet-unknown Push ID, both the associated +client request and the pushed request headers are unknown. The client can +buffer the stream data in expectation of the matching PUSH_PROMISE. The client +can use stream flow control (see section 4.1 of [QUIC-TRANSPORT]) to limit the +amount of data a server may commit to the pushed stream.¶
+If a promised server push is not needed by the client, the client SHOULD send a +CANCEL_PUSH frame. If the push stream is already open or opens after sending the +CANCEL_PUSH frame, the client can abort reading the stream with an error code of +H3_REQUEST_CANCELLED. This asks the server not to transfer additional data and +indicates that it will be discarded upon receipt.¶
+Once established, an HTTP/3 connection can be used for many requests and +responses over time until the connection is closed. Connection closure can +happen in any of several different ways.¶
+Each QUIC endpoint declares an idle timeout during the handshake. If the +connection remains idle (no packets received) for longer than this duration, the +peer will assume that the connection has been closed. HTTP/3 implementations +will need to open a new connection for new requests if the existing connection +has been idle for longer than the server's advertised idle timeout, and SHOULD +do so if approaching the idle timeout.¶
+HTTP clients are expected to request that the transport keep connections open +while there are responses outstanding for requests or server pushes, as +described in Section 19.2 of [QUIC-TRANSPORT]. If the client is not expecting +a response from the server, allowing an idle connection to time out is preferred +over expending effort maintaining a connection that might not be needed. A +gateway MAY maintain connections in anticipation of need rather than incur the +latency cost of connection establishment to servers. Servers SHOULD NOT actively +keep connections open.¶
+Even when a connection is not idle, either endpoint can decide to stop using the +connection and let the connection close gracefully. Since clients drive request +generation, clients perform a connection shutdown by not sending additional +requests on the connection; responses and pushed responses associated to +previous requests will continue to completion. Servers perform the same +function by communicating with clients.¶
+Servers initiate the shutdown of a connection by sending a GOAWAY frame +(Section 7.2.6). The GOAWAY frame indicates that client-initiated requests +on lower stream IDs were or might be processed in this connection, while +requests on the indicated stream ID and greater were rejected. This enables +client and server to agree on which requests were accepted prior to the +connection shutdown. This identifier MAY be zero if no requests were processed. +Servers SHOULD NOT permit additional QUIC streams after sending a GOAWAY frame.¶
+Clients MUST NOT send new requests on the connection after receiving GOAWAY; +a new connection MAY be established to send additional requests.¶
+Some requests might already be in transit. If the client has already sent +requests on streams with a Stream ID greater than or equal to that indicated in +the GOAWAY frame, those requests will not be processed and MAY be retried by the +client on a different connection. The client MAY cancel these requests. It is +RECOMMENDED that the server explicitly reject such requests (see +Section 4.1.2) in order to clean up transport state for the affected +streams.¶
+Requests on Stream IDs less than the Stream ID in the GOAWAY frame might have +been processed; their status cannot be known until a response is received, the +stream is reset individually, or the connection terminates. Servers MAY reject +individual requests on streams below the indicated ID if these requests were not +processed.¶
+Servers SHOULD send a GOAWAY frame when the closing of a connection is known +in advance, even if the advance notice is small, so that the remote peer can +know whether a request has been partially processed or not. For example, if an +HTTP client sends a POST at the same time that a server closes a QUIC +connection, the client cannot know if the server started to process that POST +request if the server does not send a GOAWAY frame to indicate what streams it +might have acted on.¶
+A client that is unable to retry requests loses all requests that are in flight +when the server closes the connection. A server MAY send multiple GOAWAY frames +indicating different stream IDs, but MUST NOT increase the value they send in +the last Stream ID, since clients might already have retried unprocessed +requests on another connection. A server that is attempting to gracefully shut +down a connection SHOULD send an initial GOAWAY frame with the last Stream ID +set to the maximum value allowed by QUIC's MAX_STREAMS and SHOULD NOT increase +the MAX_STREAMS limit thereafter. This signals to the client that a shutdown is +imminent and that initiating further requests is prohibited. After allowing +time for any in-flight requests (at least one round-trip time), the server MAY +send another GOAWAY frame with an updated last Stream ID. This ensures that a +connection can be cleanly shut down without losing requests.¶
+Once all accepted requests have been processed, the server can permit the +connection to become idle, or MAY initiate an immediate closure of the +connection. An endpoint that completes a graceful shutdown SHOULD use the +H3_NO_ERROR code when closing the connection.¶
+If a client has consumed all available bidirectional stream IDs with requests, +the server need not send a GOAWAY frame, since the client is unable to make +further requests.¶
+An HTTP/3 implementation can immediately close the QUIC connection at any time. +This results in sending a QUIC CONNECTION_CLOSE frame to the peer; the error +code in this frame indicates to the peer why the connection is being closed. +See Section 8 for error codes which can be used when closing a connection.¶
+Before closing the connection, a GOAWAY MAY be sent to allow the client to retry +some requests. Including the GOAWAY frame in the same packet as the QUIC +CONNECTION_CLOSE frame improves the chances of the frame being received by +clients.¶
+For various reasons, the QUIC transport could indicate to the application layer +that the connection has terminated. This might be due to an explicit closure +by the peer, a transport-level error, or a change in network topology which +interrupts connectivity.¶
+If a connection terminates without a GOAWAY frame, clients MUST assume that any +request which was sent, whether in whole or in part, might have been processed.¶
+A QUIC stream provides reliable in-order delivery of bytes, but makes no +guarantees about order of delivery with regard to bytes on other streams. On the +wire, data is framed into QUIC STREAM frames, but this framing is invisible to +the HTTP framing layer. The transport layer buffers and orders received QUIC +STREAM frames, exposing the data contained within as a reliable byte stream to +the application. Although QUIC permits out-of-order delivery within a stream, +HTTP/3 does not make use of this feature.¶
+QUIC streams can be either unidirectional, carrying data only from initiator to +receiver, or bidirectional. Streams can be initiated by either the client or +the server. For more detail on QUIC streams, see Section 2 of +[QUIC-TRANSPORT].¶
+When HTTP headers and data are sent over QUIC, the QUIC layer handles most of +the stream management. HTTP does not need to do any separate multiplexing when +using QUIC - data sent over a QUIC stream always maps to a particular HTTP +transaction or connection context.¶
+All client-initiated bidirectional streams are used for HTTP requests and +responses. A bidirectional stream ensures that the response can be readily +correlated with the request. This means that the client's first request occurs +on QUIC stream 0, with subsequent requests on stream 4, 8, and so on. In order +to permit these streams to open, an HTTP/3 server SHOULD configure non-zero +minimum values for the number of permitted streams and the initial stream flow +control window. So as to not unnecessarily limit parallelism, at least 100 +requests SHOULD be permitted at a time.¶
+HTTP/3 does not use server-initiated bidirectional streams, though an extension +could define a use for these streams. Clients MUST treat receipt of a +server-initiated bidirectional stream as a connection error of type +H3_STREAM_CREATION_ERROR unless such an extension has been negotiated.¶
+Unidirectional streams, in either direction, are used for a range of purposes. +The purpose is indicated by a stream type, which is sent as a variable-length +integer at the start of the stream. The format and structure of data that +follows this integer is determined by the stream type.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -892,27 +2043,90 @@-| Stream Type (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Unidirectional Stream Header
-Some stream types are reserved (Section 6.2.3). Two stream types are defined in this document: control streams (Section 6.2.1) and push streams (Section 6.2.2). [QPACK] defines two additional stream types. Other stream types can be defined by extensions to HTTP/3; see Section 9 for more details.
-The performance of HTTP/3 connections in the early phase of their lifetime is sensitive to the creation and exchange of data on unidirectional streams. Endpoints that excessively restrict the number of streams or the flow control window of these streams will increase the chance that the remote peer reaches the limit early and becomes blocked. In particular, implementations should consider that remote peers may wish to exercise reserved stream behavior (Section 6.2.3) with some of the unidirectional streams they are permitted to use. To avoid blocking, the transport parameters sent by both clients and servers MUST allow the peer to create at least one unidirectional stream for the HTTP control stream plus the number of unidirectional streams required by mandatory extensions (three being the minimum number required for the base HTTP/3 protocol and QPACK), and SHOULD provide at least 1,024 bytes of flow control credit to each stream.
-Note that an endpoint is not required to grant additional credits to create more unidirectional streams if its peer consumes all the initial credits before creating the critical unidirectional streams. Endpoints SHOULD create the HTTP control stream as well as the unidirectional streams required by mandatory extensions (such as the QPACK encoder and decoder streams) first, and then create additional streams as allowed by their peer.
-If the stream header indicates a stream type which is not supported by the recipient, the remainder of the stream cannot be consumed as the semantics are unknown. Recipients of unknown stream types MAY abort reading of the stream with an error code of H3_STREAM_CREATION_ERROR, but MUST NOT consider such streams to be a connection error of any kind.
-Implementations MAY send stream types before knowing whether the peer supports them. However, stream types which could modify the state or semantics of existing protocol components, including QPACK or other extensions, MUST NOT be sent until the peer is known to support them.
-A sender can close or reset a unidirectional stream unless otherwise specified. A receiver MUST tolerate unidirectional streams being closed or reset prior to the reception of the unidirectional stream header.
-A control stream is indicated by a stream type of 0x00. Data on this stream consists of HTTP/3 frames, as defined in Section 7.2.
-Each side MUST initiate a single control stream at the beginning of the connection and send its SETTINGS frame as the first frame on this stream. If the first frame of the control stream is any other frame type, this MUST be treated as a connection error of type H3_MISSING_SETTINGS. Only one control stream per peer is permitted; receipt of a second stream which claims to be a control stream MUST be treated as a connection error of type H3_STREAM_CREATION_ERROR. The sender MUST NOT close the control stream, and the receiver MUST NOT request that the sender close the control stream. If either control stream is closed at any point, this MUST be treated as a connection error of type H3_CLOSED_CRITICAL_STREAM.
-A pair of unidirectional streams is used rather than a single bidirectional stream. This allows either peer to send data as soon as it is able. Depending on whether 0-RTT is enabled on the connection, either client or server might be able to send stream data first after the cryptographic handshake completes.
-Server push is an optional feature introduced in HTTP/2 that allows a server to initiate a response before a request has been made. See Section 4.4 for more details.
-A push stream is indicated by a stream type of 0x01, followed by the Push ID of the promise that it fulfills, encoded as a variable-length integer. The remaining data on this stream consists of HTTP/3 frames, as defined in Section 7.2, and fulfills a promised server push by zero or more non-final HTTP responses followed by a single final HTTP response, as defined in Section 4.1. Server push and Push IDs are described in Section 4.4.
-Only servers can push; if a server receives a client-initiated push stream, this MUST be treated as a connection error of type H3_STREAM_CREATION_ERROR.
- - +Some stream types are reserved (Section 6.2.3). Two stream types are +defined in this document: control streams (Section 6.2.1) and push streams +(Section 6.2.2). [QPACK] defines two additional stream types. Other stream +types can be defined by extensions to HTTP/3; see Section 9 for more +details.¶
+The performance of HTTP/3 connections in the early phase of their lifetime is +sensitive to the creation and exchange of data on unidirectional streams. +Endpoints that excessively restrict the number of streams or the flow control +window of these streams will increase the chance that the remote peer reaches +the limit early and becomes blocked. In particular, implementations should +consider that remote peers may wish to exercise reserved stream behavior +(Section 6.2.3) with some of the unidirectional streams they are permitted +to use. To avoid blocking, the transport parameters sent by both clients and +servers MUST allow the peer to create at least one unidirectional stream for the +HTTP control stream plus the number of unidirectional streams required by +mandatory extensions (three being the minimum number required for the base +HTTP/3 protocol and QPACK), and SHOULD provide at least 1,024 bytes of flow +control credit to each stream.¶
+Note that an endpoint is not required to grant additional credits to create more +unidirectional streams if its peer consumes all the initial credits before +creating the critical unidirectional streams. Endpoints SHOULD create the HTTP +control stream as well as the unidirectional streams required by mandatory +extensions (such as the QPACK encoder and decoder streams) first, and then +create additional streams as allowed by their peer.¶
+If the stream header indicates a stream type which is not supported by the +recipient, the remainder of the stream cannot be consumed as the semantics are +unknown. Recipients of unknown stream types MAY abort reading of the stream with +an error code of H3_STREAM_CREATION_ERROR, but MUST NOT consider such streams +to be a connection error of any kind.¶
+Implementations MAY send stream types before knowing whether the peer supports +them. However, stream types which could modify the state or semantics of +existing protocol components, including QPACK or other extensions, MUST NOT be +sent until the peer is known to support them.¶
+A sender can close or reset a unidirectional stream unless otherwise specified. +A receiver MUST tolerate unidirectional streams being closed or reset prior to +the reception of the unidirectional stream header.¶
+A control stream is indicated by a stream type of 0x00. Data on this stream
+consists of HTTP/3 frames, as defined in Section 7.2.¶
Each side MUST initiate a single control stream at the beginning of the +connection and send its SETTINGS frame as the first frame on this stream. If +the first frame of the control stream is any other frame type, this MUST be +treated as a connection error of type H3_MISSING_SETTINGS. Only one control +stream per peer is permitted; receipt of a second stream which claims to be a +control stream MUST be treated as a connection error of type +H3_STREAM_CREATION_ERROR. The sender MUST NOT close the control stream, and +the receiver MUST NOT request that the sender close the control stream. If +either control stream is closed at any point, this MUST be treated as a +connection error of type H3_CLOSED_CRITICAL_STREAM.¶
+A pair of unidirectional streams is used rather than a single bidirectional +stream. This allows either peer to send data as soon as it is able. Depending +on whether 0-RTT is enabled on the connection, either client or server might be +able to send stream data first after the cryptographic handshake completes.¶
+Server push is an optional feature introduced in HTTP/2 that allows a server to +initiate a response before a request has been made. See Section 4.4 for +more details.¶
+A push stream is indicated by a stream type of 0x01, followed by the Push ID
+of the promise that it fulfills, encoded as a variable-length integer. The
+remaining data on this stream consists of HTTP/3 frames, as defined in
+Section 7.2, and fulfills a promised server push by zero or more non-final HTTP
+responses followed by a single final HTTP response, as defined in
+Section 4.1. Server push and Push IDs are described in
+Section 4.4.¶
Only servers can push; if a server receives a client-initiated push stream, this +MUST be treated as a connection error of type H3_STREAM_CREATION_ERROR.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -922,102 +2136,158 @@-| Push ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Push Stream Header
-Each Push ID MUST only be used once in a push stream header. If a push stream header includes a Push ID that was used in another push stream header, the client MUST treat this as a connection error of type H3_ID_ERROR.
-Stream types of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown types be ignored. These streams have no semantics, and can be sent when application-layer padding is desired. They MAY also be sent on connections where no data is currently being transferred. Endpoints MUST NOT consider these streams to have any meaning upon receipt.
-The payload and length of the stream are selected in any manner the implementation chooses.
-HTTP frames are carried on QUIC streams, as described in Section 6. HTTP/3 defines three stream types: control stream, request stream, and push stream. This section describes HTTP/3 frame formats and the streams types on which they are permitted; see Table 1 for an overview. A comparison between HTTP/2 and HTTP/3 frames is provided in Appendix A.2.
- - -| Frame | -Control Stream | -Request Stream | -Push Stream | -Section | -
|---|---|---|---|---|
| DATA | -No | -Yes | -Yes | -Section 7.2.1 | -
| HEADERS | -No | -Yes | -Yes | -Section 7.2.2 | -
| CANCEL_PUSH | -Yes | -No | -No | -Section 7.2.3 | -
| SETTINGS | -Yes (1) | -No | -No | -Section 7.2.4 | -
| PUSH_PROMISE | -No | -Yes | -No | -Section 7.2.5 | -
| GOAWAY | -Yes | -No | -No | -Section 7.2.6 | -
| MAX_PUSH_ID | -Yes | -No | -No | -Section 7.2.7 | -
| DUPLICATE_PUSH | -No | -Yes | -No | -Section 7.2.8 | -
| Reserved | -Yes | -Yes | -Yes | -Section 7.2.9 | -
Certain frames can only occur as the first frame of a particular stream type; these are indicated in Table 1 with a (1). Specific guidance is provided in the relevant section.
-Note that, unlike QUIC frames, HTTP/3 frames can span multiple packets.
-All frames have the following format:
- - +Each Push ID MUST only be used once in a push stream header. If a push stream +header includes a Push ID that was used in another push stream header, the +client MUST treat this as a connection error of type H3_ID_ERROR.¶
+Stream types of the format 0x1f * N + 0x21 for integer values of N are
+reserved to exercise the requirement that unknown types be ignored. These
+streams have no semantics, and can be sent when application-layer padding is
+desired. They MAY also be sent on connections where no data is currently being
+transferred. Endpoints MUST NOT consider these streams to have any meaning upon
+receipt.¶
The payload and length of the stream are selected in any manner the +implementation chooses.¶
+HTTP frames are carried on QUIC streams, as described in Section 6. +HTTP/3 defines three stream types: control stream, request stream, and push +stream. This section describes HTTP/3 frame formats and the streams types on +which they are permitted; see Table 1 for an overview. A +comparison between HTTP/2 and HTTP/3 frames is provided in Appendix A.2.¶
+| Frame | +Control Stream | +Request Stream | +Push Stream | +Section | +
|---|---|---|---|---|
| DATA | +No | +Yes | +Yes | ++ Section 7.2.1 + | +
| HEADERS | +No | +Yes | +Yes | ++ Section 7.2.2 + | +
| CANCEL_PUSH | +Yes | +No | +No | ++ Section 7.2.3 + | +
| SETTINGS | +Yes (1) | +No | +No | ++ Section 7.2.4 + | +
| PUSH_PROMISE | +No | +Yes | +No | ++ Section 7.2.5 + | +
| GOAWAY | +Yes | +No | +No | ++ Section 7.2.6 + | +
| MAX_PUSH_ID | +Yes | +No | +No | ++ Section 7.2.7 + | +
| DUPLICATE_PUSH | +No | +Yes | +No | ++ Section 7.2.8 + | +
| Reserved | +Yes | +Yes | +Yes | ++ Section 7.2.9 + | +
Certain frames can only occur as the first frame of a particular stream type; +these are indicated in Table 1 with a (1). Specific guidance +is provided in the relevant section.¶
+Note that, unlike QUIC frames, HTTP/3 frames can span multiple packets.¶
+All frames have the following format:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1029,30 +2299,56 @@-| Frame Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: HTTP/3 Frame Format
-A frame includes the following fields:
- - -A frame includes the following fields:¶
+Each frame’s payload MUST contain exactly the fields identified in its description. A frame payload that contains additional bytes after the identified fields or a frame payload that terminates before the end of the identified fields MUST be treated as a connection error of type H3_FRAME_ERROR.
-When a stream terminates cleanly, if the last frame on the stream was truncated, this MUST be treated as a connection error (Section 8) of type H3_FRAME_ERROR. Streams which terminate abruptly may be reset at any point in a frame.
-DATA frames (type=0x0) convey arbitrary, variable-length sequences of bytes associated with an HTTP request or response payload.
-DATA frames MUST be associated with an HTTP request or response. If a DATA frame is received on a control stream, the recipient MUST respond with a connection error (Section 8) of type H3_FRAME_UNEXPECTED.
- - +Each frame's payload MUST contain exactly the fields identified in its +description. A frame payload that contains additional bytes after the +identified fields or a frame payload that terminates before the end of the +identified fields MUST be treated as a connection error of type +H3_FRAME_ERROR.¶
+When a stream terminates cleanly, if the last frame on the stream was truncated, +this MUST be treated as a connection error (Section 8) of type +H3_FRAME_ERROR. Streams which terminate abruptly may be reset at any point in +a frame.¶
+DATA frames (type=0x0) convey arbitrary, variable-length sequences of bytes +associated with an HTTP request or response payload.¶
+DATA frames MUST be associated with an HTTP request or response. If a DATA +frame is received on a control stream, the recipient MUST respond with a +connection error (Section 8) of type H3_FRAME_UNEXPECTED.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1060,13 +2356,23 @@-| Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: DATA Frame Payload
-The HEADERS frame (type=0x1) is used to carry a header block, compressed using QPACK. See [QPACK] for more details.
- - +The HEADERS frame (type=0x1) is used to carry a header block, compressed using +QPACK. See [QPACK] for more details.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1074,18 +2380,45 @@-| Header Block (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: HEADERS Frame Payload
-HEADERS frames can only be sent on request / push streams. If a HEADERS frame is received on a control stream, the recipient MUST respond with a connection error (Section 8) of type H3_FRAME_UNEXPECTED.
-The CANCEL_PUSH frame (type=0x3) is used to request cancellation of a server push prior to the push stream being received. The CANCEL_PUSH frame identifies a server push by Push ID (see Section 7.2.5), encoded as a variable-length integer.
-When a client sends CANCEL_PUSH, it is indicating that it does not wish to receive the promised resource. The server SHOULD abort sending the resource, but the mechanism to do so depends on the state of the corresponding push stream. If the server has not yet created a push stream, it does not create one. If the push stream is open, the server SHOULD abruptly terminate that stream. If the push stream has already ended, the server MAY still abruptly terminate the stream or MAY take no action.
-When a server sends CANCEL_PUSH, it is indicating that it will not be fulfilling a promise and has not created a push stream. The client should not expect the corresponding promise to be fulfilled.
-Sending CANCEL_PUSH has no direct effect on the state of existing push streams. A server SHOULD NOT send a CANCEL_PUSH when it has already created a corresponding push stream, and a client SHOULD NOT send a CANCEL_PUSH when it has already received a corresponding push stream.
-A CANCEL_PUSH frame is sent on the control stream. Receiving a CANCEL_PUSH frame on a stream other than the control stream MUST be treated as a connection error of type H3_FRAME_UNEXPECTED.
- - +HEADERS frames can only be sent on request / push streams. If a HEADERS frame +is received on a control stream, the recipient MUST respond with a connection +error (Section 8) of type H3_FRAME_UNEXPECTED.¶
+The CANCEL_PUSH frame (type=0x3) is used to request cancellation of a server +push prior to the push stream being received. The CANCEL_PUSH frame identifies +a server push by Push ID (see Section 7.2.5), encoded as a +variable-length integer.¶
+When a client sends CANCEL_PUSH, it is indicating that it does not wish to +receive the promised resource. The server SHOULD abort sending the resource, +but the mechanism to do so depends on the state of the corresponding push +stream. If the server has not yet created a push stream, it does not create +one. If the push stream is open, the server SHOULD abruptly terminate that +stream. If the push stream has already ended, the server MAY still abruptly +terminate the stream or MAY take no action.¶
+When a server sends CANCEL_PUSH, it is indicating that it will not be fulfilling +a promise and has not created a push stream. The client should not expect the +corresponding promise to be fulfilled.¶
+Sending CANCEL_PUSH has no direct effect on the state of existing push streams. +A server SHOULD NOT send a CANCEL_PUSH when it has already created a +corresponding push stream, and a client SHOULD NOT send a CANCEL_PUSH when it +has already received a corresponding push stream.¶
+A CANCEL_PUSH frame is sent on the control stream. Receiving a CANCEL_PUSH +frame on a stream other than the control stream MUST be treated as a connection +error of type H3_FRAME_UNEXPECTED.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1093,21 +2426,59 @@-| Push ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: CANCEL_PUSH Frame Payload
-The CANCEL_PUSH frame carries a Push ID encoded as a variable-length integer. The Push ID identifies the server push that is being cancelled (see Section 7.2.5). If a CANCEL_PUSH frame is received which references a Push ID greater than currently allowed on the connection, this MUST be treated as a connection error of type H3_ID_ERROR.
-If the client receives a CANCEL_PUSH frame, that frame might identify a Push ID that has not yet been mentioned by a PUSH_PROMISE frame due to reordering. If a server receives a CANCEL_PUSH frame for a Push ID that has not yet been mentioned by a PUSH_PROMISE frame, this MUST be treated as a connection error of type H3_ID_ERROR.
-The SETTINGS frame (type=0x4) conveys configuration parameters that affect how endpoints communicate, such as preferences and constraints on peer behavior. Individually, a SETTINGS parameter can also be referred to as a “setting”; the identifier and value of each setting parameter can be referred to as a “setting identifier” and a “setting value”.
-SETTINGS frames always apply to a connection, never a single stream. A SETTINGS frame MUST be sent as the first frame of each control stream (see Section 6.2.1) by each peer, and MUST NOT be sent subsequently. If an endpoint receives a second SETTINGS frame on the control stream, the endpoint MUST respond with a connection error of type H3_FRAME_UNEXPECTED.
-SETTINGS frames MUST NOT be sent on any stream other than the control stream. If an endpoint receives a SETTINGS frame on a different stream, the endpoint MUST respond with a connection error of type H3_FRAME_UNEXPECTED.
-SETTINGS parameters are not negotiated; they describe characteristics of the sending peer, which can be used by the receiving peer. However, a negotiation can be implied by the use of SETTINGS - each peer uses SETTINGS to advertise a set of supported values. The definition of the setting would describe how each peer combines the two sets to conclude which choice will be used. SETTINGS does not provide a mechanism to identify when the choice takes effect.
-Different values for the same parameter can be advertised by each peer. For example, a client might be willing to consume a very large response header, while servers are more cautious about request size.
-The same setting identifier MUST NOT occur more than once in the SETTINGS frame. A receiver MAY treat the presence of duplicate setting identifiers as a connection error of type H3_SETTINGS_ERROR.
-The payload of a SETTINGS frame consists of zero or more parameters. Each parameter consists of a setting identifier and a value, both encoded as QUIC variable-length integers.
- - +The CANCEL_PUSH frame carries a Push ID encoded as a variable-length integer. +The Push ID identifies the server push that is being cancelled (see +Section 7.2.5). If a CANCEL_PUSH frame is received which references a +Push ID greater than currently allowed on the connection, this MUST be treated +as a connection error of type H3_ID_ERROR.¶
+If the client receives a CANCEL_PUSH frame, that frame might identify a Push ID +that has not yet been mentioned by a PUSH_PROMISE frame due to reordering. If a +server receives a CANCEL_PUSH frame for a Push ID that has not yet been +mentioned by a PUSH_PROMISE frame, this MUST be treated as a connection error of +type H3_ID_ERROR.¶
+The SETTINGS frame (type=0x4) conveys configuration parameters that affect how +endpoints communicate, such as preferences and constraints on peer behavior. +Individually, a SETTINGS parameter can also be referred to as a "setting"; the +identifier and value of each setting parameter can be referred to as a "setting +identifier" and a "setting value".¶
+SETTINGS frames always apply to a connection, never a single stream. A SETTINGS +frame MUST be sent as the first frame of each control stream (see +Section 6.2.1) by each peer, and MUST NOT be sent subsequently. If +an endpoint receives a second SETTINGS frame on the control stream, the endpoint +MUST respond with a connection error of type H3_FRAME_UNEXPECTED.¶
+SETTINGS frames MUST NOT be sent on any stream other than the control stream. +If an endpoint receives a SETTINGS frame on a different stream, the endpoint +MUST respond with a connection error of type H3_FRAME_UNEXPECTED.¶
+SETTINGS parameters are not negotiated; they describe characteristics of the +sending peer, which can be used by the receiving peer. However, a negotiation +can be implied by the use of SETTINGS - each peer uses SETTINGS to advertise a +set of supported values. The definition of the setting would describe how each +peer combines the two sets to conclude which choice will be used. SETTINGS does +not provide a mechanism to identify when the choice takes effect.¶
+Different values for the same parameter can be advertised by each peer. For +example, a client might be willing to consume a very large response header, +while servers are more cautious about request size.¶
+The same setting identifier MUST NOT occur more than once in the SETTINGS frame. +A receiver MAY treat the presence of duplicate setting identifiers as a +connection error of type H3_SETTINGS_ERROR.¶
+The payload of a SETTINGS frame consists of zero or more parameters. Each +parameter consists of a setting identifier and a value, both encoded as QUIC +variable-length integers.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1117,38 +2488,98 @@-| Value (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: SETTINGS Parameter Format
-An implementation MUST ignore the contents for any SETTINGS identifier it does not understand.
-The following settings are defined in HTTP/3:
- - -An implementation MUST ignore the contents for any SETTINGS identifier it does +not understand.¶
+The following settings are defined in HTTP/3:¶
+Setting identifiers of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown identifiers be ignored. Such settings have no defined meaning. Endpoints SHOULD include at least one such setting in their SETTINGS frame. Endpoints MUST NOT consider such settings to have any meaning upon receipt.
-Because the setting has no defined meaning, the value of the setting can be any value the implementation selects.
-Additional settings can be defined by extensions to HTTP/3; see Section 9 for more details.
-An HTTP implementation MUST NOT send frames or requests which would be invalid based on its current understanding of the peer’s settings.
-All settings begin at an initial value. Each endpoint SHOULD use these initial values to send messages before the peer’s SETTINGS frame has arrived, as packets carrying the settings can be lost or delayed. When the SETTINGS frame arrives, any settings are changed to their new values.
-This removes the need to wait for the SETTINGS frame before sending messages. Endpoints MUST NOT require any data to be received from the peer prior to sending the SETTINGS frame; settings MUST be sent as soon as the transport is ready to send data.
-For servers, the initial value of each client setting is the default value.
-For clients using a 1-RTT QUIC connection, the initial value of each server setting is the default value. 1-RTT keys will always become available prior to SETTINGS arriving, even if the server sends SETTINGS immediately. Clients SHOULD NOT wait indefinitely for SETTINGS to arrive before sending requests, but SHOULD process received datagrams in order to increase the likelihood of processing SETTINGS before sending the first request.
-When a 0-RTT QUIC connection is being used, the initial value of each server setting is the value used in the previous session. Clients SHOULD store the settings the server provided in the connection where resumption information was provided, but MAY opt not to store settings in certain cases (e.g., if the session ticket is received before the SETTINGS frame). A client MUST comply with stored settings – or default values, if no values are stored – when attempting 0-RTT. Once a server has provided new settings, clients MUST comply with those values.
-A server can remember the settings that it advertised, or store an integrity-protected copy of the values in the ticket and recover the information when accepting 0-RTT data. A server uses the HTTP/3 settings values in determining whether to accept 0-RTT data. If the server cannot determine that the settings remembered by a client are compatible with its current settings, it MUST NOT accept 0-RTT data. Remembered settings are compatible if a client complying with those settings would not violate the server’s current settings.
-A server MAY accept 0-RTT and subsequently provide different settings in its SETTINGS frame. If 0-RTT data is accepted by the server, its SETTINGS frame MUST NOT reduce any limits or alter any values that might be violated by the client with its 0-RTT data. The server MUST include all settings which differ from their default values. If a server accepts 0-RTT but then sends settings that are not compatible with the previously specified settings, this MUST be treated as a connection error of type H3_SETTINGS_ERROR. If a server accepts 0-RTT but then sends a SETTINGS frame that omits a setting value that the client understands (apart from reserved setting identifiers) that was previously specified to have a non-default value, this MUST be treated as a connection error of type H3_SETTINGS_ERROR.
-The PUSH_PROMISE frame (type=0x5) is used to carry a promised request header set from server to client on a request stream, as in HTTP/2.
- - +Setting identifiers of the format 0x1f * N + 0x21 for integer values of N are
+reserved to exercise the requirement that unknown identifiers be ignored. Such
+settings have no defined meaning. Endpoints SHOULD include at least one such
+setting in their SETTINGS frame. Endpoints MUST NOT consider such settings to
+have any meaning upon receipt.¶
Because the setting has no defined meaning, the value of the setting can be any +value the implementation selects.¶
+Additional settings can be defined by extensions to HTTP/3; see Section 9 +for more details.¶
+An HTTP implementation MUST NOT send frames or requests which would be invalid +based on its current understanding of the peer's settings.¶
+All settings begin at an initial value. Each endpoint SHOULD use these initial +values to send messages before the peer's SETTINGS frame has arrived, as packets +carrying the settings can be lost or delayed. When the SETTINGS frame arrives, +any settings are changed to their new values.¶
+This removes the need to wait for the SETTINGS frame before sending messages. +Endpoints MUST NOT require any data to be received from the peer prior to +sending the SETTINGS frame; settings MUST be sent as soon as the transport is +ready to send data.¶
+For servers, the initial value of each client setting is the default value.¶
+For clients using a 1-RTT QUIC connection, the initial value of each server +setting is the default value. 1-RTT keys will always become available prior to +SETTINGS arriving, even if the server sends SETTINGS immediately. Clients SHOULD +NOT wait indefinitely for SETTINGS to arrive before sending requests, but SHOULD +process received datagrams in order to increase the likelihood of processing +SETTINGS before sending the first request.¶
+When a 0-RTT QUIC connection is being used, the initial value of each server +setting is the value used in the previous session. Clients SHOULD store the +settings the server provided in the connection where resumption information was +provided, but MAY opt not to store settings in certain cases (e.g., if the +session ticket is received before the SETTINGS frame). A client MUST comply with +stored settings - or default values, if no values are stored - when attempting +0-RTT. Once a server has provided new settings, clients MUST comply with those +values.¶
+A server can remember the settings that it advertised, or store an +integrity-protected copy of the values in the ticket and recover the information +when accepting 0-RTT data. A server uses the HTTP/3 settings values in +determining whether to accept 0-RTT data. If the server cannot determine that +the settings remembered by a client are compatible with its current settings, it +MUST NOT accept 0-RTT data. Remembered settings are compatible if a client +complying with those settings would not violate the server's current settings.¶
+A server MAY accept 0-RTT and subsequently provide different settings in its +SETTINGS frame. If 0-RTT data is accepted by the server, its SETTINGS frame MUST +NOT reduce any limits or alter any values that might be violated by the client +with its 0-RTT data. The server MUST include all settings which differ from +their default values. If a server accepts 0-RTT but then sends settings that +are not compatible with the previously specified settings, this MUST be treated +as a connection error of type H3_SETTINGS_ERROR. If a server accepts 0-RTT but +then sends a SETTINGS frame that omits a setting value that the client +understands (apart from reserved setting identifiers) that was previously +specified to have a non-default value, this MUST be treated as a connection +error of type H3_SETTINGS_ERROR.¶
+The PUSH_PROMISE frame (type=0x5) is used to carry a promised request header +set from server to client on a request stream, as in HTTP/2.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1158,27 +2589,52 @@-| Header Block (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: PUSH_PROMISE Frame Payload
-The payload consists of:
- - -The payload consists of:¶
+A server MUST NOT use a Push ID that is larger than the client has provided in a MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat receipt of a PUSH_PROMISE frame that contains a larger Push ID than the client has advertised as a connection error of H3_ID_ERROR.
-A server MUST NOT use the same Push ID in multiple PUSH_PROMISE frames. A client MUST treat receipt of a Push ID which has already been promised as a connection error of type H3_ID_ERROR.
-If a PUSH_PROMISE frame is received on the control stream, the client MUST respond with a connection error (Section 8) of type H3_FRAME_UNEXPECTED.
-A client MUST NOT send a PUSH_PROMISE frame. A server MUST treat the receipt of a PUSH_PROMISE frame as a connection error of type H3_FRAME_UNEXPECTED.
-See Section 4.4 for a description of the overall server push mechanism.
-The GOAWAY frame (type=0x7) is used to initiate graceful shutdown of a connection by a server. GOAWAY allows a server to stop accepting new requests while still finishing processing of previously received requests. This enables administrative actions, like server maintenance. GOAWAY by itself does not close a connection.
- - +A server MUST NOT use a Push ID that is larger than the client has provided in a +MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat receipt of a +PUSH_PROMISE frame that contains a larger Push ID than the client has advertised +as a connection error of H3_ID_ERROR.¶
+A server MUST NOT use the same Push ID in multiple PUSH_PROMISE frames. A client +MUST treat receipt of a Push ID which has already been promised as a connection +error of type H3_ID_ERROR.¶
+If a PUSH_PROMISE frame is received on the control stream, the client MUST +respond with a connection error (Section 8) of type H3_FRAME_UNEXPECTED.¶
+A client MUST NOT send a PUSH_PROMISE frame. A server MUST treat the receipt +of a PUSH_PROMISE frame as a connection error of type H3_FRAME_UNEXPECTED.¶
+See Section 4.4 for a description of the overall server push mechanism.¶
+The GOAWAY frame (type=0x7) is used to initiate graceful shutdown of a +connection by a server. GOAWAY allows a server to stop accepting new requests +while still finishing processing of previously received requests. This enables +administrative actions, like server maintenance. GOAWAY by itself does not +close a connection.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1186,20 +2642,46 @@-| Stream ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: GOAWAY Frame Payload
-The GOAWAY frame is always sent on the control stream. It carries a QUIC Stream ID for a client-initiated bidirectional stream encoded as a variable-length integer. A client MUST treat receipt of a GOAWAY frame containing a Stream ID of any other type as a connection error of type H3_ID_ERROR.
-Clients do not need to send GOAWAY to initiate a graceful shutdown; they simply stop making new requests. A server MUST treat receipt of a GOAWAY frame on any stream as a connection error (Section 8) of type H3_FRAME_UNEXPECTED.
-The GOAWAY frame applies to the connection, not a specific stream. A client MUST treat a GOAWAY frame on a stream other than the control stream as a connection error (Section 8) of type H3_FRAME_UNEXPECTED.
-See Section 5.2 for more information on the use of the GOAWAY frame.
-The MAX_PUSH_ID frame (type=0xD) is used by clients to control the number of server pushes that the server can initiate. This sets the maximum value for a Push ID that the server can use in PUSH_PROMISE and CANCEL_PUSH frames. Consequently, this also limits the number of push streams that the server can initiate in addition to the limit maintained by the QUIC transport.
-The MAX_PUSH_ID frame is always sent on the control stream. Receipt of a MAX_PUSH_ID frame on any other stream MUST be treated as a connection error of type H3_FRAME_UNEXPECTED.
-A server MUST NOT send a MAX_PUSH_ID frame. A client MUST treat the receipt of a MAX_PUSH_ID frame as a connection error of type H3_FRAME_UNEXPECTED.
-The maximum Push ID is unset when a connection is created, meaning that a server cannot push until it receives a MAX_PUSH_ID frame. A client that wishes to manage the number of promised server pushes can increase the maximum Push ID by sending MAX_PUSH_ID frames as the server fulfills or cancels server pushes.
- - +The GOAWAY frame is always sent on the control stream. It carries a QUIC Stream +ID for a client-initiated bidirectional stream encoded as a variable-length +integer. A client MUST treat receipt of a GOAWAY frame containing a Stream ID +of any other type as a connection error of type H3_ID_ERROR.¶
+Clients do not need to send GOAWAY to initiate a graceful shutdown; they simply +stop making new requests. A server MUST treat receipt of a GOAWAY frame on any +stream as a connection error (Section 8) of type H3_FRAME_UNEXPECTED.¶
+The GOAWAY frame applies to the connection, not a specific stream. A client +MUST treat a GOAWAY frame on a stream other than the control stream as a +connection error (Section 8) of type H3_FRAME_UNEXPECTED.¶
+See Section 5.2 for more information on the use of the GOAWAY frame.¶
+The MAX_PUSH_ID frame (type=0xD) is used by clients to control the number of +server pushes that the server can initiate. This sets the maximum value for a +Push ID that the server can use in PUSH_PROMISE and CANCEL_PUSH frames. +Consequently, this also limits the number of push streams that the server can +initiate in addition to the limit maintained by the QUIC transport.¶
+The MAX_PUSH_ID frame is always sent on the control stream. Receipt of a +MAX_PUSH_ID frame on any other stream MUST be treated as a connection error of +type H3_FRAME_UNEXPECTED.¶
+A server MUST NOT send a MAX_PUSH_ID frame. A client MUST treat the receipt of +a MAX_PUSH_ID frame as a connection error of type H3_FRAME_UNEXPECTED.¶
+The maximum Push ID is unset when a connection is created, meaning that a server +cannot push until it receives a MAX_PUSH_ID frame. A client that wishes to +manage the number of promised server pushes can increase the maximum Push ID by +sending MAX_PUSH_ID frames as the server fulfills or cancels server pushes.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1207,16 +2689,33 @@-| Push ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 10: MAX_PUSH_ID Frame Payload
-The MAX_PUSH_ID frame carries a single variable-length integer that identifies the maximum value for a Push ID that the server can use (see Section 7.2.5). A MAX_PUSH_ID frame cannot reduce the maximum Push ID; receipt of a MAX_PUSH_ID that contains a smaller value than previously received MUST be treated as a connection error of type H3_ID_ERROR.
-The DUPLICATE_PUSH frame (type=0xE) is used by servers to indicate that an existing pushed resource is related to multiple client requests.
-The DUPLICATE_PUSH frame is always sent on a request stream. Receipt of a DUPLICATE_PUSH frame on any other stream MUST be treated as a connection error of type H3_FRAME_UNEXPECTED.
-A client MUST NOT send a DUPLICATE_PUSH frame. A server MUST treat the receipt of a DUPLICATE_PUSH frame as a connection error of type H3_FRAME_UNEXPECTED.
- - +The MAX_PUSH_ID frame carries a single variable-length integer that identifies +the maximum value for a Push ID that the server can use (see +Section 7.2.5). A MAX_PUSH_ID frame cannot reduce the maximum Push ID; +receipt of a MAX_PUSH_ID that contains a smaller value than previously received +MUST be treated as a connection error of type H3_ID_ERROR.¶
+The DUPLICATE_PUSH frame (type=0xE) is used by servers to indicate that an +existing pushed resource is related to multiple client requests.¶
+The DUPLICATE_PUSH frame is always sent on a request stream. Receipt of a +DUPLICATE_PUSH frame on any other stream MUST be treated as a connection error +of type H3_FRAME_UNEXPECTED.¶
+A client MUST NOT send a DUPLICATE_PUSH frame. A server MUST treat the receipt +of a DUPLICATE_PUSH frame as a connection error of type H3_FRAME_UNEXPECTED.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1224,987 +2723,1733 @@-| Push ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 11: DUPLICATE_PUSH Frame Payload
-The DUPLICATE_PUSH frame carries a single variable-length integer that identifies the Push ID of a resource that the server has previously promised (see Section 7.2.5), though that promise might not be received before this frame. A server MUST NOT use a Push ID that is larger than the client has provided in a MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat receipt of a DUPLICATE_PUSH that contains a larger Push ID than the client has advertised as a connection error of type H3_ID_ERROR.
-This frame allows the server to use the same server push in response to multiple concurrent requests. Referencing the same server push ensures that a promise can be made in relation to every response in which server push might be needed without duplicating request headers or pushed responses.
-Allowing duplicate references to the same Push ID is primarily to reduce duplication caused by concurrent requests. A server SHOULD avoid reusing a Push ID over a long period. Clients are likely to consume server push responses and not retain them for reuse over time. Clients that see a DUPLICATE_PUSH that uses a Push ID that they have since consumed and discarded are forced to ignore the DUPLICATE_PUSH.
-Frame types of the format 0x1f * N + 0x21 for integer values of N are reserved to exercise the requirement that unknown types be ignored (Section 9). These frames have no semantics, and can be sent on any open stream when application-layer padding is desired. They MAY also be sent on connections where no data is currently being transferred. Endpoints MUST NOT consider these frames to have any meaning upon receipt.
-The payload and length of the frames are selected in any manner the implementation chooses.
-Frame types which were used in HTTP/2 where there is no corresponding HTTP/3 frame have also been reserved (Section 11.2). These frame types MUST NOT be sent, and receipt MAY be treated as an error of type H3_FRAME_UNEXPECTED.
-QUIC allows the application to abruptly terminate (reset) individual streams or the entire connection when an error is encountered. These are referred to as “stream errors” or “connection errors” and are described in more detail in [QUIC-TRANSPORT]. An endpoint MAY choose to treat a stream error as a connection error.
-Because new error codes can be defined without negotiation (see Section 9), receipt of an unknown error code or use of an error code in an unexpected context MUST NOT be treated as an error. However, closing a stream can constitute an error regardless of the error code (see Section 4.1).
-This section describes HTTP/3-specific error codes which can be used to express the cause of a connection or stream error.
-The following error codes are defined for use when abruptly terminating streams, aborting reading of streams, or immediately closing connections.
- - -The DUPLICATE_PUSH frame carries a single variable-length integer that +identifies the Push ID of a resource that the server has previously promised +(see Section 7.2.5), though that promise might not be received before +this frame. A server MUST NOT use a Push ID that is larger than the client has +provided in a MAX_PUSH_ID frame (Section 7.2.7). A client MUST treat +receipt of a DUPLICATE_PUSH that contains a larger Push ID than the client has +advertised as a connection error of type H3_ID_ERROR.¶
+This frame allows the server to use the same server push in response to multiple +concurrent requests. Referencing the same server push ensures that a promise +can be made in relation to every response in which server push might be needed +without duplicating request headers or pushed responses.¶
+Allowing duplicate references to the same Push ID is primarily to reduce +duplication caused by concurrent requests. A server SHOULD avoid reusing a Push +ID over a long period. Clients are likely to consume server push responses and +not retain them for reuse over time. Clients that see a DUPLICATE_PUSH that +uses a Push ID that they have since consumed and discarded are forced to ignore +the DUPLICATE_PUSH.¶
+Frame types of the format 0x1f * N + 0x21 for integer values of N are reserved
+to exercise the requirement that unknown types be ignored (Section 9).
+These frames have no semantics, and can be sent on any open stream when
+application-layer padding is desired. They MAY also be sent on connections where
+no data is currently being transferred. Endpoints MUST NOT consider these frames
+to have any meaning upon receipt.¶
The payload and length of the frames are selected in any manner the +implementation chooses.¶
+Frame types which were used in HTTP/2 where there is no corresponding HTTP/3 +frame have also been reserved (Section 11.2.1). These frame types MUST NOT be +sent, and receipt MAY be treated as an error of type H3_FRAME_UNEXPECTED.¶
+QUIC allows the application to abruptly terminate (reset) individual streams or +the entire connection when an error is encountered. These are referred to as +"stream errors" or "connection errors" and are described in more detail in +[QUIC-TRANSPORT]. An endpoint MAY choose to treat a stream error as a +connection error.¶
+Because new error codes can be defined without negotiation (see Section 9), +receipt of an unknown error code or use of an error code in an unexpected +context MUST NOT be treated as an error. However, closing a stream can +constitute an error regardless of the error code (see Section 4.1).¶
+This section describes HTTP/3-specific error codes which can be used to express +the cause of a connection or stream error.¶
+The following error codes are defined for use when abruptly terminating streams, +aborting reading of streams, or immediately closing connections.¶
+HTTP/3 permits extension of the protocol. Within the limitations described in this section, protocol extensions can be used to provide additional services or alter any aspect of the protocol. Extensions are effective only within the scope of a single HTTP/3 connection.
-This applies to the protocol elements defined in this document. This does not affect the existing options for extending HTTP, such as defining new methods, status codes, or header fields.
-Extensions are permitted to use new frame types (Section 7.2), new settings (Section 7.2.4.1), new error codes (Section 8), or new unidirectional stream types (Section 6.2). Registries are established for managing these extension points: frame types (Section 11.2), settings (Section 11.3), error codes (Section 11.4), and stream types (Section 11.5).
-Implementations MUST ignore unknown or unsupported values in all extensible protocol elements. Implementations MUST discard frames and unidirectional streams that have unknown or unsupported types. This means that any of these extension points can be safely used by extensions without prior arrangement or negotiation. However, where a known frame type is required to be in a specific location, such as the SETTINGS frame as the first frame of the control stream (see Section 6.2.1), an unknown frame type does not satisfy that requirement and SHOULD be treated as an error.
-Extensions that could change the semantics of existing protocol components MUST be negotiated before being used. For example, an extension that changes the layout of the HEADERS frame cannot be used until the peer has given a positive signal that this is acceptable. In this case, it could also be necessary to coordinate when the revised layout comes into effect.
-This document doesn’t mandate a specific method for negotiating the use of an extension but notes that a setting (Section 7.2.4.1) could be used for that purpose. If both peers set a value that indicates willingness to use the extension, then the extension can be used. If a setting is used for extension negotiation, the default value MUST be defined in such a fashion that the extension is disabled if the setting is omitted.
-The security considerations of HTTP/3 should be comparable to those of HTTP/2 with TLS; the considerations from Section 10 of [HTTP2] apply in addition to those listed here.
-When HTTP Alternative Services is used for discovery for HTTP/3 endpoints, the security considerations of [ALTSVC] also apply.
-Where HTTP/2 employs PADDING frames and Padding fields in other frames to make a connection more resistant to traffic analysis, HTTP/3 can either rely on transport-layer padding or employ the reserved frame and stream types discussed in Section 7.2.9 and Section 6.2.3. These methods of padding produce different results in terms of the granularity of padding, the effect of packet loss and recovery, and how an implementation might control padding.
-Several protocol elements contain nested length elements, typically in the form of frames with an explicit length containing variable-length integers. This could pose a security risk to an incautious implementer. An implementation MUST ensure that the length of a frame exactly matches the length of the fields it contains.
-The use of 0-RTT with HTTP/3 creates an exposure to replay attack. The anti-replay mitigations in [HTTP-REPLAY] MUST be applied when using HTTP/3 with 0-RTT.
-Certain HTTP implementations use the client address for logging or access-control purposes. Since a QUIC client’s address might change during a connection (and future versions might support simultaneous use of multiple addresses), such implementations will need to either actively retrieve the client’s current address or addresses when they are relevant or explicitly accept that the original address might change.
-This document creates a new registration for the identification of HTTP/3 in the “Application Layer Protocol Negotiation (ALPN) Protocol IDs” registry established in [RFC7301].
-The “h3” string identifies HTTP/3:
- - -HTTP/3 permits extension of the protocol. Within the limitations described in +this section, protocol extensions can be used to provide additional services or +alter any aspect of the protocol. Extensions are effective only within the +scope of a single HTTP/3 connection.¶
+This applies to the protocol elements defined in this document. This does not +affect the existing options for extending HTTP, such as defining new methods, +status codes, or header fields.¶
+Extensions are permitted to use new frame types (Section 7.2), new settings +(Section 7.2.4.1), new error codes (Section 8), or new unidirectional +stream types (Section 6.2). Registries are established for +managing these extension points: frame types (Section 11.2.1), settings +(Section 11.2.2), error codes (Section 11.2.3), and stream types +(Section 11.2.4).¶
+Implementations MUST ignore unknown or unsupported values in all extensible +protocol elements. Implementations MUST discard frames and unidirectional +streams that have unknown or unsupported types. This means that any of these +extension points can be safely used by extensions without prior arrangement or +negotiation. However, where a known frame type is required to be in a specific +location, such as the SETTINGS frame as the first frame of the control stream +(see Section 6.2.1), an unknown frame type does not satisfy that +requirement and SHOULD be treated as an error.¶
+Extensions that could change the semantics of existing protocol components MUST +be negotiated before being used. For example, an extension that changes the +layout of the HEADERS frame cannot be used until the peer has given a positive +signal that this is acceptable. Coordinating when such a revised layout comes +into effect could prove complex. As such, allocating new identifiers for +new definitions of existing protocol elements is likely to be more effective.¶
+This document doesn't mandate a specific method for negotiating the use of an +extension but notes that a setting (Section 7.2.4.1) could be used for +that purpose. If both peers set a value that indicates willingness to use the +extension, then the extension can be used. If a setting is used for extension +negotiation, the default value MUST be defined in such a fashion that the +extension is disabled if the setting is omitted.¶
+The security considerations of HTTP/3 should be comparable to those of HTTP/2 +with TLS; the considerations from Section 10 of [HTTP2] apply in addition to +those listed here.¶
+When HTTP Alternative Services is used for discovery for HTTP/3 endpoints, the +security considerations of [ALTSVC] also apply.¶
+Where HTTP/2 employs PADDING frames and Padding fields in other frames to make a +connection more resistant to traffic analysis, HTTP/3 can either rely on +transport-layer padding or employ the reserved frame and stream types discussed +in Section 7.2.9 and Section 6.2.3. These methods of padding produce +different results in terms of the granularity of padding, the effect of packet +loss and recovery, and how an implementation might control padding.¶
+Several protocol elements contain nested length elements, typically in the form +of frames with an explicit length containing variable-length integers. This +could pose a security risk to an incautious implementer. An implementation MUST +ensure that the length of a frame exactly matches the length of the fields it +contains.¶
+The use of 0-RTT with HTTP/3 creates an exposure to replay attack. The +anti-replay mitigations in [HTTP-REPLAY] MUST be applied when using +HTTP/3 with 0-RTT.¶
+Certain HTTP implementations use the client address for logging or +access-control purposes. Since a QUIC client's address might change during a +connection (and future versions might support simultaneous use of multiple +addresses), such implementations will need to either actively retrieve the +client's current address or addresses when they are relevant or explicitly +accept that the original address might change.¶
+This document registers a new ALPN protocol ID (Section 11.1) and creates new +registries that manage the assignment of codepoints in HTTP/3.¶
+This document creates a new registration for the identification of +HTTP/3 in the "Application Layer Protocol Negotiation (ALPN) +Protocol IDs" registry established in [RFC7301].¶
+The "h3" string identifies HTTP/3:¶
+This document establishes a registry for HTTP/3 frame type codes. The “HTTP/3 Frame Type” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-While this registry is separate from the “HTTP/2 Frame Type” registry defined in [HTTP2], it is preferable that the assignments parallel each other where the code spaces overlap. If an entry is present in only one registry, every effort SHOULD be made to avoid assigning the corresponding value to an unrelated operation.
-New entries in this registry require the following information:
- - -New registries created in this document operate under the QUIC registration +policy documented in Section 22.1 of [QUIC-TRANSPORT]. These registries all +include the common set of fields listed in Section 22.1.1 of [QUIC-TRANSPORT].¶
+The initial allocations in these registries created in this document are all +assigned permanent status and list as contact both the IESG (iesg@ietf.org) and +the HTTP working group (ietf-http-wg@w3.org).¶
+This document establishes a registry for HTTP/3 frame type codes. The "HTTP/3 +Frame Type" registry governs a 62-bit space. This registry follows the QUIC +registry policy; see Section 11.2. Permanent registrations in this registry +are assigned using the Specification Required policy [RFC8126], except for +values between 0x00 and 0x3f (in hexadecimal; inclusive), which are assigned +using Standards Action or IESG Approval as defined in Section 4.9 and 4.10 of +[RFC8126].¶
+While this registry is separate from the "HTTP/2 Frame Type" registry defined in +[HTTP2], it is preferable that the assignments parallel each other where the +code spaces overlap. If an entry is present in only one registry, every effort +SHOULD be made to avoid assigning the corresponding value to an unrelated +operation.¶
+In addition to common fields as described in Section 11.2, permanent +registrations in this registry MUST include the following field:¶
+The entries in the following table are registered by this document.
-| Frame Type | -Code | -Specification | -
|---|---|---|
| DATA | -0x0 | -Section 7.2.1 | -
| HEADERS | -0x1 | -Section 7.2.2 | -
| Reserved | -0x2 | -N/A | -
| CANCEL_PUSH | -0x3 | -Section 7.2.3 | -
| SETTINGS | -0x4 | -Section 7.2.4 | -
| PUSH_PROMISE | -0x5 | -Section 7.2.5 | -
| Reserved | -0x6 | -N/A | -
| GOAWAY | -0x7 | -Section 7.2.6 | -
| Reserved | -0x8 | -N/A | -
| Reserved | -0x9 | -N/A | -
| MAX_PUSH_ID | -0xD | -Section 7.2.7 | -
| DUPLICATE_PUSH | -0xE | -Section 7.2.8 | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-This document establishes a registry for HTTP/3 settings. The “HTTP/3 Settings” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126]. The designated experts are the same as those for the “HTTP/2 Settings” registry defined in [HTTP2].
-While this registry is separate from the “HTTP/2 Settings” registry defined in [HTTP2], it is preferable that the assignments parallel each other. If an entry is present in only one registry, every effort SHOULD be made to avoid assigning the corresponding value to an unrelated operation.
-New registrations are advised to provide the following information:
- - -Specifications of frame types MUST include a description of the frame layout and +its semantics, including any parts of the frame that are conditionally present.¶
+The entries in Table 2 are registered by this document.¶
+| Frame Type | +Value | +Specification | +
|---|---|---|
| DATA | +0x0 | ++ Section 7.2.1 + | +
| HEADERS | +0x1 | ++ Section 7.2.2 + | +
| Reserved | +0x2 | +N/A | +
| CANCEL_PUSH | +0x3 | ++ Section 7.2.3 + | +
| SETTINGS | +0x4 | ++ Section 7.2.4 + | +
| PUSH_PROMISE | +0x5 | ++ Section 7.2.5 + | +
| Reserved | +0x6 | +N/A | +
| GOAWAY | +0x7 | ++ Section 7.2.6 + | +
| Reserved | +0x8 | +N/A | +
| Reserved | +0x9 | +N/A | +
| MAX_PUSH_ID | +0xD | ++ Section 7.2.7 + | +
| DUPLICATE_PUSH | +0xE | ++ Section 7.2.8 + | +
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N
+(that is, 0x21, 0x40, ..., through 0x3FFFFFFFFFFFFFFE) MUST NOT be
+assigned by IANA.¶
This document establishes a registry for HTTP/3 settings. The "HTTP/3 Settings" +registry governs a 62-bit space. This registry follows the QUIC registry +policy; see Section 11.2. Permanent registrations in this registry are +assigned using the Specification Required policy [RFC8126], except for values +between 0x00 and 0x3f (in hexadecimal; inclusive), which are assigned using +Standards Action or IESG Approval as defined in Section 4.9 and 4.10 of +[RFC8126].¶
+While this registry is separate from the "HTTP/2 Settings" registry defined in +[HTTP2], it is preferable that the assignments parallel each other. If an +entry is present in only one registry, every effort SHOULD be made to avoid +assigning the corresponding value to an unrelated operation.¶
+In addition to common fields as described in Section 11.2, permanent +registrations in this registry MUST include the following fields:¶
+The entries in the following table are registered by this document.
-| Setting Name | -Code | -Specification | -Default | -
|---|---|---|---|
| Reserved | -0x2 | -N/A | -N/A | -
| Reserved | -0x3 | -N/A | -N/A | -
| Reserved | -0x4 | -N/A | -N/A | -
| Reserved | -0x5 | -N/A | -N/A | -
| MAX_HEADER_LIST_SIZE | -0x6 | -Section 7.2.4.1 | -Unlimited | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-This document establishes a registry for HTTP/3 error codes. The “HTTP/3 Error Code” registry manages a 62-bit space. The “HTTP/3 Error Code” registry operates under the “Expert Review” policy [RFC8126].
-Registrations for error codes are required to include a description of the error code. An expert reviewer is advised to examine new registrations for possible duplication with existing error codes. Use of existing registrations is to be encouraged, but not mandated.
-New registrations are advised to provide the following information:
- - -The entries in Table 3 are registered by this document.¶
+| Setting Name | +Value | +Specification | +Default | +
|---|---|---|---|
| Reserved | +0x2 | +N/A | +N/A | +
| Reserved | +0x3 | +N/A | +N/A | +
| Reserved | +0x4 | +N/A | +N/A | +
| Reserved | +0x5 | +N/A | +N/A | +
| MAX_HEADER_LIST_SIZE | +0x6 | ++ Section 7.2.4.1 + | +Unlimited | +
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N
+(that is, 0x21, 0x40, ..., through 0x3FFFFFFFFFFFFFFE) MUST NOT be
+assigned by IANA.¶
This document establishes a registry for HTTP/3 error codes. The "HTTP/3 Error +Code" registry manages a 62-bit space. This registry follows the QUIC registry +policy; see Section 11.2. Permanent registrations in this registry are +assigned using the Specification Required policy [RFC8126], except for values +between 0x00 and 0x3f (in hexadecimal; inclusive), which are assigned using +Standards Action or IESG Approval as defined in Section 4.9 and 4.10 of +[RFC8126].¶
+Registrations for error codes are required to include a description +of the error code. An expert reviewer is advised to examine new +registrations for possible duplication with existing error codes. +Use of existing registrations is to be encouraged, but not mandated.¶
+In addition to common fields as described in Section 11.2, permanent +registrations in this registry MUST include the following fields:¶
+The entries in the following table are registered by this document.
-| Name | -Code | -Description | -Specification | -
|---|---|---|---|
| H3_NO_ERROR | -0x0100 | -No error | -Section 8.1 | -
| H3_GENERAL_PROTOCOL_ERROR | -0x0101 | -General protocol error | -Section 8.1 | -
| H3_INTERNAL_ERROR | -0x0102 | -Internal error | -Section 8.1 | -
| H3_STREAM_CREATION_ERROR | -0x0103 | -Stream creation error | -Section 8.1 | -
| H3_CLOSED_CRITICAL_STREAM | -0x0104 | -Critical stream was closed | -Section 8.1 | -
| H3_FRAME_UNEXPECTED | -0x0105 | -Frame not permitted in the current state | -Section 8.1 | -
| H3_FRAME_ERROR | -0x0106 | -Frame violated layout or size rules | -Section 8.1 | -
| H3_EXCESSIVE_LOAD | -0x0107 | -Peer generating excessive load | -Section 8.1 | -
| H3_ID_ERROR | -0x0108 | -An identifier was used incorrectly | -Section 8.1 | -
| H3_SETTINGS_ERROR | -0x0109 | -SETTINGS frame contained invalid values | -Section 8.1 | -
| H3_MISSING_SETTINGS | -0x010A | -No SETTINGS frame received | -Section 8.1 | -
| H3_REQUEST_REJECTED | -0x010B | -Request not processed | -Section 8.1 | -
| H3_REQUEST_CANCELLED | -0x010C | -Data no longer needed | -Section 8.1 | -
| H3_REQUEST_INCOMPLETE | -0x010D | -Stream terminated early | -Section 8.1 | -
| H3_EARLY_RESPONSE | -0x010E | -Remainder of request not needed | -Section 8.1 | -
| H3_CONNECT_ERROR | -0x010F | -TCP reset or error on CONNECT request | -Section 8.1 | -
| H3_VERSION_FALLBACK | -0x0110 | -Retry over HTTP/1.1 | -Section 8.1 | -
This document establishes a registry for HTTP/3 unidirectional stream types. The “HTTP/3 Stream Type” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-New entries in this registry require the following information:
- - -The entries in the Table 4 are registered by this document.¶
+| Name | +Value | +Description | +Specification | +
|---|---|---|---|
| H3_NO_ERROR | +0x0100 | +No error | ++ Section 8.1 + | +
| H3_GENERAL_PROTOCOL_ERROR | +0x0101 | +General protocol error | ++ Section 8.1 + | +
| H3_INTERNAL_ERROR | +0x0102 | +Internal error | ++ Section 8.1 + | +
| H3_STREAM_CREATION_ERROR | +0x0103 | +Stream creation error | ++ Section 8.1 + | +
| H3_CLOSED_CRITICAL_STREAM | +0x0104 | +Critical stream was closed | ++ Section 8.1 + | +
| H3_FRAME_UNEXPECTED | +0x0105 | +Frame not permitted in the current state | ++ Section 8.1 + | +
| H3_FRAME_ERROR | +0x0106 | +Frame violated layout or size rules | ++ Section 8.1 + | +
| H3_EXCESSIVE_LOAD | +0x0107 | +Peer generating excessive load | ++ Section 8.1 + | +
| H3_ID_ERROR | +0x0108 | +An identifier was used incorrectly | ++ Section 8.1 + | +
| H3_SETTINGS_ERROR | +0x0109 | +SETTINGS frame contained invalid values | ++ Section 8.1 + | +
| H3_MISSING_SETTINGS | +0x010A | +No SETTINGS frame received | ++ Section 8.1 + | +
| H3_REQUEST_REJECTED | +0x010B | +Request not processed | ++ Section 8.1 + | +
| H3_REQUEST_CANCELLED | +0x010C | +Data no longer needed | ++ Section 8.1 + | +
| H3_REQUEST_INCOMPLETE | +0x010D | +Stream terminated early | ++ Section 8.1 + | +
| H3_CONNECT_ERROR | +0x010F | +TCP reset or error on CONNECT request | ++ Section 8.1 + | +
| H3_VERSION_FALLBACK | +0x0110 | +Retry over HTTP/1.1 | ++ Section 8.1 + | +
This document establishes a registry for HTTP/3 unidirectional stream types. The +"HTTP/3 Stream Type" registry governs a 62-bit space. This registry follows the +QUIC registry policy; see Section 11.2. Permanent registrations in this +registry are assigned using the Specification Required policy [RFC8126], +except for values between 0x00 and 0x3f (in hexadecimal; inclusive), which are +assigned using Standards Action or IESG Approval as defined in Section 4.9 and +4.10 of [RFC8126].¶
+In addition to common fields as described in Section 11.2, permanent +registrations in this registry MUST include the following fields:¶
+The entries in the following table are registered by this document.
-| Stream Type | -Code | -Specification | -Sender | -
|---|---|---|---|
| Control Stream | -0x00 | -Section 6.2.1 | -Both | -
| Push Stream | -0x01 | -Section 4.4 | -Server | -
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N (that is, 0x21, 0x40, …, through 0x3FFFFFFFFFFFFFFE) MUST NOT be assigned by IANA.
-| [ALTSVC] | --Nottingham, M., McManus, P. and J. Reschke, "HTTP Alternative Services", RFC 7838, DOI 10.17487/RFC7838, April 2016. | -
| [HTTP-REPLAY] | --Thomson, M., Nottingham, M. and W. Tarreau, "Using Early Data in HTTP", RFC 8470, DOI 10.17487/RFC8470, September 2018. | -
| [HTTP2] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| [QPACK] | --Krasic, C., Bishop, M. and A. Frindell, "QPACK: Header Compression for HTTP over QUIC", Internet-Draft draft-ietf-quic-qpack, November 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, November 2019. | -
| [RFC0793] | --Postel, J., "Transmission Control Protocol", STD 7, RFC 793, DOI 10.17487/RFC0793, September 1981. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC5234] | --Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008. | -
| [RFC6066] | --Eastlake 3rd, D., "Transport Layer Security (TLS) Extensions: Extension Definitions", RFC 6066, DOI 10.17487/RFC6066, January 2011. | -
| [RFC6265] | --Barth, A., "HTTP State Management Mechanism", RFC 6265, DOI 10.17487/RFC6265, April 2011. | -
| [RFC7230] | --Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, DOI 10.17487/RFC7230, June 2014. | -
| [RFC7231] | --Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI 10.17487/RFC7231, June 2014. | -
| [RFC7838] | --Nottingham, M., McManus, P. and J. Reschke, "HTTP Alternative Services", RFC 7838, DOI 10.17487/RFC7838, April 2016. | -
| [RFC8126] | --Cotton, M., Leiba, B. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 8126, DOI 10.17487/RFC8126, June 2017. | -
| [RFC8164] | --Nottingham, M. and M. Thomson, "Opportunistic Security for HTTP/2", RFC 8164, DOI 10.17487/RFC8164, May 2017. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [HPACK] | --Peon, R. and H. Ruellan, "HPACK: Header Compression for HTTP/2", RFC 7541, DOI 10.17487/RFC7541, May 2015. | -
| [RFC6585] | --Nottingham, M. and R. Fielding, "Additional HTTP Status Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC7413] | --Cheng, Y., Chu, J., Radhakrishnan, S. and A. Jain, "TCP Fast Open", RFC 7413, DOI 10.17487/RFC7413, December 2014. | -
HTTP/3 is strongly informed by HTTP/2, and bears many similarities. This section describes the approach taken to design HTTP/3, points out important differences from HTTP/2, and describes how to map HTTP/2 extensions into HTTP/3.
-HTTP/3 begins from the premise that similarity to HTTP/2 is preferable, but not a hard requirement. HTTP/3 departs from HTTP/2 where QUIC differs from TCP, either to take advantage of QUIC features (like streams) or to accommodate important shortcomings (such as a lack of total ordering). These differences make HTTP/3 similar to HTTP/2 in key aspects, such as the relationship of requests and responses to streams. However, the details of the HTTP/3 design are substantially different than HTTP/2.
-These departures are noted in this section.
-HTTP/3 permits use of a larger number of streams (2^62-1) than HTTP/2. The considerations about exhaustion of stream identifier space apply, though the space is significantly larger such that it is likely that other limits in QUIC are reached first, such as the limit on the connection flow control window.
-In contrast to HTTP/2, stream concurrency in HTTP/3 is managed by QUIC. QUIC considers a stream closed when all data has been received and sent data has been acknowledged by the peer. HTTP/2 considers a stream closed when the frame containing the END_STREAM bit has been committed to the transport. As a result, the stream for an equivalent exchange could remain “active” for a longer period of time. HTTP/3 servers might choose to permit a larger number of concurrent client-initiated bidirectional streams to achieve equivalent concurrency to HTTP/2, depending on the expected usage patterns.
-Due to the presence of other unidirectional stream types, HTTP/3 does not rely exclusively on the number of concurrent unidirectional streams to control the number of concurrent in-flight pushes. Instead, HTTP/3 clients use the MAX_PUSH_ID frame to control the number of pushes received from an HTTP/3 server.
-Many framing concepts from HTTP/2 can be elided on QUIC, because the transport deals with them. Because frames are already on a stream, they can omit the stream number. Because frames do not block multiplexing (QUIC’s multiplexing occurs below this layer), the support for variable-maximum-length packets can be removed. Because stream termination is handled by QUIC, an END_STREAM flag is not required. This permits the removal of the Flags field from the generic frame layout.
-Frame payloads are largely drawn from [HTTP2]. However, QUIC includes many features (e.g., flow control) which are also present in HTTP/2. In these cases, the HTTP mapping does not re-implement them. As a result, several HTTP/2 frame types are not required in HTTP/3. Where an HTTP/2-defined frame is no longer used, the frame ID has been reserved in order to maximize portability between HTTP/2 and HTTP/3 implementations. However, even equivalent frames between the two mappings are not identical.
-Many of the differences arise from the fact that HTTP/2 provides an absolute ordering between frames across all streams, while QUIC provides this guarantee on each stream only. As a result, if a frame type makes assumptions that frames from different streams will still be received in the order sent, HTTP/3 will break them.
-Some examples of feature adaptations are described below, as well as general guidance to extension frame implementors converting an HTTP/2 extension to HTTP/3.
-HTTP/2 specifies priority assignments in PRIORITY frames and (optionally) in HEADERS frames. HTTP/3 does not provide a means of signaling priority.
-Note that while there is no explicit signaling for priority, this does not mean that prioritization is not important for achieving good performance.
-HPACK was designed with the assumption of in-order delivery. A sequence of encoded header blocks must arrive (and be decoded) at an endpoint in the same order in which they were encoded. This ensures that the dynamic state at the two endpoints remains in sync.
-Because this total ordering is not provided by QUIC, HTTP/3 uses a modified version of HPACK, called QPACK. QPACK uses a single unidirectional stream to make all modifications to the dynamic table, ensuring a total order of updates. All frames which contain encoded headers merely reference the table state at a given time without modifying it.
-[QPACK] provides additional details.
-Frame type definitions in HTTP/3 often use the QUIC variable-length integer encoding. In particular, Stream IDs use this encoding, which allows for a larger range of possible values than the encoding used in HTTP/2. Some frames in HTTP/3 use an identifier rather than a Stream ID (e.g., Push IDs). Redefinition of the encoding of extension frame types might be necessary if the encoding includes a Stream ID.
-Because the Flags field is not present in generic HTTP/3 frames, those frames which depend on the presence of flags need to allocate space for flags as part of their frame payload.
-Other than this issue, frame type HTTP/2 extensions are typically portable to QUIC simply by replacing Stream 0 in HTTP/2 with a control stream in HTTP/3. HTTP/3 extensions will not assume ordering, but would not be harmed by ordering, and would be portable to HTTP/2 in the same manner.
-Specifications for permanent registrations MUST include a description of the +stream type, including the layout semantics of the stream contents.¶
+The entries in the following table are registered by this document.¶
+| Stream Type | +Value | +Specification | +Sender | +
|---|---|---|---|
| Control Stream | +0x00 | ++ Section 6.2.1 + | +Both | +
| Push Stream | +0x01 | ++ Section 4.4 + | +Server | +
Additionally, each code of the format 0x1f * N + 0x21 for integer values of N
+(that is, 0x21, 0x40, ..., through 0x3FFFFFFFFFFFFFFE) MUST NOT be
+assigned by IANA.¶
Frame types defined by extensions to HTTP/2 need to be separately registered for HTTP/3 if still applicable. The IDs of frames defined in [HTTP2] have been reserved for simplicity. Note that the frame type space in HTTP/3 is substantially larger (62 bits versus 8 bits), so many HTTP/3 frame types have no equivalent HTTP/2 code points. See Section 11.2.
-An important difference from HTTP/2 is that settings are sent once, as the first frame of the control stream, and thereafter cannot change. This eliminates many corner cases around synchronization of changes.
-Some transport-level options that HTTP/2 specifies via the SETTINGS frame are superseded by QUIC transport parameters in HTTP/3. The HTTP-level options that are retained in HTTP/3 have the same value as in HTTP/2.
-Below is a listing of how each HTTP/2 SETTINGS parameter is mapped:
- - -In HTTP/3, setting values are variable-length integers (6, 14, 30, or 62 bits long) rather than fixed-length 32-bit fields as in HTTP/2. This will often produce a shorter encoding, but can produce a longer encoding for settings which use the full 32-bit space. Settings ported from HTTP/2 might choose to redefine the format of their settings to avoid using the 62-bit encoding.
-Settings need to be defined separately for HTTP/2 and HTTP/3. The IDs of settings defined in [HTTP2] have been reserved for simplicity. Note that the settings identifier space in HTTP/3 is substantially larger (62 bits versus 16 bits), so many HTTP/3 settings have no equivalent HTTP/2 code point. See Section 11.3.
-As QUIC streams might arrive out-of-order, endpoints are advised to not wait for the peers’ settings to arrive before responding to other streams. See Section 7.2.4.2.
-QUIC has the same concepts of “stream” and “connection” errors that HTTP/2 provides. However, there is no direct portability of HTTP/2 error codes to HTTP/3 error codes; the values are shifted in order to prevent accidental or implicit conversion.
-The HTTP/2 error codes defined in Section 7 of [HTTP2] logically map to the HTTP/3 error codes as follows:
- - -HTTP/3 is strongly informed by HTTP/2, and bears many similarities. This +section describes the approach taken to design HTTP/3, points out important +differences from HTTP/2, and describes how to map HTTP/2 extensions into HTTP/3.¶
+HTTP/3 begins from the premise that similarity to HTTP/2 is preferable, but not +a hard requirement. HTTP/3 departs from HTTP/2 where QUIC differs from TCP, +either to take advantage of QUIC features (like streams) or to accommodate +important shortcomings (such as a lack of total ordering). These differences +make HTTP/3 similar to HTTP/2 in key aspects, such as the relationship of +requests and responses to streams. However, the details of the HTTP/3 design are +substantially different than HTTP/2.¶
+These departures are noted in this section.¶
+HTTP/3 permits use of a larger number of streams (2^62-1) than HTTP/2. The +considerations about exhaustion of stream identifier space apply, though the +space is significantly larger such that it is likely that other limits in QUIC +are reached first, such as the limit on the connection flow control window.¶
+In contrast to HTTP/2, stream concurrency in HTTP/3 is managed by QUIC. QUIC +considers a stream closed when all data has been received and sent data has been +acknowledged by the peer. HTTP/2 considers a stream closed when the frame +containing the END_STREAM bit has been committed to the transport. As a result, +the stream for an equivalent exchange could remain "active" for a longer period +of time. HTTP/3 servers might choose to permit a larger number of concurrent +client-initiated bidirectional streams to achieve equivalent concurrency to +HTTP/2, depending on the expected usage patterns.¶
+Due to the presence of other unidirectional stream types, HTTP/3 does not rely +exclusively on the number of concurrent unidirectional streams to control the +number of concurrent in-flight pushes. Instead, HTTP/3 clients use the +MAX_PUSH_ID frame to control the number of pushes received from an HTTP/3 +server.¶
+Many framing concepts from HTTP/2 can be elided on QUIC, because the transport +deals with them. Because frames are already on a stream, they can omit the +stream number. Because frames do not block multiplexing (QUIC's multiplexing +occurs below this layer), the support for variable-maximum-length packets can be +removed. Because stream termination is handled by QUIC, an END_STREAM flag is +not required. This permits the removal of the Flags field from the generic +frame layout.¶
+Frame payloads are largely drawn from [HTTP2]. However, QUIC includes many +features (e.g., flow control) which are also present in HTTP/2. In these cases, +the HTTP mapping does not re-implement them. As a result, several HTTP/2 frame +types are not required in HTTP/3. Where an HTTP/2-defined frame is no longer +used, the frame ID has been reserved in order to maximize portability between +HTTP/2 and HTTP/3 implementations. However, even equivalent frames between the +two mappings are not identical.¶
+Many of the differences arise from the fact that HTTP/2 provides an absolute +ordering between frames across all streams, while QUIC provides this guarantee +on each stream only. As a result, if a frame type makes assumptions that frames +from different streams will still be received in the order sent, HTTP/3 will +break them.¶
+Some examples of feature adaptations are described below, as well as general +guidance to extension frame implementors converting an HTTP/2 extension to +HTTP/3.¶
+HTTP/2 specifies priority assignments in PRIORITY frames and (optionally) in +HEADERS frames. HTTP/3 does not provide a means of signaling priority.¶
+Note that while there is no explicit signaling for priority, this does not mean +that prioritization is not important for achieving good performance.¶
+HPACK was designed with the assumption of in-order delivery. A sequence of +encoded header blocks must arrive (and be decoded) at an endpoint in the same +order in which they were encoded. This ensures that the dynamic state at the two +endpoints remains in sync.¶
+Because this total ordering is not provided by QUIC, HTTP/3 uses a modified +version of HPACK, called QPACK. QPACK uses a single unidirectional stream to +make all modifications to the dynamic table, ensuring a total order of updates. +All frames which contain encoded headers merely reference the table state at a +given time without modifying it.¶
+ +Frame type definitions in HTTP/3 often use the QUIC variable-length integer +encoding. In particular, Stream IDs use this encoding, which allows for a +larger range of possible values than the encoding used in HTTP/2. Some frames +in HTTP/3 use an identifier rather than a Stream ID (e.g., Push +IDs). Redefinition of the encoding of extension frame types might be necessary +if the encoding includes a Stream ID.¶
+Because the Flags field is not present in generic HTTP/3 frames, those frames +which depend on the presence of flags need to allocate space for flags as part +of their frame payload.¶
+Other than this issue, frame type HTTP/2 extensions are typically portable to +QUIC simply by replacing Stream 0 in HTTP/2 with a control stream in HTTP/3. +HTTP/3 extensions will not assume ordering, but would not be harmed by ordering, +and would be portable to HTTP/2 in the same manner.¶
+Error codes need to be defined for HTTP/2 and HTTP/3 separately. See Section 11.4.
-Frame types defined by extensions to HTTP/2 need to be separately registered for +HTTP/3 if still applicable. The IDs of frames defined in [HTTP2] have been +reserved for simplicity. Note that the frame type space in HTTP/3 is +substantially larger (62 bits versus 8 bits), so many HTTP/3 frame types have no +equivalent HTTP/2 code points. See Section 11.2.1.¶
+An important difference from HTTP/2 is that settings are sent once, as the first +frame of the control stream, and thereafter cannot change. This eliminates many +corner cases around synchronization of changes.¶
+Some transport-level options that HTTP/2 specifies via the SETTINGS frame are +superseded by QUIC transport parameters in HTTP/3. The HTTP-level options that +are retained in HTTP/3 have the same value as in HTTP/2.¶
+Below is a listing of how each HTTP/2 SETTINGS parameter is mapped:¶
+In HTTP/3, setting values are variable-length integers (6, 14, 30, or 62 bits +long) rather than fixed-length 32-bit fields as in HTTP/2. This will often +produce a shorter encoding, but can produce a longer encoding for settings which +use the full 32-bit space. Settings ported from HTTP/2 might choose to redefine +their value to limit it to 30 bits for more efficient encoding, or to make use +of the 62-bit space if more than 30 bits are required.¶
+Settings need to be defined separately for HTTP/2 and HTTP/3. The IDs of +settings defined in [HTTP2] have been reserved for simplicity. Note that +the settings identifier space in HTTP/3 is substantially larger (62 bits versus +16 bits), so many HTTP/3 settings have no equivalent HTTP/2 code point. See +Section 11.2.2.¶
+As QUIC streams might arrive out-of-order, endpoints are advised to not wait for +the peers' settings to arrive before responding to other streams. See +Section 7.2.4.2.¶
+QUIC has the same concepts of "stream" and "connection" errors that HTTP/2 +provides. However, there is no direct portability of HTTP/2 error codes to +HTTP/3 error codes; the values are shifted in order to prevent accidental +or implicit conversion.¶
+The HTTP/2 error codes defined in Section 7 of [HTTP2] logically map to +the HTTP/3 error codes as follows:¶
+Error codes need to be defined for HTTP/2 and HTTP/3 separately. See +Section 11.2.3.¶
+Further changes to error codes (#2662,#2551):¶
+http-opportunistic resource (RFC 8164) when scheme is
+http (#2439,#2973)¶
+Changes to SETTINGS frames in 0-RTT (#2972,#2790,#2945):¶
+No changes
-No changes¶
+Extensive changes to error codes and conditions of their sending¶
Use variable-length integers throughout (#2437,#2233,#2253,#2275)¶
Changes to PRIORITY frame (#1865, #2075)¶
Substantial editorial reorganization; no technical changes.
-Substantial editorial reorganization; no technical changes.¶
+None.
-None.¶
+The original authors of this specification were Robbie Shade and Mike Warres.
-A substantial portion of Mike’s contribution was supported by Microsoft during his employment there.
-SETTINGS changes (#181):¶
+ +The original authors of this specification were Robbie Shade and Mike Warres.¶
+A substantial portion of Mike's contribution was supported by Microsoft during +his employment there.¶
+| QUIC | -M. Thomson | -||||||
| Internet-Draft | -Mozilla | -
| Intended status: Standards Track | -November 12, 2019 | -
| Expires: May 15, 2020 | -- |
Version-Independent Properties of QUIC
- draft-ietf-quic-invariants-latest
This document defines the properties of the QUIC transport protocol that are expected to remain unchanged over time as new versions of the protocol are developed.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-invariants.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -This document defines the properties of the QUIC transport protocol that are +expected to remain unchanged over time as new versions of the protocol are +developed.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +https://mailarchive.ietf.org/arch/search/?email_list=quic.¶
+Working Group information can be found at https://github.com/quicwg; source +code and issues list for this draft can be found at +https://github.com/quicwg/base-drafts/labels/-invariants.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+In addition to providing secure, multiplexed transport, QUIC [QUIC-TRANSPORT] +includes the ability to negotiate a version. This allows the protocol to change +over time in response to new requirements. Many characteristics of the protocol +will change between versions.¶
+This document describes the subset of QUIC that is intended to remain stable as +new versions are developed and deployed. All of these invariants are +IP-version-independent.¶
+The primary goal of this document is to ensure that it is possible to deploy new +versions of QUIC. By documenting the properties that can't change, this +document aims to preserve the ability to change any other aspect of the +protocol. Thus, unless specifically described in this document, any aspect of +the protocol can change between different versions.¶
+Appendix A is a non-exhaustive list of some incorrect assumptions that +might be made based on knowledge of QUIC version 1; these do not apply to every +version of QUIC.¶
+The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+This document uses terms and notational conventions from [QUIC-TRANSPORT].¶
+QUIC is a connection-oriented protocol between two endpoints. Those endpoints +exchange UDP datagrams. These UDP datagrams contain QUIC packets. QUIC +endpoints use QUIC packets to establish a QUIC connection, which is shared +protocol state between those endpoints.¶
+A QUIC packet is the content of the UDP datagrams exchanged by QUIC endpoints. +This document describes the contents of those datagrams.¶
+QUIC defines two types of packet header: long and short. Packets with long +headers are identified by the most significant bit of the first byte being set; +packets with a short header have that bit cleared.¶
+Aside from the values described here, the payload of QUIC packets is +version-specific and of arbitrary length.¶
+Long headers take the form described in Figure 1. Bits that have +version-specific semantics are marked with an X.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -442,18 +1056,35 @@-|X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: QUIC Long Header
-A QUIC packet with a long header has the high bit of the first byte set to 1. All other bits in that byte are version specific.
-The next four bytes include a 32-bit Version field (see Section 4.4).
-The next byte contains the length in bytes of the Destination Connection ID (see Section 4.3) field that follows it. This length is encoded as an 8-bit unsigned integer. The Destination Connection ID field follows the DCID Len field and is between 0 and 255 bytes in length.
-The next byte contains the length in bytes of the Source Connection ID field that follows it. This length is encoded as a 8-bit unsigned integer. The Source Connection ID field follows the SCID Len field and is between 0 and 255 bytes in length.
-The remainder of the packet contains version-specific content.
-Short headers take the form described in Figure 2. Bits that have version-specific semantics are marked with an X.
- - +A QUIC packet with a long header has the high bit of the first byte set to 1. +All other bits in that byte are version specific.¶
+The next four bytes include a 32-bit Version field (see Section 4.4).¶
+The next byte contains the length in bytes of the Destination Connection ID (see +Section 4.3) field that follows it. This length is encoded as an 8-bit +unsigned integer. The Destination Connection ID field follows the DCID Len +field and is between 0 and 255 bytes in length.¶
+The next byte contains the length in bytes +of the Source Connection ID field that follows it. This length is encoded as +a 8-bit unsigned integer. The Source Connection ID field follows the SCID Len +field and is between 0 and 255 bytes in length.¶
+The remainder of the packet contains version-specific content.¶
+Short headers take the form described in Figure 2. Bits that have +version-specific semantics are marked with an X.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -465,28 +1096,69 @@-|X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: QUIC Short Header
-A QUIC packet with a short header has the high bit of the first byte set to 0.
-A QUIC packet with a short header includes a Destination Connection ID immediately following the first byte. The short header does not include the Connection ID Lengths, Source Connection ID, or Version fields. The length of the Destination Connection ID is not specified in packets with a short header and is not constrained by this specification.
-The remainder of the packet has version-specific semantics.
-A connection ID is an opaque field of arbitrary length.
-The primary function of a connection ID is to ensure that changes in addressing at lower protocol layers (UDP, IP, and below) don’t cause packets for a QUIC connection to be delivered to the wrong endpoint. The connection ID is used by endpoints and the intermediaries that support them to ensure that each QUIC packet can be delivered to the correct instance of an endpoint. At the endpoint, the connection ID is used to identify which QUIC connection the packet is intended for.
-The connection ID is chosen by each endpoint using version-specific methods. Packets for the same QUIC connection might use different connection ID values.
-QUIC versions are identified with a 32-bit integer, encoded in network byte order. Version 0 is reserved for version negotiation (see Section 5). All other version numbers are potentially valid.
-The properties described in this document apply to all versions of QUIC. A protocol that does not conform to the properties described in this document is not QUIC. Future documents might describe additional properties which apply to a specific QUIC version, or to a range of QUIC versions.
-A QUIC endpoint that receives a packet with a long header and a version it either does not understand or does not support might send a Version Negotiation packet in response. Packets with a short header do not trigger version negotiation.
-A Version Negotiation packet sets the high bit of the first byte, and thus it conforms with the format of a packet with a long header as defined in Section 4.1. A Version Negotiation packet is identifiable as such by the Version field, which is set to 0x00000000.
- - +A QUIC packet with a short header has the high bit of the first byte set to 0.¶
+A QUIC packet with a short header includes a Destination Connection ID +immediately following the first byte. The short header does not include the +Connection ID Lengths, Source Connection ID, or Version fields. The length of +the Destination Connection ID is not specified in packets with a short header +and is not constrained by this specification.¶
+The remainder of the packet has version-specific semantics.¶
+A connection ID is an opaque field of arbitrary length.¶
+The primary function of a connection ID is to ensure that changes in addressing +at lower protocol layers (UDP, IP, and below) don't cause packets for a QUIC +connection to be delivered to the wrong endpoint. The connection ID is used by +endpoints and the intermediaries that support them to ensure that each QUIC +packet can be delivered to the correct instance of an endpoint. At the +endpoint, the connection ID is used to identify which QUIC connection the packet +is intended for.¶
+The connection ID is chosen by each endpoint using version-specific methods. +Packets for the same QUIC connection might use different connection ID values.¶
+QUIC versions are identified with a 32-bit integer, encoded in network byte +order. Version 0 is reserved for version negotiation (see +Section 5). All other version numbers are potentially valid.¶
+The properties described in this document apply to all versions of QUIC. A +protocol that does not conform to the properties described in this document is +not QUIC. Future documents might describe additional properties which apply to +a specific QUIC version, or to a range of QUIC versions.¶
+A QUIC endpoint that receives a packet with a long header and a version it +either does not understand or does not support might send a Version Negotiation +packet in response. Packets with a short header do not trigger version +negotiation.¶
+A Version Negotiation packet sets the high bit of the first byte, and thus it +conforms with the format of a packet with a long header as defined in +Section 4.1. A Version Negotiation packet is identifiable as such by the +Version field, which is set to 0x00000000.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -512,106 +1184,203 @@-| [Supported Version N (32)] | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Version Negotiation Packet
-The Version Negotiation packet contains a list of Supported Version fields, each identifying a version that the endpoint sending the packet supports. The Supported Version fields follow the Version field. A Version Negotiation packet contains no other fields. An endpoint MUST ignore a packet that contains no Supported Version fields, or a truncated Supported Version.
-Version Negotiation packets do not use integrity or confidentiality protection. A specific QUIC version might authenticate the packet as part of its connection establishment process.
-An endpoint MUST include the value from the Source Connection ID field of the packet it receives in the Destination Connection ID field. The value for Source Connection ID MUST be copied from the Destination Connection ID of the received packet, which is initially randomly selected by a client. Echoing both connection IDs gives clients some assurance that the server received the packet and that the Version Negotiation packet was not generated by an off-path attacker.
-An endpoint that receives a Version Negotiation packet might change the version that it decides to use for subsequent packets. The conditions under which an endpoint changes QUIC version will depend on the version of QUIC that it chooses.
-See [QUIC-TRANSPORT] for a more thorough description of how an endpoint that supports QUIC version 1 generates and consumes a Version Negotiation packet.
-It is possible that middleboxes could use traits of a specific version of QUIC and assume that when other versions of QUIC exhibit similar traits the same underlying semantic is being expressed. There are potentially many such traits (see Appendix A). Some effort has been made to either eliminate or obscure some observable traits in QUIC version 1, but many of these remain. Other QUIC versions might make different design decisions and so exhibit different traits.
-The QUIC version number does not appear in all QUIC packets, which means that reliably extracting information from a flow based on version-specific traits requires that middleboxes retain state for every connection ID they see.
-The Version Negotiation packet described in this document is not integrity-protected; it only has modest protection against insertion by off-path attackers. QUIC versions MUST define a mechanism that authenticates the values it contains.
-This document makes no request of IANA.
-| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, November 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using Transport Layer Security (TLS) to Secure QUIC", Internet-Draft draft-ietf-quic-tls, November 2019. | -
| [RFC5116] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
There are several traits of QUIC version 1 [QUIC-TRANSPORT] that are not protected from observation, but are nonetheless considered to be changeable when a new version is deployed.
-This section lists a sampling of incorrect assumptions that might be made based on knowledge of QUIC version 1. Some of these statements are not even true for QUIC version 1. This is not an exhaustive list, it is intended to be illustrative only.
-The following statements are NOT guaranteed to be true for every QUIC version:
- - +The Version Negotiation packet contains a list of Supported Version fields, each +identifying a version that the endpoint sending the packet supports. The +Supported Version fields follow the Version field. A Version Negotiation packet +contains no other fields. An endpoint MUST ignore a packet that contains no +Supported Version fields, or a truncated Supported Version.¶
+Version Negotiation packets do not use integrity or confidentiality protection. +A specific QUIC version might authenticate the packet as part of its connection +establishment process.¶
+An endpoint MUST include the value from the Source Connection ID field of the +packet it receives in the Destination Connection ID field. The value for Source +Connection ID MUST be copied from the Destination Connection ID of the received +packet, which is initially randomly selected by a client. Echoing both +connection IDs gives clients some assurance that the server received the packet +and that the Version Negotiation packet was not generated by an off-path +attacker.¶
+An endpoint that receives a Version Negotiation packet might change the version +that it decides to use for subsequent packets. The conditions under which an +endpoint changes QUIC version will depend on the version of QUIC that it +chooses.¶
+See [QUIC-TRANSPORT] for a more thorough description of how an endpoint that +supports QUIC version 1 generates and consumes a Version Negotiation packet.¶
+It is possible that middleboxes could use traits of a specific version of QUIC +and assume that when other versions of QUIC exhibit similar traits the same +underlying semantic is being expressed. There are potentially many such traits +(see Appendix A). Some effort has been made to either eliminate or +obscure some observable traits in QUIC version 1, but many of these remain. +Other QUIC versions might make different design decisions and so exhibit +different traits.¶
+The QUIC version number does not appear in all QUIC packets, which means that +reliably extracting information from a flow based on version-specific traits +requires that middleboxes retain state for every connection ID they see.¶
+The Version Negotiation packet described in this document is not +integrity-protected; it only has modest protection against insertion by off-path +attackers. QUIC versions MUST define a mechanism that authenticates the values +it contains.¶
+This document makes no request of IANA.¶
+There are several traits of QUIC version 1 [QUIC-TRANSPORT] that are not +protected from observation, but are nonetheless considered to be changeable when +a new version is deployed.¶
+This section lists a sampling of incorrect assumptions that might be made based +on knowledge of QUIC version 1. Some of these statements are not even true for +QUIC version 1. This is not an exhaustive list, it is intended to be +illustrative only.¶
+The following statements are NOT guaranteed to be true for every QUIC version:¶
| QUIC | -C. Krasic | -||||||||||||
| Internet-Draft | -Netflix | -
| Intended status: Standards Track | -M. Bishop | -
| Expires: May 15, 2020 | -Akamai Technologies | -
| - | A. Frindell, Ed. | -
| - | |
| - | November 12, 2019 | -
QPACK: Header Compression for HTTP/3
- draft-ietf-quic-qpack-latest
This specification defines QPACK, a compression format for efficiently representing HTTP header fields, to be used in HTTP/3. This is a variation of HPACK header compression that seeks to reduce head-of-line blocking.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-qpack.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
- - - -This specification defines QPACK, a compression format for efficiently +representing HTTP header fields, to be used in HTTP/3. This is a variation of +HPACK header compression that seeks to reduce head-of-line blocking.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +https://mailarchive.ietf.org/arch/search/?email_list=quic.¶
+Working Group information can be found at https://github.com/quicwg; source +code and issues list for this draft can be found at +https://github.com/quicwg/base-drafts/labels/-qpack.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+The QUIC transport protocol [QUIC-TRANSPORT] is designed to support HTTP +semantics, and its design subsumes many of the features of HTTP/2 [RFC7540]. +HTTP/2 uses HPACK [RFC7541] for header compression. If HPACK were used for +HTTP/3 [HTTP3], it would induce head-of-line blocking due to built-in +assumptions of a total ordering across frames on all streams.¶
+QPACK reuses core concepts from HPACK, but is redesigned to allow correctness in +the presence of out-of-order delivery, with flexibility for implementations to +balance between resilience against head-of-line blocking and optimal compression +ratio. The design goals are to closely approach the compression ratio of HPACK +with substantially less head-of-line blocking under the same loss conditions.¶
+The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+Definitions of terms that are used in this document:¶
+QPACK is a name, not an acronym.
-Diagrams use the format described in Section 3.1 of [RFC2360], with the following additional conventions:
- - -QPACK is a name, not an acronym.¶
+Diagrams use the format described in Section 3.1 of [RFC2360], with the +following additional conventions:¶
+Like HPACK, QPACK uses two tables for associating header fields to indices. The static table (see Section 3.1) is predefined and contains common header fields (some of them with an empty value). The dynamic table (see Section 3.2) is built up over the course of the connection and can be used by the encoder to index header fields in the encoded header lists.
-QPACK defines unidirectional streams for sending instructions from encoder to decoder and vice versa.
-An encoder converts a header list into a header block by emitting either an indexed or a literal representation for each header field in the list (see Section 4.5). Indexed representations achieve high compression by replacing the literal name and possibly the value with an index to either the static or dynamic table. References to the static table and literal representations do not require any dynamic state and never risk head-of-line blocking. References to the dynamic table risk head-of-line blocking if the encoder has not received an acknowledgement indicating the entry is available at the decoder.
-An encoder MAY insert any entry in the dynamic table it chooses; it is not limited to header fields it is compressing.
-QPACK preserves the ordering of header fields within each header list. An encoder MUST emit header field representations in the order they appear in the input header list.
-QPACK is designed to contain the more complex state tracking to the encoder, while the decoder is relatively simple.
-An encoder MUST ensure that a header block which references a dynamic table entry is not processed by the decoder after the referenced entry has been evicted. Hence the encoder needs to retain information about each compressed header block that references the dynamic table until that header block is acknowledged by the decoder (see Section 4.4.1).
-Inserting entries into the dynamic table might not be possible if the table contains entries which cannot be evicted.
-A dynamic table entry cannot be evicted immediately after insertion, even if it has never been referenced. Once the insertion of a dynamic table entry has been acknowledged and there are no outstanding unacknowledged references to the entry, the entry becomes evictable.
-If the dynamic table does not contain enough room for a new entry without evicting other entries, and the entries which would be evicted are not evictable, the encoder MUST NOT insert that entry into the dynamic table (including duplicates of existing entries). In order to avoid this, an encoder that uses the dynamic table has to keep track of whether each entry is currently evictable or not.
-To ensure that the encoder is not prevented from adding new entries, the encoder can avoid referencing entries that are close to eviction. Rather than reference such an entry, the encoder can emit a Duplicate instruction (see Section 4.3.4), and reference the duplicate instead.
-Determining which entries are too close to eviction to reference is an encoder preference. One heuristic is to target a fixed amount of available space in the dynamic table: either unused space or space that can be reclaimed by evicting non-blocking entries. To achieve this, the encoder can maintain a draining index, which is the smallest absolute index (see Section 3.2.4) in the dynamic table that it will emit a reference for. As new entries are inserted, the encoder increases the draining index to maintain the section of the table that it will not reference. If the encoder does not create new references to entries with an absolute index lower than the draining index, the number of unacknowledged references to those entries will eventually become zero, allowing them to be evicted.
- - +Like HPACK, QPACK uses two tables for associating header fields to indices. The +static table (see Section 3.1) is predefined and contains common +header fields (some of them with an empty value). The dynamic table (see +Section 3.2) is built up over the course of the connection and can +be used by the encoder to index header fields in the encoded header lists.¶
+QPACK defines unidirectional streams for sending instructions from encoder to +decoder and vice versa.¶
+An encoder converts a header list into a header block by emitting either an +indexed or a literal representation for each header field in the list (see +Section 4.5). Indexed representations achieve high +compression by replacing the literal name and possibly the value with an index +to either the static or dynamic table. References to the static table and +literal representations do not require any dynamic state and never risk +head-of-line blocking. References to the dynamic table risk head-of-line +blocking if the encoder has not received an acknowledgement indicating the entry +is available at the decoder.¶
+An encoder MAY insert any entry in the dynamic table it chooses; it is not +limited to header fields it is compressing.¶
+QPACK preserves the ordering of header fields within each header list. An +encoder MUST emit header field representations in the order they appear in the +input header list.¶
+QPACK is designed to contain the more complex state tracking to the encoder, +while the decoder is relatively simple.¶
+An encoder MUST ensure that a header block which references a dynamic table +entry is not processed by the decoder after the referenced entry has been +evicted. Hence the encoder needs to retain information about each compressed +header block that references the dynamic table until that header block is +acknowledged by the decoder (see Section 4.4.1).¶
+Inserting entries into the dynamic table might not be possible if the table +contains entries which cannot be evicted.¶
+A dynamic table entry cannot be evicted immediately after insertion, even if it +has never been referenced. Once the insertion of a dynamic table entry has been +acknowledged and there are no outstanding unacknowledged references to the +entry, the entry becomes evictable.¶
+If the dynamic table does not contain enough room for a new entry without +evicting other entries, and the entries which would be evicted are not +evictable, the encoder MUST NOT insert that entry into the dynamic table +(including duplicates of existing entries). In order to avoid this, an encoder +that uses the dynamic table has to keep track of whether each entry is currently +evictable or not.¶
+To ensure that the encoder is not prevented from adding new entries, the encoder +can avoid referencing entries that are close to eviction. Rather than +reference such an entry, the encoder can emit a Duplicate instruction (see +Section 4.3.4), and reference the duplicate instead.¶
+Determining which entries are too close to eviction to reference is an encoder +preference. One heuristic is to target a fixed amount of available space in the +dynamic table: either unused space or space that can be reclaimed by evicting +non-blocking entries. To achieve this, the encoder can maintain a draining +index, which is the smallest absolute index (see Section 3.2.4) in the dynamic +table that it will emit a reference for. As new entries are inserted, the +encoder increases the draining index to maintain the section of the table that +it will not reference. If the encoder does not create new references to entries +with an absolute index lower than the draining index, the number of +unacknowledged references to those entries will eventually become zero, allowing +them to be evicted.¶
+
+----------+---------------------------------+--------+
| Draining | Referenceable | Unused |
@@ -650,100 +1350,318 @@
Dropping Draining Index Insertion Point
Point
-Figure 1: Draining Dynamic Table Entries
-Because QUIC does not guarantee order between data on different streams, a decoder might encounter a header block that references a dynamic table entry that it has not yet received.
-Each header block contains a Required Insert Count (see Section 4.5.1), the lowest possible value for the Insert Count with which the header block can be decoded. For a header block with references to the dynamic table, the Required Insert Count is one larger than the largest absolute index of all referenced dynamic table entries. For a header block with no references to the dynamic table, the Required Insert Count is zero.
-When the decoder receives a header block with a Required Insert Count greater than its own Insert Count, the stream cannot be processed immediately, and is considered “blocked” (see Section 2.2.1).
-The decoder specifies an upper bound on the number of streams which can be blocked using the SETTINGS_QPACK_BLOCKED_STREAMS setting (see Section 5). An encoder MUST limit the number of streams which could become blocked to the value of SETTINGS_QPACK_BLOCKED_STREAMS at all times. If a decoder encounters more blocked streams than it promised to support, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-Note that the decoder might not become blocked on every stream which risks becoming blocked.
-An encoder can decide whether to risk having a stream become blocked. If permitted by the value of SETTINGS_QPACK_BLOCKED_STREAMS, compression efficiency can often be improved by referencing dynamic table entries that are still in transit, but if there is loss or reordering the stream can become blocked at the decoder. An encoder can avoid the risk of blocking by only referencing dynamic table entries which have been acknowledged, but this could mean using literals. Since literals make the header block larger, this can result in the encoder becoming blocked on congestion or flow control limits.
-The Known Received Count is the total number of dynamic table insertions and duplications acknowledged by the decoder. The encoder tracks the Known Received Count in order to identify which dynamic table entries can be referenced without potentially blocking a stream. The decoder tracks the Known Received Count in order to be able to send Insert Count Increment instructions.
-A Header Acknowledgement instruction (Section 4.4.1) implies that the decoder has received all dynamic table state necessary to process corresponding the header block. If the Required Insert Count of the acknowledged header block is greater than the current Known Received Count, Known Received Count is updated to the value of the Required Insert Count.
-An Insert Count Increment instruction Section 4.4.3 increases the Known Received Count by its Increment parameter. See Section 2.2.2.3 for guidance.
-As in HPACK, the decoder processes header blocks and emits the corresponding header lists. It also processes instructions received on the encoder stream that modify the dynamic table. Note that header blocks and encoder stream instructions arrive on separate streams. This is unlike HPACK, where header blocks can contain instructions that modify the dynamic table, and there is no dedicated stream of HPACK instructions.
-The decoder MUST emit header fields in the order their representations appear in the input header block.
-Upon receipt of a header block, the decoder examines the Required Insert Count. When the Required Insert Count is less than or equal to the decoder’s Insert Count, the header block can be processed immediately. Otherwise, the stream on which the header block was received becomes blocked.
-While blocked, header block data SHOULD remain in the blocked stream’s flow control window. A stream becomes unblocked when the Insert Count becomes greater than or equal to the Required Insert Count for all header blocks the decoder has started reading from the stream.
-When processing header blocks, the decoder expects the Required Insert Count to exactly match the value defined in Section 2.1.3. If it encounters a smaller value than expected, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED (see Section 2.2.3). If it encounters a larger value than expected, it MAY treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-The decoder signals the following events by emitting decoder instructions (Section 4.4) on the decoder stream.
-After the decoder finishes decoding a header block containing dynamic table references, it MUST emit a Header Acknowledgement instruction (Section 4.4.1). A stream may carry multiple header blocks in the case of intermediate responses, trailers, and pushed requests. The encoder interprets each Header Acknowledgement instruction as acknowledging the earliest unacknowledged header block containing dynamic table references sent on the given stream.
-When an endpoint receives a stream reset before the end of a stream or before all header blocks are processed on that stream, or when it abandons reading of a stream, it generates a Stream Cancellation instruction (see Section 4.4.2). This signals to the encoder that all references to the dynamic table on that stream are no longer outstanding. A decoder with a maximum dynamic table capacity equal to zero (see Section 3.2.3) MAY omit sending Stream Cancellations, because the encoder cannot have any dynamic table references. An encoder cannot infer from this instruction that any updates to the dynamic table have been received.
-The Header Acknowledgement and Stream Cancellation instructions permit the encoder to remove references to entries in the dynamic table. When an entry with absolute index lower than the Known Received Count has zero references, then it is considered evictable (see Section 2.1.2).
-After receiving new table entries on the encoder stream, the decoder chooses when to emit Insert Count Increment instructions (see Section 4.4.3). Emitting this instruction after adding each new dynamic table entry will provide the timeliest feedback to the encoder, but could be redundant with other decoder feedback. By delaying an Insert Count Increment instruction, the decoder might be able to coalesce multiple Insert Count Increment instructions, or replace them entirely with Header Acknowledgements (see Section 4.4.1). However, delaying too long may lead to compression inefficiencies if the encoder waits for an entry to be acknowledged before using it.
-If the decoder encounters a reference in a header block representation to a dynamic table entry which has already been evicted or which has an absolute index greater than or equal to the declared Required Insert Count (see Section 4.5.1), it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-If the decoder encounters a reference in an encoder instruction to a dynamic table entry which has already been evicted, it MUST treat this as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-Unlike in HPACK, entries in the QPACK static and dynamic tables are addressed separately. The following sections describe how entries in each table are addressed.
-The static table consists of a predefined static list of header fields, each of which has a fixed index over time. Its entries are defined in Appendix A.
-All entries in the static table have a name and a value. However, values can be empty (that is, have a length of 0). Each entry is identified by a unique index.
-Note that the QPACK static table is indexed from 0, whereas the HPACK static table is indexed from 1.
-When the decoder encounters an invalid static table index in a header block representation it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED. If this index is received on the encoder stream, this MUST be treated as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-The dynamic table consists of a list of header fields maintained in first-in, first-out order. Each HTTP/3 endpoint holds a dynamic table that is initially empty. Entries are added by encoder instructions received on the encoder stream (see Section 4.3).
-The dynamic table can contain duplicate entries (i.e., entries with the same name and same value). Therefore, duplicate entries MUST NOT be treated as an error by the decoder.
-Dynamic table entries can have empty values.
-The size of the dynamic table is the sum of the size of its entries.
-The size of an entry is the sum of its name’s length in bytes, its value’s length in bytes, and 32. The size of an entry is calculated using the length of its name and value without Huffman encoding applied.
-The encoder sets the capacity of the dynamic table, which serves as the upper limit on its size. The initial capacity of the dynamic table is zero. The encoder sends a Set Dynamic Table Capacity instruction (Section 4.3.1) with a non-zero capacity to begin using the dynamic table.
-Before a new entry is added to the dynamic table, entries are evicted from the end of the dynamic table until the size of the dynamic table is less than or equal to (table capacity - size of new entry). The encoder MUST NOT cause a dynamic table entry to be evicted unless that entry is evictable (see Section 2.1.2). The new entry is then added to the table. It is an error if the encoder attempts to add an entry that is larger than the dynamic table capacity; the decoder MUST treat this as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-A new entry can reference an entry in the dynamic table that will be evicted when adding this new entry into the dynamic table. Implementations are cautioned to avoid deleting the referenced name or value if the referenced entry is evicted from the dynamic table prior to inserting the new entry.
-Whenever the dynamic table capacity is reduced by the encoder (see Section 4.3.1), entries are evicted from the end of the dynamic table until the size of the dynamic table is less than or equal to the new table capacity. This mechanism can be used to completely clear entries from the dynamic table by setting a capacity of 0, which can subsequently be restored.
-To bound the memory requirements of the decoder, the decoder limits the maximum value the encoder is permitted to set for the dynamic table capacity. In HTTP/3, this limit is determined by the value of SETTINGS_QPACK_MAX_TABLE_CAPACITY sent by the decoder (see Section 5). The encoder MUST not set a dynamic table capacity that exceeds this maximum, but it can choose to use a lower dynamic table capacity (see Section 4.3.1).
-For clients using 0-RTT data in HTTP/3, the server’s maximum table capacity is the remembered value of the setting, or zero if the value was not previously sent. When the client’s 0-RTT value of the SETTING is zero, the server MAY set it to a non-zero value in its SETTINGS frame. If the remembered value is non-zero, the server MUST send the same non-zero value in its SETTINGS frame. If it specifies any other value, or omits SETTINGS_QPACK_MAX_TABLE_CAPACITY from SETTINGS, the encoder must treat this as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-For HTTP/3 servers and HTTP/3 clients when 0-RTT is not attempted or is rejected, the maximum table capacity is 0 until the encoder processes a SETTINGS frame with a non-zero value of SETTINGS_QPACK_MAX_TABLE_CAPACITY.
-When the maximum table capacity is zero, the encoder MUST NOT insert entries into the dynamic table, and MUST NOT send any encoder instructions on the encoder stream.
-Each entry possesses an absolute index which is fixed for the lifetime of that entry. The first entry inserted has an absolute index of “0”; indices increase by one with each insertion.
-Relative indices begin at zero and increase in the opposite direction from the absolute index. Determining which entry has a relative index of “0” depends on the context of the reference.
-In encoder instructions (see Section 4.3), a relative index of “0” refers to the most recently inserted value in the dynamic table. Note that this means the entry referenced by a given relative index will change while interpreting instructions on the encoder stream.
+Because QUIC does not guarantee order between data on different streams, a +decoder might encounter a header block that references a dynamic table entry +that it has not yet received.¶
+Each header block contains a Required Insert Count (see Section 4.5.1), the +lowest possible value for the Insert Count with which the header block can be +decoded. For a header block with references to the dynamic table, the Required +Insert Count is one larger than the largest absolute index of all referenced +dynamic table entries. For a header block with no references to the dynamic +table, the Required Insert Count is zero.¶
+When the decoder receives a header block with a Required Insert Count greater +than its own Insert Count, the stream cannot be processed immediately, and is +considered "blocked" (see Section 2.2.1).¶
+The decoder specifies an upper bound on the number of streams which can be +blocked using the SETTINGS_QPACK_BLOCKED_STREAMS setting (see +Section 5). An encoder MUST limit the number of streams which could +become blocked to the value of SETTINGS_QPACK_BLOCKED_STREAMS at all times. +If a decoder encounters more blocked streams than it promised to support, it +MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.¶
+Note that the decoder might not become blocked on every stream which risks +becoming blocked.¶
+An encoder can decide whether to risk having a stream become blocked. If +permitted by the value of SETTINGS_QPACK_BLOCKED_STREAMS, compression efficiency +can often be improved by referencing dynamic table entries that are still in +transit, but if there is loss or reordering the stream can become blocked at the +decoder. An encoder can avoid the risk of blocking by only referencing dynamic +table entries which have been acknowledged, but this could mean using +literals. Since literals make the header block larger, this can result in the +encoder becoming blocked on congestion or flow control limits.¶
+The Known Received Count is the total number of dynamic table insertions and +duplications acknowledged by the decoder. The encoder tracks the Known Received +Count in order to identify which dynamic table entries can be referenced without +potentially blocking a stream. The decoder tracks the Known Received Count in +order to be able to send Insert Count Increment instructions.¶
+A Header Acknowledgement instruction (Section 4.4.1) implies that +the decoder has received all dynamic table state necessary to process +corresponding the header block. If the Required Insert Count of the +acknowledged header block is greater than the current Known Received Count, +Known Received Count is updated to the value of the Required Insert Count.¶
+An Insert Count Increment instruction Section 4.4.3 increases the +Known Received Count by its Increment parameter. See Section 2.2.2.3 for +guidance.¶
+As in HPACK, the decoder processes header blocks and emits the corresponding +header lists. It also processes instructions received on the encoder stream that +modify the dynamic table. Note that header blocks and encoder stream +instructions arrive on separate streams. This is unlike HPACK, where header +blocks can contain instructions that modify the dynamic table, and there is no +dedicated stream of HPACK instructions.¶
+The decoder MUST emit header fields in the order their representations appear in +the input header block.¶
+Upon receipt of a header block, the decoder examines the Required Insert Count. +When the Required Insert Count is less than or equal to the decoder's Insert +Count, the header block can be processed immediately. Otherwise, the stream on +which the header block was received becomes blocked.¶
+While blocked, header block data SHOULD remain in the blocked stream's flow +control window. A stream becomes unblocked when the Insert Count becomes +greater than or equal to the Required Insert Count for all header blocks the +decoder has started reading from the stream.¶
+When processing header blocks, the decoder expects the Required Insert Count to +exactly match the value defined in Section 2.1.3. If it encounters a +smaller value than expected, it MUST treat this as a connection error of type +HTTP_QPACK_DECOMPRESSION_FAILED (see Section 2.2.3). If it encounters a +larger value than expected, it MAY treat this as a connection error of type +HTTP_QPACK_DECOMPRESSION_FAILED.¶
+The decoder signals the following events by emitting decoder instructions +(Section 4.4) on the decoder stream.¶
+After the decoder finishes decoding a header block containing dynamic table +references, it MUST emit a Header Acknowledgement instruction +(Section 4.4.1). A stream may carry multiple header blocks in the +case of intermediate responses, trailers, and pushed requests. The encoder +interprets each Header Acknowledgement instruction as acknowledging the earliest +unacknowledged header block containing dynamic table references sent on the +given stream.¶
+When an endpoint receives a stream reset before the end of a stream or before +all header blocks are processed on that stream, or when it abandons reading of a +stream, it generates a Stream Cancellation instruction (see +Section 4.4.2). This signals to the encoder that all references to +the dynamic table on that stream are no longer outstanding. A decoder with a +maximum dynamic table capacity equal to zero (see +Section 3.2.3) MAY omit sending Stream Cancellations, +because the encoder cannot have any dynamic table references. An encoder cannot +infer from this instruction that any updates to the dynamic table have been +received.¶
+The Header Acknowledgement and Stream Cancellation instructions permit the +encoder to remove references to entries in the dynamic table. When an entry +with absolute index lower than the Known Received Count has zero references, +then it is considered evictable (see Section 2.1.2).¶
+After receiving new table entries on the encoder stream, the decoder chooses +when to emit Insert Count Increment instructions (see +Section 4.4.3). Emitting this instruction after adding each new +dynamic table entry will provide the timeliest feedback to the encoder, but +could be redundant with other decoder feedback. By delaying an Insert Count +Increment instruction, the decoder might be able to coalesce multiple Insert +Count Increment instructions, or replace them entirely with Header +Acknowledgements (see Section 4.4.1). However, delaying too long +may lead to compression inefficiencies if the encoder waits for an entry to be +acknowledged before using it.¶
+If the decoder encounters a reference in a header block representation to a
+dynamic table entry which has already been evicted or which has an absolute
+index greater than or equal to the declared Required Insert Count (see
+Section 4.5.1), it MUST treat this as a connection error of type
+HTTP_QPACK_DECOMPRESSION_FAILED.¶
If the decoder encounters a reference in an encoder instruction to a dynamic
+table entry which has already been evicted, it MUST treat this as a connection
+error of type HTTP_QPACK_ENCODER_STREAM_ERROR.¶
Unlike in HPACK, entries in the QPACK static and dynamic tables are addressed +separately. The following sections describe how entries in each table are +addressed.¶
+The static table consists of a predefined static list of header fields, each of +which has a fixed index over time. Its entries are defined in Appendix A.¶
+All entries in the static table have a name and a value. However, values can be +empty (that is, have a length of 0). Each entry is identified by a unique +index.¶
+Note that the QPACK static table is indexed from 0, whereas the HPACK static +table is indexed from 1.¶
+When the decoder encounters an invalid static table index in a header block
+representation it MUST treat this as a connection error of type
+HTTP_QPACK_DECOMPRESSION_FAILED. If this index is received on the encoder
+stream, this MUST be treated as a connection error of type
+HTTP_QPACK_ENCODER_STREAM_ERROR.¶
The dynamic table consists of a list of header fields maintained in first-in, +first-out order. Each HTTP/3 endpoint holds a dynamic table that is initially +empty. Entries are added by encoder instructions received on the encoder stream +(see Section 4.3).¶
+The dynamic table can contain duplicate entries (i.e., entries with the same +name and same value). Therefore, duplicate entries MUST NOT be treated as an +error by the decoder.¶
+Dynamic table entries can have empty values.¶
+The size of the dynamic table is the sum of the size of its entries.¶
+The size of an entry is the sum of its name's length in bytes, its value's +length in bytes, and 32. The size of an entry is calculated using the length of +its name and value without Huffman encoding applied.¶
+The encoder sets the capacity of the dynamic table, which serves as the upper +limit on its size. The initial capacity of the dynamic table is zero. The +encoder sends a Set Dynamic Table Capacity instruction +(Section 4.3.1) with a non-zero capacity to begin using the dynamic +table.¶
+Before a new entry is added to the dynamic table, entries are evicted from the
+end of the dynamic table until the size of the dynamic table is less than or
+equal to (table capacity - size of new entry). The encoder MUST NOT cause a
+dynamic table entry to be evicted unless that entry is evictable (see
+Section 2.1.2). The new entry is then added to the table. It is an
+error if the encoder attempts to add an entry that is larger than the dynamic
+table capacity; the decoder MUST treat this as a connection error of type
+HTTP_QPACK_ENCODER_STREAM_ERROR.¶
A new entry can reference an entry in the dynamic table that will be evicted +when adding this new entry into the dynamic table. Implementations are +cautioned to avoid deleting the referenced name or value if the referenced entry +is evicted from the dynamic table prior to inserting the new entry.¶
+Whenever the dynamic table capacity is reduced by the encoder (see +Section 4.3.1), entries are evicted from the end of the dynamic table +until the size of the dynamic table is less than or equal to the new table +capacity. This mechanism can be used to completely clear entries from the +dynamic table by setting a capacity of 0, which can subsequently be restored.¶
+To bound the memory requirements of the decoder, the decoder limits the maximum +value the encoder is permitted to set for the dynamic table capacity. In +HTTP/3, this limit is determined by the value of +SETTINGS_QPACK_MAX_TABLE_CAPACITY sent by the decoder (see Section 5). +The encoder MUST not set a dynamic table capacity that exceeds this maximum, but +it can choose to use a lower dynamic table capacity (see +Section 4.3.1).¶
+For clients using 0-RTT data in HTTP/3, the server's maximum table capacity is
+the remembered value of the setting, or zero if the value was not previously
+sent. When the client's 0-RTT value of the SETTING is zero, the server MAY set
+it to a non-zero value in its SETTINGS frame. If the remembered value is
+non-zero, the server MUST send the same non-zero value in its SETTINGS frame.
+If it specifies any other value, or omits SETTINGS_QPACK_MAX_TABLE_CAPACITY from
+SETTINGS, the encoder must treat this as a connection error of type
+HTTP_QPACK_DECODER_STREAM_ERROR.¶
For HTTP/3 servers and HTTP/3 clients when 0-RTT is not attempted or is +rejected, the maximum table capacity is 0 until the encoder processes a SETTINGS +frame with a non-zero value of SETTINGS_QPACK_MAX_TABLE_CAPACITY.¶
+When the maximum table capacity is zero, the encoder MUST NOT insert entries +into the dynamic table, and MUST NOT send any encoder instructions on the +encoder stream.¶
+Each entry possesses an absolute index which is fixed for the lifetime of that +entry. The first entry inserted has an absolute index of "0"; indices increase +by one with each insertion.¶
+Relative indices begin at zero and increase in the opposite direction from the +absolute index. Determining which entry has a relative index of "0" depends on +the context of the reference.¶
+In encoder instructions (see Section 4.3), a relative index of "0" +refers to the most recently inserted value in the dynamic table. Note that this +means the entry referenced by a given relative index will change while +interpreting instructions on the encoder stream.¶
+
+-----+---------------+-------+
| n-1 | ... | d | Absolute Index
@@ -757,9 +1675,18 @@
n = count of entries inserted
d = count of entries dropped
-Example Dynamic Table Indexing - Encoder Stream
-Unlike in encoder instructions, relative indices in header block representations are relative to the Base at the beginning of the header block (see Section 4.5.1). This ensures that references are stable even if header blocks and dynamic table updates are processed out of order.
-In a header block a relative index of “0” refers to the entry with absolute index equal to Base - 1.
+Unlike in encoder instructions, relative indices in header block representations +are relative to the Base at the beginning of the header block (see +Section 4.5.1). This ensures that references are stable even if header +blocks and dynamic table updates are processed out of order.¶
+In a header block a relative index of "0" refers to the entry with absolute +index equal to Base - 1.¶
+
Base
|
@@ -774,12 +1701,26 @@
d = count of entries dropped
In this example, Base = n - 2
-Example Dynamic Table Indexing - Relative Index in Header Block
-Post-Base indices are used in header block instructions for entries with absolute indices greater than or equal to Base, starting at 0 for the entry with absolute index equal to Base, and increasing in the same direction as the absolute index.
-Post-Base indices allow an encoder to process a header block in a single pass and include references to entries added while processing this (or other) header blocks.
+Post-Base indices are used in header block instructions for entries with +absolute indices greater than or equal to Base, starting at 0 for the entry with +absolute index equal to Base, and increasing in the same direction as the +absolute index.¶
+Post-Base indices allow an encoder to process a header block in a single pass +and include references to entries added while processing this (or other) header +blocks.¶
+
Base
|
@@ -794,64 +1735,157 @@
d = count of entries dropped
In this example, Base = n - 2
-Example Dynamic Table Indexing - Post-Base Index in Header Block
-The prefixed integer from Section 5.1 of [RFC7541] is used heavily throughout this document. The format from [RFC7541] is used unmodified. Note, however, that QPACK uses some prefix sizes not actually used in HPACK.
-QPACK implementations MUST be able to decode integers up to and including 62 bits long.
-The string literal defined by Section 5.2 of [RFC7541] is also used throughout. This string format includes optional Huffman encoding.
-HPACK defines string literals to begin on a byte boundary. They begin with a single bit flag, denoted as ‘H’ in this document (indicating whether the string is Huffman-coded), followed by the Length encoded as a 7-bit prefix integer, and finally Length bytes of data. When Huffman encoding is enabled, the Huffman table from Appendix B of [RFC7541] is used without modification.
-This document expands the definition of string literals and permits them to begin other than on a byte boundary. An “N-bit prefix string literal” begins with the same Huffman flag, followed by the length encoded as an (N-1)-bit prefix integer. The prefix size, N, can have a value between 2 and 8 inclusive. The remainder of the string literal is unmodified.
-A string literal without a prefix length noted is an 8-bit prefix string literal and follows the definitions in [RFC7541] without modification.
-QPACK defines two unidirectional stream types:
- - +The prefixed integer from Section 5.1 of [RFC7541] is used heavily throughout +this document. The format from [RFC7541] is used unmodified. Note, however, +that QPACK uses some prefix sizes not actually used in HPACK.¶
+QPACK implementations MUST be able to decode integers up to and including 62 +bits long.¶
+The string literal defined by Section 5.2 of [RFC7541] is also used throughout. +This string format includes optional Huffman encoding.¶
+HPACK defines string literals to begin on a byte boundary. They begin with a +single bit flag, denoted as 'H' in this document (indicating whether the string +is Huffman-coded), followed by the Length encoded as a 7-bit prefix integer, +and finally Length bytes of data. When Huffman encoding is enabled, the Huffman +table from Appendix B of [RFC7541] is used without modification.¶
+This document expands the definition of string literals and permits them to +begin other than on a byte boundary. An "N-bit prefix string literal" begins +with the same Huffman flag, followed by the length encoded as an (N-1)-bit +prefix integer. The prefix size, N, can have a value between 2 and 8 inclusive. +The remainder of the string literal is unmodified.¶
+A string literal without a prefix length noted is an 8-bit prefix string literal +and follows the definitions in [RFC7541] without modification.¶
+QPACK defines two unidirectional stream types:¶
0x02.
+It carries an unframed sequence of encoder instructions from encoder
+to decoder.¶
+0x03.
+It carries an unframed sequence of decoder instructions from decoder
+to encoder.¶
+HTTP/3 endpoints contain a QPACK encoder and decoder. Each endpoint MUST initiate at most one encoder stream and at most one decoder stream. Receipt of a second instance of either stream type MUST be treated as a connection error of type HTTP_STREAM_CREATION_ERROR. These streams MUST NOT be closed. Closure of either unidirectional stream type MUST be treated as a connection error of type HTTP_CLOSED_CRITICAL_STREAM.
-An endpoint MAY avoid creating an encoder stream if it’s not going to be used (for example if its encoder doesn’t wish to use the dynamic table, or if the maximum size of the dynamic table permitted by the peer is zero).
-An endpoint MAY avoid creating a decoder stream if its decoder sets the maximum capacity of the dynamic table to zero.
-An endpoint MUST allow its peer to create an encoder stream and a decoder stream even if the connection’s settings prevent their use.
-An encoder sends encoder instructions on the encoder stream to set the capacity of the dynamic table and add dynamic table entries. Instructions adding table entries can use existing entries to avoid transmitting redundant information. The name can be transmitted as a reference to an existing entry in the static or the dynamic table or as a string literal. For entries which already exist in the dynamic table, the full entry can also be used by reference, creating a duplicate entry.
-This section specifies the following encoder instructions.
-An encoder informs the decoder of a change to the dynamic table capacity using an instruction which begins with the ‘001’ three-bit pattern. This is followed by the new dynamic table capacity represented as an integer with a 5-bit prefix (see Section 4.1.1).
- - +HTTP/3 endpoints contain a QPACK encoder and decoder. Each endpoint MUST +initiate at most one encoder stream and at most one decoder stream. Receipt of a +second instance of either stream type MUST be treated as a connection error of +type HTTP_STREAM_CREATION_ERROR. These streams MUST NOT be closed. Closure of +either unidirectional stream type MUST be treated as a connection error of type +HTTP_CLOSED_CRITICAL_STREAM.¶
+An endpoint MAY avoid creating an encoder stream if it's not going to be used +(for example if its encoder doesn't wish to use the dynamic table, or if the +maximum size of the dynamic table permitted by the peer is zero).¶
+An endpoint MAY avoid creating a decoder stream if its decoder sets the maximum +capacity of the dynamic table to zero.¶
+An endpoint MUST allow its peer to create an encoder stream and a decoder stream +even if the connection's settings prevent their use.¶
+An encoder sends encoder instructions on the encoder stream to set the capacity +of the dynamic table and add dynamic table entries. Instructions adding table +entries can use existing entries to avoid transmitting redundant information. +The name can be transmitted as a reference to an existing entry in the static or +the dynamic table or as a string literal. For entries which already exist in +the dynamic table, the full entry can also be used by reference, creating a +duplicate entry.¶
+This section specifies the following encoder instructions.¶
+An encoder informs the decoder of a change to the dynamic table capacity using +an instruction which begins with the '001' three-bit pattern. This is followed +by the new dynamic table capacity represented as an integer with a 5-bit prefix +(see Section 4.1.1).¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 0 | 0 | 1 | Capacity (5+) | +---+---+---+-------------------+-
Figure 2: Set Dynamic Table Capacity
-The new capacity MUST be lower than or equal to the limit described in Section 3.2.3. In HTTP/3, this limit is the value of the SETTINGS_QPACK_MAX_TABLE_CAPACITY parameter (see Section 5) received from the decoder. The decoder MUST treat a new dynamic table capacity value that exceeds this limit as a connection error of type HTTP_QPACK_ENCODER_STREAM_ERROR.
-Reducing the dynamic table capacity can cause entries to be evicted (see Section 3.2.2). This MUST NOT cause the eviction of entries which are not evictable (see Section 2.1.2). Changing the capacity of the dynamic table is not acknowledged as this instruction does not insert an entry.
-An encoder adds an entry to the dynamic table where the header field name matches the header field name of an entry stored in the static or the dynamic table using an instruction that starts with the ‘1’ one-bit pattern. The second (‘T’) bit indicates whether the reference is to the static or dynamic table. The 6-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header name. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-The header name reference is followed by the header field value represented as a string literal (see Section 4.1.2).
+The new capacity MUST be lower than or equal to the limit described in
+Section 3.2.3. In HTTP/3, this limit is the value of the
+SETTINGS_QPACK_MAX_TABLE_CAPACITY parameter (see Section 5) received
+from the decoder. The decoder MUST treat a new dynamic table capacity value
+that exceeds this limit as a connection error of type
+HTTP_QPACK_ENCODER_STREAM_ERROR.¶
Reducing the dynamic table capacity can cause entries to be evicted (see +Section 3.2.2). This MUST NOT cause the eviction of entries which are not +evictable (see Section 2.1.2). Changing the capacity of the dynamic +table is not acknowledged as this instruction does not insert an entry.¶
+An encoder adds an entry to the dynamic table where the header field name +matches the header field name of an entry stored in the static or the dynamic +table using an instruction that starts with the '1' one-bit pattern. The second +('T') bit indicates whether the reference is to the static or dynamic table. The +6-bit prefix integer (see Section 4.1.1) that follows is used to locate +the table entry for the header name. When T=1, the number represents the static +table index; when T=0, the number is the relative index of the entry in the +dynamic table.¶
+The header name reference is followed by the header field value represented as a +string literal (see Section 4.1.2).¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
@@ -862,12 +1896,25 @@
| Value String (Length bytes) |
+-------------------------------+
-Insert Header Field -- Indexed Name
-An encoder adds an entry to the dynamic table where both the header field name and the header field value are represented as string literals using an instruction that starts with the ‘01’ two-bit pattern.
-This is followed by the name represented as a 6-bit prefix string literal, and the value represented as an 8-bit prefix string literal (see Section 4.1.2).
+An encoder adds an entry to the dynamic table where both the header field name +and the header field value are represented as string literals using an +instruction that starts with the '01' two-bit pattern.¶
+This is followed by the name represented as a 6-bit prefix string literal, and +the value represented as an 8-bit prefix string literal (see +Section 4.1.2).¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
@@ -880,81 +1927,166 @@
| Value String (Length bytes) |
+-------------------------------+
-Insert Header Field -- New Name
-An encoder duplicates an existing entry in the dynamic table using an instruction that begins with the ‘000’ three-bit pattern. This is followed by the relative index of the existing entry represented as an integer with a 5-bit prefix (see Section 4.1.1.
- - +An encoder duplicates an existing entry in the dynamic table using an +instruction that begins with the '000' three-bit pattern. This is followed by +the relative index of the existing entry represented as an integer with a 5-bit +prefix (see Section 4.1.1.¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
| 0 | 0 | 0 | Index (5+) |
+---+---+---+-------------------+
-Figure 3: Duplicate
-The existing entry is re-inserted into the dynamic table without resending either the name or the value. This is useful to avoid adding a reference to an older entry, which might block inserting new entries.
-A decoder sends decoder instructions on the decoder stream to inform the encoder about the processing of header blocks and table updates to ensure consistency of the dynamic table.
-This section specifies the following decoder instructions.
-After processing a header block whose declared Required Insert Count is not zero, the decoder emits a Header Acknowledgement instruction. The instruction begins with the ‘1’ one-bit pattern which is followed by the header block’s associated stream ID encoded as a 7-bit prefix integer (see Section 4.1.1).
-This instruction is used as described in Section 2.1.4 and in Section 2.2.2.
- - +The existing entry is re-inserted into the dynamic table without resending +either the name or the value. This is useful to avoid adding a reference to an +older entry, which might block inserting new entries.¶
+A decoder sends decoder instructions on the decoder stream to inform the encoder +about the processing of header blocks and table updates to ensure consistency of +the dynamic table.¶
+This section specifies the following decoder instructions.¶
+After processing a header block whose declared Required Insert Count is not +zero, the decoder emits a Header Acknowledgement instruction. The instruction +begins with the '1' one-bit pattern which is followed by the header block's +associated stream ID encoded as a 7-bit prefix integer (see +Section 4.1.1).¶
+This instruction is used as described in Section 2.1.4 and +in Section 2.2.2.¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 1 | Stream ID (7+) | +---+---------------------------+-
Figure 4: Header Acknowledgement
-If an encoder receives a Header Acknowledgement instruction referring to a stream on which every header block with a non-zero Required Insert Count has already been acknowledged, that MUST be treated as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-The Header Acknowledgement instruction might increase the Known Received Count, see Section 2.1.4.
-When a stream is reset or reading is abandoned, the decoder emits a Stream Cancellation instruction. The instruction begins with the ‘01’ two-bit pattern, which is followed by the stream ID of the affected stream encoded as a 6-bit prefix integer.
-This instruction is used as described in Section 2.2.2.
- - +If an encoder receives a Header Acknowledgement instruction referring to a
+stream on which every header block with a non-zero Required Insert Count has
+already been acknowledged, that MUST be treated as a connection error of type
+HTTP_QPACK_DECODER_STREAM_ERROR.¶
The Header Acknowledgement instruction might increase the Known Received Count; +see Section 2.1.4.¶
+When a stream is reset or reading is abandoned, the decoder emits a Stream +Cancellation instruction. The instruction begins with the '01' two-bit +pattern, which is followed by the stream ID of the affected stream encoded as a +6-bit prefix integer.¶
+This instruction is used as described in Section 2.2.2.¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 0 | 1 | Stream ID (6+) | +---+---+-----------------------+-
Figure 5: Stream Cancellation
-The Insert Count Increment instruction begins with the ‘00’ two-bit pattern, followed by the Increment encoded as a 6-bit prefix integer. This instruction increases the Known Received Count (see Section 2.1.4) by the value of the Increment parameter. The decoder should send an Increment value that increases the Known Received Count to the total number of dynamic table insertions and duplications processed so far.
- - +The Insert Count Increment instruction begins with the '00' two-bit pattern, +followed by the Increment encoded as a 6-bit prefix integer. This instruction +increases the Known Received Count (see Section 2.1.4) by the value +of the Increment parameter. The decoder should send an Increment value that +increases the Known Received Count to the total number of dynamic table +insertions and duplications processed so far.¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 0 | 0 | Increment (6+) | +---+---+-----------------------+-
Figure 6: Insert Count Increment
-An encoder that receives an Increment field equal to zero, or one that increases the Known Received Count beyond what the encoder has sent MUST treat this as a connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.
-A header block consists of a prefix and a possibly empty sequence of representations defined in this section. Each representation corresponds to a single header field. These representations reference the static table or the dynamic table in a particular state, but do not modify that state.
-Header blocks are carried in frames on streams defined by the enclosing protocol.
-Each header block is prefixed with two integers. The Required Insert Count is encoded as an integer with an 8-bit prefix after the encoding described in Section 4.5.1.1). The Base is encoded as a sign bit (‘S’) and a Delta Base value with a 7-bit prefix (see Section 4.5.1.2).
- - +An encoder that receives an Increment field equal to zero, or one that increases
+the Known Received Count beyond what the encoder has sent MUST treat this as a
+connection error of type HTTP_QPACK_DECODER_STREAM_ERROR.¶
A header block consists of a prefix and a possibly empty sequence of +representations defined in this section. Each representation corresponds to a +single header field. These representations reference the static table or the +dynamic table in a particular state, but do not modify that state.¶
+Header blocks are carried in frames on streams defined by the enclosing +protocol.¶
+Each header block is prefixed with two integers. The Required Insert Count is +encoded as an integer with an 8-bit prefix after the encoding described in +Section 4.5.1.1). The Base is encoded as a sign bit ('S') and a Delta Base value with a +7-bit prefix (see Section 4.5.1.2).¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ @@ -965,26 +2097,46 @@-| Compressed Headers ... +-------------------------------+
Figure 7: Header Block
-Required Insert Count identifies the state of the dynamic table needed to process the header block. Blocking decoders use the Required Insert Count to determine when it is safe to process the rest of the block.
-The encoder transforms the Required Insert Count as follows before encoding:
+Required Insert Count identifies the state of the dynamic table needed to +process the header block. Blocking decoders use the Required Insert Count to +determine when it is safe to process the rest of the block.¶
+The encoder transforms the Required Insert Count as follows before encoding:¶
+
if ReqInsertCount == 0:
EncInsertCount = 0
else:
EncInsertCount = (ReqInsertCount mod (2 * MaxEntries)) + 1
-
-Here MaxEntries is the maximum number of entries that the dynamic table can have. The smallest entry has empty name and value strings and has the size of 32. Hence MaxEntries is calculated as
+¶ +Here MaxEntries is the maximum number of entries that the dynamic table can
+have. The smallest entry has empty name and value strings and has the size of
+32. Hence MaxEntries is calculated as¶
MaxEntries = floor( MaxTableCapacity / 32 )
-
-MaxTableCapacity is the maximum capacity of the dynamic table as specified by the decoder (see Section 3.2.3).
-This encoding limits the length of the prefix on long-lived connections.
-The decoder can reconstruct the Required Insert Count using an algorithm such as the following. If the decoder encounters a value of EncodedInsertCount that could not have been produced by a conformant encoder, it MUST treat this as a connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.
-TotalNumberOfInserts is the total number of inserts into the decoder’s dynamic table.
+¶ +MaxTableCapacity is the maximum capacity of the dynamic table as specified by
+the decoder (see Section 3.2.3).¶
This encoding limits the length of the prefix on long-lived connections.¶
+The decoder can reconstruct the Required Insert Count using an algorithm such as
+the following. If the decoder encounters a value of EncodedInsertCount that
+could not have been produced by a conformant encoder, it MUST treat this as a
+connection error of type HTTP_QPACK_DECOMPRESSION_FAILED.¶
TotalNumberOfInserts is the total number of inserts into the decoder's dynamic +table.¶
+
FullRange = 2 * MaxEntries
if EncodedInsertCount == 0:
@@ -1009,51 +2161,119 @@
# Value of 0 must be encoded as 0.
if ReqInsertCount == 0:
Error
-
-For example, if the dynamic table is 100 bytes, then the Required Insert Count will be encoded modulo 6. If a decoder has received 10 inserts, then an encoded value of 3 indicates that the Required Insert Count is 9 for the header block.
-The Base is used to resolve references in the dynamic table as described in Section 3.2.5.
-To save space, the Base is encoded relative to the Required Insert Count using a one-bit sign (‘S’) and the Delta Base value. A sign bit of 0 indicates that the Base is greater than or equal to the value of the Required Insert Count; the decoder adds the value of Delta Base to the Required Insert Count to determine the value of the Base. A sign bit of 1 indicates that the Base is less than the Required Insert Count; the decoder subtracts the value of Delta Base from the Required Insert Count and also subtracts one to determine the value of the Base. That is:
+¶ +For example, if the dynamic table is 100 bytes, then the Required Insert Count +will be encoded modulo 6. If a decoder has received 10 inserts, then an encoded +value of 3 indicates that the Required Insert Count is 9 for the header block.¶
+The Base is used to resolve references in the dynamic table as described in
+Section 3.2.5.¶
To save space, the Base is encoded relative to the Required Insert Count using a
+one-bit sign ('S') and the Delta Base value. A sign bit of 0 indicates that
+the Base is greater than or equal to the value of the Required Insert Count; the
+decoder adds the value of Delta Base to the Required Insert Count to determine
+the value of the Base. A sign bit of 1 indicates that the Base is less than the
+Required Insert Count; the decoder subtracts the value of Delta Base from the
+Required Insert Count and also subtracts one to determine the value of the Base.
+That is:¶
if S == 0:
Base = ReqInsertCount + DeltaBase
else:
Base = ReqInsertCount - DeltaBase - 1
-
-A single-pass encoder determines the Base before encoding a header block. If the encoder inserted entries in the dynamic table while encoding the header block, Required Insert Count will be greater than the Base, so the encoded difference is negative and the sign bit is set to 1. If the header block did not reference the most recent entry in the table and did not insert any new entries, the Base will be greater than the Required Insert Count, so the delta will be positive and the sign bit is set to 0.
-An encoder that produces table updates before encoding a header block might set Base to the value of Required Insert Count. In such case, both the sign bit and the Delta Base will be set to zero.
-A header block that does not reference the dynamic table can use any value for the Base; setting Delta Base to zero is one of the most efficient encodings.
-For example, with a Required Insert Count of 9, a decoder receives an S bit of 1 and a Delta Base of 2. This sets the Base to 6 and enables post-base indexing for three entries. In this example, a relative index of 1 refers to the 5th entry that was added to the table; a post-base index of 1 refers to the 8th entry.
-An indexed header field representation identifies an entry in the static table, or an entry in the dynamic table with an absolute index less than the Base.
+¶ +A single-pass encoder determines the Base before encoding a header block. If +the encoder inserted entries in the dynamic table while encoding the header +block, Required Insert Count will be greater than the Base, so the encoded +difference is negative and the sign bit is set to 1. If the header block did +not reference the most recent entry in the table and did not insert any new +entries, the Base will be greater than the Required Insert Count, so the delta +will be positive and the sign bit is set to 0.¶
+An encoder that produces table updates before encoding a header block might set +Base to the value of Required Insert Count. In such case, both the sign bit and +the Delta Base will be set to zero.¶
+A header block that does not reference the dynamic table can use any value for +the Base; setting Delta Base to zero is one of the most efficient encodings.¶
+For example, with a Required Insert Count of 9, a decoder receives an S bit of 1 +and a Delta Base of 2. This sets the Base to 6 and enables post-base indexing +for three entries. In this example, a relative index of 1 refers to the 5th +entry that was added to the table; a post-base index of 1 refers to the 8th +entry.¶
+An indexed header field representation identifies an entry in the static table, +or an entry in the dynamic table with an absolute index less than the Base.¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 1 | T | Index (6+) | +---+---+-----------------------+-
Indexed Header Field
-This representation starts with the ‘1’ 1-bit pattern, followed by the ‘T’ bit indicating whether the reference is into the static or dynamic table. The 6-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header field. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-An indexed header field with post-base index representation identifies an entry in the dynamic table with an absolute index greater than or equal to the Base.
+This representation starts with the '1' 1-bit pattern, followed by the 'T' bit +indicating whether the reference is into the static or dynamic table. The 6-bit +prefix integer (see Section 4.1.1) that follows is used to locate the +table entry for the header field. When T=1, the number represents the static +table index; when T=0, the number is the relative index of the entry in the +dynamic table.¶
+An indexed header field with post-base index representation identifies an entry +in the dynamic table with an absolute index greater than or equal to the Base.¶
+0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 0 | 0 | 0 | 1 | Index (4+) | +---+---+---+---+---------------+-
Indexed Header Field with Post-Base Index
-This representation starts with the ‘0001’ 4-bit pattern. This is followed by the post-base index (see Section 3.2.6) of the matching header field, represented as an integer with a 4-bit prefix (see Section 4.1.1).
-A literal header field with name reference representation encodes a header field where the header field name matches the header field name of an entry in the static table, or the header field name of an entry in the dynamic table with an absolute index less than the Base.
+This representation starts with the '0001' 4-bit pattern. This is followed by +the post-base index (see Section 3.2.6) of the matching header field, +represented as an integer with a 4-bit prefix (see Section 4.1.1).¶
+A literal header field with name reference representation encodes a header field +where the header field name matches the header field name of an entry in the +static table, or the header field name of an entry in the dynamic table with an +absolute index less than the Base.¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
@@ -1064,14 +2284,39 @@
| Value String (Length bytes) |
+-------------------------------+
-Literal Header Field With Name Reference
-This representation starts with the ‘01’ two-bit pattern. The following bit, ‘N’, indicates whether an intermediary is permitted to add this header to the dynamic header table on subsequent hops. When the ‘N’ bit is set, the encoded header MUST always be encoded with a literal representation. In particular, when a peer sends a header field that it received represented as a literal header field with the ‘N’ bit set, it MUST use a literal representation to forward this header field. This bit is intended for protecting header field values that are not to be put at risk by compressing them (see Section 7 for more details).
-The fourth (‘T’) bit indicates whether the reference is to the static or dynamic table. The 4-bit prefix integer (see Section 4.1.1) that follows is used to locate the table entry for the header name. When T=1, the number represents the static table index; when T=0, the number is the relative index of the entry in the dynamic table.
-Only the header field name is taken from the dynamic table entry; the header field value is encoded as an 8-bit prefix string literal (see Section 4.1.2).
-A literal header field with post-base name reference representation encodes a header field where the header field name matches the header field name of a dynamic table entry with an absolute index greater than or equal to the Base.
+This representation starts with the '01' two-bit pattern. The following bit, +'N', indicates whether an intermediary is permitted to add this header to the +dynamic header table on subsequent hops. When the 'N' bit is set, the encoded +header MUST always be encoded with a literal representation. In particular, when +a peer sends a header field that it received represented as a literal header +field with the 'N' bit set, it MUST use a literal representation to forward this +header field. This bit is intended for protecting header field values that are +not to be put at risk by compressing them (see Section 7 for +more details).¶
+The fourth ('T') bit indicates whether the reference is to the static or dynamic +table. The 4-bit prefix integer (see Section 4.1.1) that follows is +used to locate the table entry for the header name. When T=1, the number +represents the static table index; when T=0, the number is the relative index of +the entry in the dynamic table.¶
+Only the header field name is taken from the dynamic table entry; the header +field value is encoded as an 8-bit prefix string literal (see +Section 4.1.2).¶
+A literal header field with post-base name reference representation encodes a +header field where the header field name matches the header field name of a +dynamic table entry with an absolute index greater than or equal to the Base.¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
@@ -1082,13 +2327,28 @@
| Value String (Length bytes) |
+-------------------------------+
-Literal Header Field With Post-Base Name Reference
-This representation starts with the ‘0000’ four-bit pattern. The fifth bit is the ‘N’ bit as described in Section 4.5.4. This is followed by a post-base index of the dynamic table entry (see Section 3.2.6) encoded as an integer with a 3-bit prefix (see Section 4.1.1).
-Only the header field name is taken from the dynamic table entry; the header field value is encoded as an 8-bit prefix string literal (see Section 4.1.2).
-The literal header field without name reference representation encodes a header field name and a header field value as string literals.
+This representation starts with the '0000' four-bit pattern. The fifth bit is +the 'N' bit as described in Section 4.5.4. This is followed by a +post-base index of the dynamic table entry (see Section 3.2.6) encoded as an +integer with a 3-bit prefix (see Section 4.1.1).¶
+Only the header field name is taken from the dynamic table entry; the header +field value is encoded as an 8-bit prefix string literal (see +Section 4.1.2).¶
+The literal header field without name reference representation encodes a header +field name and a header field value as string literals.¶
+
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
@@ -1101,682 +2361,772 @@
| Value String (Length bytes) |
+-------------------------------+
-Literal Header Field Without Name Reference
-This representation begins with the ‘001’ three-bit pattern. The fourth bit is the ‘N’ bit as described in Section 4.5.4. The name follows, represented as a 4-bit prefix string literal, then the value, represented as an 8-bit prefix string literal (see Section 4.1.2).
-QPACK defines two settings which are included in the HTTP/3 SETTINGS frame.
- - -This representation begins with the '001' three-bit pattern. The fourth bit is +the 'N' bit as described in Section 4.5.4. The name follows, +represented as a 4-bit prefix string literal, then the value, represented as an +8-bit prefix string literal (see Section 4.1.2).¶
+QPACK defines two settings which are included in the HTTP/3 SETTINGS frame.¶
+The following error codes are defined for HTTP/3 to indicate failures of QPACK which prevent the connection from continuing:
- - -The following error codes are defined for HTTP/3 to indicate failures of +QPACK which prevent the connection from continuing:¶
+TBD. Also see Section 7.1 of [RFC7541].
-While the negotiated limit on the dynamic table size accounts for much of the memory that can be consumed by a QPACK implementation, data which cannot be immediately sent due to flow control is not affected by this limit. Implementations should limit the size of unsent data, especially on the decoder stream where flexibility to choose what to send is limited. Possible responses to an excess of unsent data might include limiting the ability of the peer to open new streams, reading only from the encoder stream, or closing the connection.
-This document specifies two settings. The entries in the following table are registered in the “HTTP/3 Settings” registry established in [HTTP3].
-| Setting Name | -Code | -Specification | -Default | -
|---|---|---|---|
| QPACK_MAX_TABLE_CAPACITY | -0x1 | -Section 5 | -0 | -
| QPACK_BLOCKED_STREAMS | -0x7 | -Section 5 | -0 | -
This document specifies two stream types. The entries in the following table are registered in the “HTTP/3 Stream Type” registry established in [HTTP3].
-| Stream Type | -Code | -Specification | -Sender | -
|---|---|---|---|
| QPACK Encoder Stream | -0x02 | -Section 4.2 | -Both | -
| QPACK Decoder Stream | -0x03 | -Section 4.2 | -Both | -
This document specifies three error codes. The entries in the following table are registered in the “HTTP/3 Error Code” registry established in [HTTP3].
-| Name | -Code | -Description | -Specification | -
|---|---|---|---|
| HTTP_QPACK_DECOMPRESSION_FAILED | -0x200 | -Decompression of a header block failed | -Section 6 | -
| HTTP_QPACK_ENCODER_STREAM_ERROR | -0x201 | -Error on the encoder stream | -Section 6 | -
| HTTP_QPACK_DECODER_STREAM_ERROR | -0x202 | -Error on the decoder stream | -Section 6 | -
| [HTTP3] | --Bishop, M., "Hypertext Transfer Protocol Version 3 (HTTP/3)", Internet-Draft draft-ietf-quic-http, November 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, November 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC7541] | --Peon, R. and H. Ruellan, "HPACK: Header Compression for HTTP/2", RFC 7541, DOI 10.17487/RFC7541, May 2015. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC2360] | --Scott, G., "Guide for Internet Standards Writers", BCP 22, RFC 2360, DOI 10.17487/RFC2360, June 1998. | -
| [RFC7540] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| Index | -Name | -Value | -
|---|---|---|
| 0 | -:authority | -- |
| 1 | -:path | -/ | -
| 2 | -age | -0 | -
| 3 | -content-disposition | -- |
| 4 | -content-length | -0 | -
| 5 | -cookie | -- |
| 6 | -date | -- |
| 7 | -etag | -- |
| 8 | -if-modified-since | -- |
| 9 | -if-none-match | -- |
| 10 | -last-modified | -- |
| 11 | -link | -- |
| 12 | -location | -- |
| 13 | -referer | -- |
| 14 | -set-cookie | -- |
| 15 | -:method | -CONNECT | -
| 16 | -:method | -DELETE | -
| 17 | -:method | -GET | -
| 18 | -:method | -HEAD | -
| 19 | -:method | -OPTIONS | -
| 20 | -:method | -POST | -
| 21 | -:method | -PUT | -
| 22 | -:scheme | -http | -
| 23 | -:scheme | -https | -
| 24 | -:status | -103 | -
| 25 | -:status | -200 | -
| 26 | -:status | -304 | -
| 27 | -:status | -404 | -
| 28 | -:status | -503 | -
| 29 | -accept | -*/* | -
| 30 | -accept | -application/dns-message | -
| 31 | -accept-encoding | -gzip, deflate, br | -
| 32 | -accept-ranges | -bytes | -
| 33 | -access-control-allow-headers | -cache-control | -
| 34 | -access-control-allow-headers | -content-type | -
| 35 | -access-control-allow-origin | -* | -
| 36 | -cache-control | -max-age=0 | -
| 37 | -cache-control | -max-age=2592000 | -
| 38 | -cache-control | -max-age=604800 | -
| 39 | -cache-control | -no-cache | -
| 40 | -cache-control | -no-store | -
| 41 | -cache-control | -public, max-age=31536000 | -
| 42 | -content-encoding | -br | -
| 43 | -content-encoding | -gzip | -
| 44 | -content-type | -application/dns-message | -
| 45 | -content-type | -application/javascript | -
| 46 | -content-type | -application/json | -
| 47 | -content-type | -application/x-www-form-urlencoded | -
| 48 | -content-type | -image/gif | -
| 49 | -content-type | -image/jpeg | -
| 50 | -content-type | -image/png | -
| 51 | -content-type | -text/css | -
| 52 | -content-type | -text/html; charset=utf-8 | -
| 53 | -content-type | -text/plain | -
| 54 | -content-type | -text/plain;charset=utf-8 | -
| 55 | -range | -bytes=0- | -
| 56 | -strict-transport-security | -max-age=31536000 | -
| 57 | -strict-transport-security | -max-age=31536000; includesubdomains | -
| 58 | -strict-transport-security | -max-age=31536000; includesubdomains; preload | -
| 59 | -vary | -accept-encoding | -
| 60 | -vary | -origin | -
| 61 | -x-content-type-options | -nosniff | -
| 62 | -x-xss-protection | -1; mode=block | -
| 63 | -:status | -100 | -
| 64 | -:status | -204 | -
| 65 | -:status | -206 | -
| 66 | -:status | -302 | -
| 67 | -:status | -400 | -
| 68 | -:status | -403 | -
| 69 | -:status | -421 | -
| 70 | -:status | -425 | -
| 71 | -:status | -500 | -
| 72 | -accept-language | -- |
| 73 | -access-control-allow-credentials | -FALSE | -
| 74 | -access-control-allow-credentials | -TRUE | -
| 75 | -access-control-allow-headers | -* | -
| 76 | -access-control-allow-methods | -get | -
| 77 | -access-control-allow-methods | -get, post, options | -
| 78 | -access-control-allow-methods | -options | -
| 79 | -access-control-expose-headers | -content-length | -
| 80 | -access-control-request-headers | -content-type | -
| 81 | -access-control-request-method | -get | -
| 82 | -access-control-request-method | -post | -
| 83 | -alt-svc | -clear | -
| 84 | -authorization | -- |
| 85 | -content-security-policy | -script-src 'none'; object-src 'none'; base-uri 'none' | -
| 86 | -early-data | -1 | -
| 87 | -expect-ct | -- |
| 88 | -forwarded | -- |
| 89 | -if-range | -- |
| 90 | -origin | -- |
| 91 | -purpose | -prefetch | -
| 92 | -server | -- |
| 93 | -timing-allow-origin | -* | -
| 94 | -upgrade-insecure-requests | -1 | -
| 95 | -user-agent | -- |
| 96 | -x-forwarded-for | -- |
| 97 | -x-frame-options | -deny | -
| 98 | -x-frame-options | -sameorigin | -
Pseudo-code for single pass encoding, excluding handling of duplicates, non-blocking mode, and reference tracking.
+TBD. Also see Section 7.1 of [RFC7541].¶
+While the negotiated limit on the dynamic table size accounts for much of the +memory that can be consumed by a QPACK implementation, data which cannot be +immediately sent due to flow control is not affected by this limit. +Implementations should limit the size of unsent data, especially on the decoder +stream where flexibility to choose what to send is limited. Possible responses +to an excess of unsent data might include limiting the ability of the peer to +open new streams, reading only from the encoder stream, or closing the +connection.¶
+This document specifies two settings. The entries in the following table are +registered in the "HTTP/3 Settings" registry established in [HTTP3].¶
+| Setting Name | +Code | +Specification | +Default | +
|---|---|---|---|
| QPACK_MAX_TABLE_CAPACITY | +0x1 | ++ Section 5 + | +0 | +
| QPACK_BLOCKED_STREAMS | +0x7 | ++ Section 5 + | +0 | +
This document specifies two stream types. The entries in the following table are +registered in the "HTTP/3 Stream Type" registry established in [HTTP3].¶
+| Stream Type | +Code | +Specification | +Sender | +
|---|---|---|---|
| QPACK Encoder Stream | +0x02 | ++ Section 4.2 + | +Both | +
| QPACK Decoder Stream | +0x03 | ++ Section 4.2 + | +Both | +
This document specifies three error codes. The entries in the following table +are registered in the "HTTP/3 Error Code" registry established in [HTTP3].¶
+| Name | +Code | +Description | +Specification | +
|---|---|---|---|
| HTTP_QPACK_DECOMPRESSION_FAILED | +0x200 | +Decompression of a header block failed | ++ Section 6 + | +
| HTTP_QPACK_ENCODER_STREAM_ERROR | +0x201 | +Error on the encoder stream | ++ Section 6 + | +
| HTTP_QPACK_DECODER_STREAM_ERROR | +0x202 | +Error on the decoder stream | ++ Section 6 + | +
| Index | +Name | +Value | +
|---|---|---|
| 0 | +:authority | +Â | +
| 1 | +:path | +/ | +
| 2 | +age | +0 | +
| 3 | +content-disposition | +Â | +
| 4 | +content-length | +0 | +
| 5 | +cookie | +Â | +
| 6 | +date | +Â | +
| 7 | +etag | +Â | +
| 8 | +if-modified-since | +Â | +
| 9 | +if-none-match | +Â | +
| 10 | +last-modified | +Â | +
| 11 | +link | +Â | +
| 12 | +location | +Â | +
| 13 | +referer | +Â | +
| 14 | +set-cookie | +Â | +
| 15 | +:method | +CONNECT | +
| 16 | +:method | +DELETE | +
| 17 | +:method | +GET | +
| 18 | +:method | +HEAD | +
| 19 | +:method | +OPTIONS | +
| 20 | +:method | +POST | +
| 21 | +:method | +PUT | +
| 22 | +:scheme | +http | +
| 23 | +:scheme | +https | +
| 24 | +:status | +103 | +
| 25 | +:status | +200 | +
| 26 | +:status | +304 | +
| 27 | +:status | +404 | +
| 28 | +:status | +503 | +
| 29 | +accept | +*/* | +
| 30 | +accept | +application/dns-message | +
| 31 | +accept-encoding | +gzip, deflate, br | +
| 32 | +accept-ranges | +bytes | +
| 33 | +access-control-allow-headers | +cache-control | +
| 34 | +access-control-allow-headers | +content-type | +
| 35 | +access-control-allow-origin | +* | +
| 36 | +cache-control | +max-age=0 | +
| 37 | +cache-control | +max-age=2592000 | +
| 38 | +cache-control | +max-age=604800 | +
| 39 | +cache-control | +no-cache | +
| 40 | +cache-control | +no-store | +
| 41 | +cache-control | +public, max-age=31536000 | +
| 42 | +content-encoding | +br | +
| 43 | +content-encoding | +gzip | +
| 44 | +content-type | +application/dns-message | +
| 45 | +content-type | +application/javascript | +
| 46 | +content-type | +application/json | +
| 47 | +content-type | +application/x-www-form-urlencoded | +
| 48 | +content-type | +image/gif | +
| 49 | +content-type | +image/jpeg | +
| 50 | +content-type | +image/png | +
| 51 | +content-type | +text/css | +
| 52 | +content-type | +text/html; charset=utf-8 | +
| 53 | +content-type | +text/plain | +
| 54 | +content-type | +text/plain;charset=utf-8 | +
| 55 | +range | +bytes=0- | +
| 56 | +strict-transport-security | +max-age=31536000 | +
| 57 | +strict-transport-security | +max-age=31536000; includesubdomains | +
| 58 | +strict-transport-security | +max-age=31536000; includesubdomains; preload | +
| 59 | +vary | +accept-encoding | +
| 60 | +vary | +origin | +
| 61 | +x-content-type-options | +nosniff | +
| 62 | +x-xss-protection | +1; mode=block | +
| 63 | +:status | +100 | +
| 64 | +:status | +204 | +
| 65 | +:status | +206 | +
| 66 | +:status | +302 | +
| 67 | +:status | +400 | +
| 68 | +:status | +403 | +
| 69 | +:status | +421 | +
| 70 | +:status | +425 | +
| 71 | +:status | +500 | +
| 72 | +accept-language | +Â | +
| 73 | +access-control-allow-credentials | +FALSE | +
| 74 | +access-control-allow-credentials | +TRUE | +
| 75 | +access-control-allow-headers | +* | +
| 76 | +access-control-allow-methods | +get | +
| 77 | +access-control-allow-methods | +get, post, options | +
| 78 | +access-control-allow-methods | +options | +
| 79 | +access-control-expose-headers | +content-length | +
| 80 | +access-control-request-headers | +content-type | +
| 81 | +access-control-request-method | +get | +
| 82 | +access-control-request-method | +post | +
| 83 | +alt-svc | +clear | +
| 84 | +authorization | +Â | +
| 85 | +content-security-policy | +script-src 'none'; object-src 'none'; base-uri 'none' | +
| 86 | +early-data | +1 | +
| 87 | +expect-ct | +Â | +
| 88 | +forwarded | +Â | +
| 89 | +if-range | +Â | +
| 90 | +origin | +Â | +
| 91 | +purpose | +prefetch | +
| 92 | +server | +Â | +
| 93 | +timing-allow-origin | +* | +
| 94 | +upgrade-insecure-requests | +1 | +
| 95 | +user-agent | +Â | +
| 96 | +x-forwarded-for | +Â | +
| 97 | +x-frame-options | +deny | +
| 98 | +x-frame-options | +sameorigin | +
Pseudo-code for single pass encoding, excluding handling of duplicates, +non-blocking mode, and reference tracking.¶
+baseIndex = dynamicTable.baseIndex largestReference = 0 @@ -1823,193 +3173,288 @@-largestReference - baseIndex, 7) return controlBuffer, prefixBuffer + streamBuffer -
Editorial changes
-Editorial changes¶
+This draft draws heavily on the text of [RFC7541]. The indirect input of those authors is gratefully acknowledged, as well as ideas from:
- - +Buck’s contribution was supported by Google during his employment there.
-A substantial portion of Mike’s contribution was supported by Microsoft during his employment there.
-This draft draws heavily on the text of [RFC7541]. The indirect input of +those authors is gratefully acknowledged, as well as ideas from:¶
+Buck's contribution was supported by Google during his employment there.¶
+A substantial portion of Mike's contribution was supported by Microsoft during +his employment there.¶
+| QUIC | -J. Iyengar, Ed. | -
| Internet-Draft | -Fastly | -
| Intended status: Standards Track | -I. Swett, Ed. | -
| Expires: May 15, 2020 | -|
| - | November 12, 2019 | -
QUIC Loss Detection and Congestion Control
- draft-ietf-quic-recovery-latest
This document describes loss detection and congestion control mechanisms for QUIC.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-recovery.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
+address.vcard { + max-width: 20em; + margin-right: auto; +} - - -| Internet-Draft | +QUIC Loss Detection | +January 2020 | +
| Iyengar & Swett | +Expires 16 July 2020 | +[Page] | +
This document describes loss detection and congestion control mechanisms for +QUIC.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +https://mailarchive.ietf.org/arch/search/?email_list=quic.¶
+Working Group information can be found at https://github.com/quicwg; source +code and issues list for this draft can be found at +https://github.com/quicwg/base-drafts/labels/-recovery.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades +of transport and security experience, and implements mechanisms that make it +attractive as a modern general-purpose transport. The QUIC protocol is +described in [QUIC-TRANSPORT].¶
+QUIC implements the spirit of existing TCP congestion control and loss recovery +mechanisms, described in RFCs, various Internet-drafts, and also those prevalent +in the Linux TCP implementation. This document describes QUIC congestion +control and loss recovery, and where applicable, attributes the TCP equivalent +in RFCs, Internet-drafts, academic papers, and/or TCP implementations.¶
+The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+Definitions of terms that are used in this document:¶
+All transmissions in QUIC are sent with a packet-level header, which indicates +the encryption level and includes a packet sequence number (referred to below as +a packet number). The encryption level indicates the packet number space, as +described in [QUIC-TRANSPORT]. Packet numbers never repeat within a packet +number space for the lifetime of a connection. Packet numbers are sent in +monotonically increasing order within a space, preventing ambiguity.¶
+This design obviates the need for disambiguating between transmissions and +retransmissions and eliminates significant complexity from QUIC's interpretation +of TCP loss detection mechanisms.¶
+QUIC packets can contain multiple frames of different types. The recovery +mechanisms ensure that data and frames that need reliable delivery are +acknowledged or declared lost and sent in new packets as necessary. The types +of frames contained in a packet affect recovery and congestion control logic:¶
+QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades of transport and security experience, and implements mechanisms that make it attractive as a modern general-purpose transport. The QUIC protocol is described in [QUIC-TRANSPORT].
-QUIC implements the spirit of existing TCP loss recovery mechanisms, described in RFCs, various Internet-drafts, and also those prevalent in the Linux TCP implementation. This document describes QUIC congestion control and loss recovery, and where applicable, attributes the TCP equivalent in RFCs, Internet-drafts, academic papers, and/or TCP implementations.
-The key words “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Definitions of terms that are used in this document:
- - -All transmissions in QUIC are sent with a packet-level header, which indicates the encryption level and includes a packet sequence number (referred to below as a packet number). The encryption level indicates the packet number space, as described in [QUIC-TRANSPORT]. Packet numbers never repeat within a packet number space for the lifetime of a connection. Packet numbers monotonically increase within a space, preventing ambiguity.
-This design obviates the need for disambiguating between transmissions and retransmissions and eliminates significant complexity from QUIC’s interpretation of TCP loss detection mechanisms.
-QUIC packets can contain multiple frames of different types. The recovery mechanisms ensure that data and frames that need reliable delivery are acknowledged or declared lost and sent in new packets as necessary. The types of frames contained in a packet affect recovery and congestion control logic:
- - -Readers familiar with TCP’s loss detection and congestion control will find algorithms here that parallel well-known TCP ones. Protocol differences between QUIC and TCP however contribute to algorithmic differences. We briefly describe these protocol differences below.
-QUIC uses separate packet number spaces for each encryption level, except 0-RTT and all generations of 1-RTT keys use the same packet number space. Separate packet number spaces ensures acknowledgement of packets sent with one level of encryption will not cause spurious retransmission of packets sent with a different encryption level. Congestion control and round-trip time (RTT) measurement are unified across packet number spaces.
-TCP conflates transmission order at the sender with delivery order at the receiver, which results in retransmissions of the same data carrying the same sequence number, and consequently leads to “retransmission ambiguity”. QUIC separates the two: QUIC uses a packet number to indicate transmission order, and any application data is sent in one or more streams, with delivery order determined by stream offsets encoded within STREAM frames.
-QUIC’s packet number is strictly increasing within a packet number space, and directly encodes transmission order. A higher packet number signifies that the packet was sent later, and a lower packet number signifies that the packet was sent earlier. When a packet containing ack-eliciting frames is detected lost, QUIC rebundles necessary frames in a new packet with a new packet number, removing ambiguity about which packet is acknowledged when an ACK is received. Consequently, more accurate RTT measurements can be made, spurious retransmissions are trivially detected, and mechanisms such as Fast Retransmit can be applied universally, based only on packet number.
-This design point significantly simplifies loss detection mechanisms for QUIC. Most TCP mechanisms implicitly attempt to infer transmission ordering based on TCP sequence numbers - a non-trivial task, especially when TCP timestamps are not available.
-QUIC starts a loss epoch when a packet is lost and ends one when any packet sent after the epoch starts is acknowledged. TCP waits for the gap in the sequence number space to be filled, and so if a segment is lost multiple times in a row, the loss epoch may not end for several round trips. Because both should reduce their congestion windows only once per epoch, QUIC will do it correctly once for every round trip that experiences loss, while TCP may only do it once across multiple round trips.
-QUIC ACKs contain information that is similar to TCP SACK, but QUIC does not allow any acked packet to be reneged, greatly simplifying implementations on both sides and reducing memory pressure on the sender.
-QUIC supports many ACK ranges, opposed to TCP’s 3 SACK ranges. In high loss environments, this speeds recovery, reduces spurious retransmits, and ensures forward progress without relying on timeouts.
-QUIC endpoints measure the delay incurred between when a packet is received and when the corresponding acknowledgment is sent, allowing a peer to maintain a more accurate round-trip time estimate (see Section 13.2 of [QUIC-TRANSPORT]).
-At a high level, an endpoint measures the time from when a packet was sent to when it is acknowledged as a round-trip time (RTT) sample. The endpoint uses RTT samples and peer-reported host delays (see Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical description of the path’s RTT. An endpoint computes the following three values observed during the congestion control context: the minimum value (min_rtt), an exponentially-weighted moving average (smoothed_rtt), and the variance in the observed RTT samples (rttvar).
-An endpoint generates an RTT sample on receiving an ACK frame that meets the following two conditions:
- - +Readers familiar with TCP's loss detection and congestion control will find +algorithms here that parallel well-known TCP ones. Protocol differences between +QUIC and TCP however contribute to algorithmic differences. We briefly describe +these protocol differences below.¶
+QUIC uses separate packet number spaces for each encryption level, except 0-RTT +and all generations of 1-RTT keys use the same packet number space. Separate +packet number spaces ensures acknowledgement of packets sent with one level of +encryption will not cause spurious retransmission of packets sent with a +different encryption level. Congestion control and round-trip time (RTT) +measurement are unified across packet number spaces.¶
+TCP conflates transmission order at the sender with delivery order at the +receiver, which results in retransmissions of the same data carrying the same +sequence number, and consequently leads to "retransmission ambiguity". QUIC +separates the two: QUIC uses a packet number to indicate transmission order, +and any application data is sent in one or more streams, with delivery order +determined by stream offsets encoded within STREAM frames.¶
+QUIC's packet number is strictly increasing within a packet number space, +and directly encodes transmission order. A higher packet number signifies +that the packet was sent later, and a lower packet number signifies that +the packet was sent earlier. When a packet containing ack-eliciting +frames is detected lost, QUIC rebundles necessary frames in a new packet +with a new packet number, removing ambiguity about which packet is +acknowledged when an ACK is received. Consequently, more accurate RTT +measurements can be made, spurious retransmissions are trivially detected, and +mechanisms such as Fast Retransmit can be applied universally, based only on +packet number.¶
+This design point significantly simplifies loss detection mechanisms for QUIC. +Most TCP mechanisms implicitly attempt to infer transmission ordering based on +TCP sequence numbers - a non-trivial task, especially when TCP timestamps are +not available.¶
+QUIC starts a loss epoch when a packet is lost and ends one when any packet +sent after the epoch starts is acknowledged. TCP waits for the gap in the +sequence number space to be filled, and so if a segment is lost multiple times +in a row, the loss epoch may not end for several round trips. Because both +should reduce their congestion windows only once per epoch, QUIC will do it +once for every round trip that experiences loss, while TCP may only do it +once across multiple round trips.¶
+QUIC ACKs contain information that is similar to TCP SACK, but QUIC does not +allow any acked packet to be reneged, greatly simplifying implementations on +both sides and reducing memory pressure on the sender.¶
+QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In high loss +environments, this speeds recovery, reduces spurious retransmits, and ensures +forward progress without relying on timeouts.¶
+QUIC endpoints measure the delay incurred between when a packet is received and +when the corresponding acknowledgment is sent, allowing a peer to maintain a +more accurate round-trip time estimate (see Section 13.2 of [QUIC-TRANSPORT]).¶
+At a high level, an endpoint measures the time from when a packet was sent to +when it is acknowledged as a round-trip time (RTT) sample. The endpoint uses +RTT samples and peer-reported host delays (see Section 13.2 of +[QUIC-TRANSPORT]) to generate a statistical description of the network +path's RTT. An endpoint computes the following three values for each path: +the minimum value observed over the lifetime of the path (min_rtt), an +exponentially-weighted moving average (smoothed_rtt), and the mean deviation +(referred to as "variation" in the rest of this document) in the observed RTT +samples (rttvar).¶
+An endpoint generates an RTT sample on receiving an ACK frame that meets the +following two conditions:¶
The RTT sample, latest_rtt, is generated as the time elapsed since the largest acknowledged packet was sent:
+The RTT sample, latest_rtt, is generated as the time elapsed since the largest +acknowledged packet was sent:¶
+latest_rtt = ack_time - send_time_of_largest_acked --
An RTT sample is generated using only the largest acknowledged packet in the received ACK frame. This is because a peer reports host delays for only the largest acknowledged packet in an ACK frame. While the reported host delay is not used by the RTT sample measurement, it is used to adjust the RTT sample in subsequent computations of smoothed_rtt and rttvar Section 4.3.
-To avoid generating multiple RTT samples using the same packet, an ACK frame SHOULD NOT be used to update RTT estimates if it does not newly acknowledge the largest acknowledged packet.
-An RTT sample MUST NOT be generated on receiving an ACK frame that does not newly acknowledge at least one ack-eliciting packet. A peer does not send an ACK frame on receiving only non-ack-eliciting packets, so an ACK frame that is subsequently sent can include an arbitrarily large Ack Delay field. Ignoring such ACK frames avoids complications in subsequent smoothed_rtt and rttvar computations.
-A sender might generate multiple RTT samples per RTT when multiple ACK frames are received within an RTT. As suggested in [RFC6298], doing so might result in inadequate history in smoothed_rtt and rttvar. Ensuring that RTT estimates retain sufficient history is an open research question.
-min_rtt is the minimum RTT observed for a given congestion control context. min_rtt is set to the latest_rtt on the first RTT sample, and to the lesser of min_rtt and latest_rtt on subsequent samples. min_rtt is used to reject implausible RTT samples, but is not used directly in congestion control or loss recovery.
-An endpoint uses only locally observed times in computing the min_rtt and does not adjust for host delays reported by the peer. Doing so allows the endpoint to set a lower bound for the smoothed_rtt based entirely on what it observes (see Section 4.3), and limits potential underestimation due to erroneously-reported delays by the peer.
-The RTT for a path may change over time. If a path’s actual RTT decreases, the min_rtt will adapt immediately on the first low sample. If the path’s actual RTT increases, the min_rtt will not adapt to it, allowing future RTT samples that are smaller than the new RTT be included in smoothed_rtt.
-smoothed_rtt is an exponentially-weighted moving average of an endpoint’s RTT samples, and rttvar is the endpoint’s estimated variance in the RTT samples.
-The calculation of smoothed_rtt uses path latency after adjusting RTT samples for host delays. For packets sent in the ApplicationData packet number space, a peer limits any delay in sending an acknowledgement for an ack-eliciting packet to no greater than the value it advertised in the max_ack_delay transport parameter. Consequently, when a peer reports an Ack Delay that is greater than its max_ack_delay, the delay is attributed to reasons out of the peer’s control, such as scheduler latency at the peer or loss of previous ACK frames. Any delays beyond the peer’s max_ack_delay are therefore considered effectively part of path delay and incorporated into the smoothed_rtt estimate.
-When adjusting an RTT sample using peer-reported acknowledgement delays, an endpoint:
- - +¶ +An RTT sample is generated using only the largest acknowledged packet in the +received ACK frame. This is because a peer reports ACK delays for only the +largest acknowledged packet in an ACK frame. While the reported ACK delay is +not used by the RTT sample measurement, it is used to adjust the RTT sample in +subsequent computations of smoothed_rtt and rttvar Section 4.3.¶
+To avoid generating multiple RTT samples for a single packet, an ACK frame +SHOULD NOT be used to update RTT estimates if it does not newly acknowledge the +largest acknowledged packet.¶
+An RTT sample MUST NOT be generated on receiving an ACK frame that does not +newly acknowledge at least one ack-eliciting packet. A peer does not send an +ACK frame on receiving only non-ack-eliciting packets, so an ACK frame that is +subsequently sent can include an arbitrarily large Ack Delay field. Ignoring +such ACK frames avoids complications in subsequent smoothed_rtt and rttvar +computations.¶
+A sender might generate multiple RTT samples per RTT when multiple ACK frames +are received within an RTT. As suggested in [RFC6298], doing so might result +in inadequate history in smoothed_rtt and rttvar. Ensuring that RTT estimates +retain sufficient history is an open research question.¶
+min_rtt is the minimum RTT observed for a given network path. min_rtt is set +to the latest_rtt on the first RTT sample, and to the lesser of min_rtt and +latest_rtt on subsequent samples. In this document, min_rtt is used by loss +detection to reject implausibly small rtt samples.¶
+An endpoint uses only locally observed times in computing the min_rtt and does +not adjust for ACK delays reported by the peer. Doing so allows the endpoint +to set a lower bound for the smoothed_rtt based entirely on what it observes +(see Section 4.3), and limits potential underestimation due to +erroneously-reported delays by the peer.¶
+The RTT for a network path may change over time. If a path's actual RTT +decreases, the min_rtt will adapt immediately on the first low sample. If +the path's actual RTT increases, the min_rtt will not adapt to it, allowing +future RTT samples that are smaller than the new RTT be included in +smoothed_rtt.¶
+smoothed_rtt is an exponentially-weighted moving average of an endpoint's RTT +samples, and rttvar is the variation in the RTT samples, estimated using a +mean variation.¶
+The calculation of smoothed_rtt uses path latency after adjusting RTT samples +for ACK delays. For packets sent in the ApplicationData packet number space, +a peer limits any delay in sending an acknowledgement for an ack-eliciting +packet to no greater than the value it advertised in the max_ack_delay transport +parameter. Consequently, when a peer reports an Ack Delay that is greater than +its max_ack_delay, the delay is attributed to reasons out of the peer's control, +such as scheduler latency at the peer or loss of previous ACK frames. Any +delays beyond the peer's max_ack_delay are therefore considered effectively +part of path delay and incorporated into the smoothed_rtt estimate.¶
+When adjusting an RTT sample using peer-reported acknowledgement delays, an +endpoint:¶
On the first RTT sample in a congestion control context, the smoothed_rtt is set to the latest_rtt.
-smoothed_rtt and rttvar are computed as follows, similar to [RFC6298]. On the first RTT sample in a congestion control context:
+On the first RTT sample for a network path, the smoothed_rtt is set to the +latest_rtt.¶
+smoothed_rtt and rttvar are computed as follows, similar to [RFC6298]. On +the first RTT sample for a network path:¶
+smoothed_rtt = latest_rtt rttvar = latest_rtt / 2 --
On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:
+¶ +On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:¶
+ack_delay = min(Ack Delay in ACK Frame, max_ack_delay) adjusted_rtt = latest_rtt @@ -750,397 +1537,823 @@-smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt rttvar_sample = abs(smoothed_rtt - adjusted_rtt) rttvar = 3/4 * rttvar + 1/4 * rttvar_sample -
QUIC senders use both ack information and timeouts to detect lost packets, and this section provides a description of these algorithms.
-If a packet is lost, the QUIC transport needs to recover from that loss, such as by retransmitting the data, sending an updated frame, or abandoning the frame. For more information, see Section 13.3 of [QUIC-TRANSPORT].
-Acknowledgement-based loss detection implements the spirit of TCP’s Fast Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK], SACK loss recovery [RFC6675], and RACK [RACK]. This section provides an overview of how these algorithms are implemented in QUIC.
-A packet is declared lost if it meets all the following conditions:
- - +¶ +QUIC senders use acknowledgements to detect lost packets, and a probe +time out Section 5.2 to ensure acknowledgements are received. This section +provides a description of these algorithms.¶
+If a packet is lost, the QUIC transport needs to recover from that loss, such +as by retransmitting the data, sending an updated frame, or abandoning the +frame. For more information, see Section 13.3 of [QUIC-TRANSPORT].¶
+Acknowledgement-based loss detection implements the spirit of TCP's Fast +Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK], SACK loss +recovery [RFC6675], and RACK [RACK]. This section +provides an overview of how these algorithms are implemented in QUIC.¶
+A packet is declared lost if it meets all the following conditions:¶
The acknowledgement indicates that a packet sent later was delivered, while the packet and time thresholds provide some tolerance for packet reordering.
-Spuriously declaring packets as lost leads to unnecessary retransmissions and may result in degraded performance due to the actions of the congestion controller upon detecting loss. Implementations that detect spurious retransmissions and increase the reordering threshold in packets or time MAY choose to start with smaller initial reordering thresholds to minimize recovery latency.
-The RECOMMENDED initial value for the packet reordering threshold (kPacketThreshold) is 3, based on best practices for TCP loss detection [RFC5681] [RFC6675].
-Some networks may exhibit higher degrees of reordering, causing a sender to detect spurious losses. Implementers MAY use algorithms developed for TCP, such as TCP-NCR [RFC4653], to improve QUIC’s reordering resilience.
-Once a later packet packet within the same packet number space has been acknowledged, an endpoint SHOULD declare an earlier packet lost if it was sent a threshold amount of time in the past. To avoid declaring packets as lost too early, this time threshold MUST be set to at least kGranularity. The time threshold is:
+The acknowledgement indicates that a packet sent later was delivered, and the +packet and time thresholds provide some tolerance for packet reordering.¶
+Spuriously declaring packets as lost leads to unnecessary retransmissions and +may result in degraded performance due to the actions of the congestion +controller upon detecting loss. Implementations that detect spurious +retransmissions and increase the reordering threshold in packets or time MAY +choose to start with smaller initial reordering thresholds to minimize recovery +latency.¶
+The RECOMMENDED initial value for the packet reordering threshold +(kPacketThreshold) is 3, based on best practices for TCP loss detection +[RFC5681] [RFC6675]. Implementations SHOULD NOT use a packet threshold +less than 3, to keep in line with TCP [RFC5681].¶
+Some networks may exhibit higher degrees of reordering, causing a sender to +detect spurious losses. Implementers MAY use algorithms developed for TCP, such +as TCP-NCR [RFC4653], to improve QUIC's reordering resilience.¶
+Once a later packet within the same packet number space has been acknowledged, +an endpoint SHOULD declare an earlier packet lost if it was sent a threshold +amount of time in the past. To avoid declaring packets as lost too early, this +time threshold MUST be set to at least kGranularity. The time threshold is:¶
+-kTimeThreshold * max(smoothed_rtt, latest_rtt, kGranularity) --
If packets sent prior to the largest acknowledged packet cannot yet be declared lost, then a timer SHOULD be set for the remaining time.
-Using max(smoothed_rtt, latest_rtt) protects from the two following cases:
- - +max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity) +¶ +If packets sent prior to the largest acknowledged packet cannot yet be declared +lost, then a timer SHOULD be set for the remaining time.¶
+Using max(smoothed_rtt, latest_rtt) protects from the two following cases:¶
The RECOMMENDED time threshold (kTimeThreshold), expressed as a round-trip time multiplier, is 9/8.
-Implementations MAY experiment with absolute thresholds, thresholds from previous connections, adaptive thresholds, or including RTT variance. Smaller thresholds reduce reordering resilience and increase spurious retransmissions, and larger thresholds increase loss detection delay.
-A Probe Timeout (PTO) triggers sending one or two probe datagrams when ack-eliciting packets are not acknowledged within the expected period of time or the handshake has not been completed. A PTO enables a connection to recover from loss of tail packets or acknowledgements.
-As with loss detection, the probe timeout is per packet number space. The PTO algorithm used in QUIC implements the reliability functions of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for TCP [RFC5682]. The timeout computation is based on TCP’s retransmission timeout period [RFC6298].
-When an ack-eliciting packet is transmitted, the sender schedules a timer for the PTO period as follows:
+The RECOMMENDED time threshold (kTimeThreshold), expressed as a round-trip time +multiplier, is 9/8.¶
+Implementations MAY experiment with absolute thresholds, thresholds from +previous connections, adaptive thresholds, or including RTT variation. Smaller +thresholds reduce reordering resilience and increase spurious retransmissions, +and larger thresholds increase loss detection delay.¶
+A Probe Timeout (PTO) triggers sending one or two probe datagrams when +ack-eliciting packets are not acknowledged within the expected period of +time or the handshake has not been completed. A PTO enables a connection to +recover from loss of tail packets or acknowledgements.¶
+As with loss detection, the probe timeout is per packet number space. +The PTO algorithm used in QUIC implements the reliability functions of +Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for +TCP [RFC5682]. The timeout computation is based on TCP's retransmission +timeout period [RFC6298].¶
+When an ack-eliciting packet is transmitted, the sender schedules a timer for +the PTO period as follows:¶
+PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay --
kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in Appendix A.2 and Appendix A.3.
-The PTO period is the amount of time that a sender ought to wait for an acknowledgement of a sent packet. This time period includes the estimated network roundtrip-time (smoothed_rtt), the variance in the estimate (4*rttvar), and max_ack_delay, to account for the maximum time by which a receiver might delay sending an acknowledgement. When the PTO is armed for Initial or Handshake packet number spaces, the max_ack_delay is 0, as specified in 13.2.5 of [QUIC-TRANSPORT].
-The PTO value MUST be set to at least kGranularity, to avoid the timer expiring immediately.
-A sender computes its PTO timer every time an ack-eliciting packet is sent. When ack-eliciting packets are in-flight in multiple packet number spaces, the timer MUST be set for the packet number space with the earliest timeout, except for ApplicationData, which MUST be ignored until the handshake completes; see Section 4.1.1 of [QUIC-TLS]. Not arming the PTO for ApplicationData prioritizes completing the handshake and prevents the server from sending a 1-RTT packet on a PTO before before it has the keys to process a 1-RTT packet.
-When a PTO timer expires, the PTO period MUST be set to twice its current value. This exponential reduction in the sender’s rate is important because the PTOs might be caused by loss of packets or acknowledgements due to severe congestion. Even when there are ack-eliciting packets in-flight in multiple packet number spaces, the exponential increase in probe timeout occurs across all spaces to prevent excess load on the network. For example, a timeout in the Initial packet number space doubles the length of the timeout in the Handshake packet number space.
-The life of a connection that is experiencing consecutive PTOs is limited by the endpoint’s idle timeout.
-The probe timer is not set if the time threshold Section 5.1.2 loss detection timer is set. The time threshold loss detection timer is expected to both expire earlier than the PTO and be less likely to spuriously retransmit data.
-The initial probe timeout for a new connection or new path SHOULD be set to twice the initial RTT. Resumed connections over the same network SHOULD use the previous connection’s final smoothed RTT value as the resumed connection’s initial RTT. If no previous RTT is available, the initial RTT SHOULD be set to 500ms, resulting in a 1 second initial timeout as recommended in [RFC6298].
-A connection MAY use the delay between sending a PATH_CHALLENGE and receiving a PATH_RESPONSE to seed initial_rtt for a new path, but the delay SHOULD NOT be considered an RTT sample.
-Until the server has validated the client’s address on the path, the amount of data it can send is limited to three times the amount of data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If no data can be sent, then the PTO alarm MUST NOT be armed until datagrams have been received from the client.
-Since the server could be blocked until more packets are received from the client, it is the client’s responsibility to send packets to unblock the server until it is certain that the server has finished its address validation (see Section 8 of [QUIC-TRANSPORT]). That is, the client MUST set the probe timer if the client has not received an acknowledgement for one of its Handshake or 1-RTT packets.
-Prior to handshake completion, when few to none RTT samples have been generated, it is possible that the probe timer expiration is due to an incorrect RTT estimate at the client. To allow the client to improve its RTT estimate, the new packet that it sends MUST be ack-eliciting. If Handshake keys are available to the client, it MUST send a Handshake packet, and otherwise it MUST send an Initial packet in a UDP datagram of at least 1200 bytes.
-Initial packets and Handshake packets may never be acknowledged, but they are removed from bytes in flight when the Initial and Handshake keys are discarded.
-When a PTO timer expires, a sender MUST send at least one ack-eliciting packet in the packet number space as a probe, unless there is no data available to send. An endpoint MAY send up to two full-sized datagrams containing ack-eliciting packets, to avoid an expensive consecutive PTO expiration due to a single lost datagram or transmit data from multiple packet number spaces.
-In addition to sending data in the packet number space for which the timer expired, the sender SHOULD send ack-eliciting packets from other packet number spaces with in-flight data, coalescing packets if possible.
-When the PTO timer expires, and there is new or previously sent unacknowledged data, it MUST be sent. Data that was previously sent with Initial encryption MUST be sent before Handshake data and data previously sent at Handshake encryption MUST be sent before any ApplicationData data.
-It is possible the sender has no new or previously-sent data to send. As an example, consider the following sequence of events: new application data is sent in a STREAM frame, deemed lost, then retransmitted in a new packet, and then the original transmission is acknowledged. When there is no data to send, the sender SHOULD send a PING or other ack-eliciting frame in a single packet, re-arming the PTO timer.
-Alternatively, instead of sending an ack-eliciting packet, the sender MAY mark any packets still in flight as lost. Doing so avoids sending an additional packet, but increases the risk that loss is declared too aggressively, resulting in an unnecessary rate reduction by the congestion controller.
-Consecutive PTO periods increase exponentially, and as a result, connection recovery latency increases exponentially as packets continue to be dropped in the network. Sending two packets on PTO expiration increases resilience to packet drops, thus reducing the probability of consecutive PTO events.
-Probe packets sent on a PTO MUST be ack-eliciting. A probe packet SHOULD carry new data when possible. A probe packet MAY carry retransmitted unacknowledged data when new data is unavailable, when flow control does not permit new data to be sent, or to opportunistically reduce loss recovery delay. Implementations MAY use alternative strategies for determining the content of probe packets, including sending new or retransmitted data based on the application’s priorities.
-When the PTO timer expires multiple times and new data cannot be sent, implementations must choose between sending the same payload every time or sending different payloads. Sending the same payload may be simpler and ensures the highest priority frames arrive first. Sending different payloads each time reduces the chances of spurious retransmission.
-Delivery or loss of packets in flight is established when an ACK frame is received that newly acknowledges one or more packets.
-A PTO timer expiration event does not indicate packet loss and MUST NOT cause prior unacknowledged packets to be marked as lost. When an acknowledgement is received that newly acknowledges packets, loss detection proceeds as dictated by packet and time threshold mechanisms; see Section 5.1.
-A Retry packet causes a client to send another Initial packet, effectively restarting the connection process. A Retry packet indicates that the Initial was received, but not processed. A Retry packet cannot be treated as an acknowledgment, because it does not indicate that a packet was processed or specify the packet number.
-Clients that receive a Retry packet reset congestion control and loss recovery state, including resetting any pending timers. Other connection state, in particular cryptographic handshake messages, is retained; see Section 17.2.5 of [QUIC-TRANSPORT].
-The client MAY compute an RTT estimate to the server as the time period from when the first Initial was sent to when a Retry or a Version Negotiation packet is received. The client MAY use this value in place of its default for the initial RTT estimate.
-When packet protection keys are discarded (see Section 4.9 of [QUIC-TLS]), all packets that were sent with those keys can no longer be acknowledged because their acknowledgements cannot be processed anymore. The sender MUST discard all recovery state associated with those packets and MUST remove them from the count of bytes in flight.
-Endpoints stop sending and receiving Initial packets once they start exchanging Handshake packets (see Section 17.2.2.1 of [QUIC-TRANSPORT]). At this point, recovery state for all in-flight Initial packets is discarded.
-When 0-RTT is rejected, recovery state for all in-flight 0-RTT packets is discarded.
-If a server accepts 0-RTT, but does not buffer 0-RTT packets that arrive before Initial packets, early 0-RTT packets will be declared lost, but that is expected to be infrequent.
-It is expected that keys are discarded after packets encrypted with them would be acknowledged or declared lost. Initial secrets however might be destroyed sooner, as soon as handshake keys are available (see Section 4.9.1 of [QUIC-TLS]).
-QUIC’s congestion control is based on TCP NewReno [RFC6582]. NewReno is a congestion window based congestion control. QUIC specifies the congestion window in bytes rather than packets due to finer control and the ease of appropriate byte counting [RFC3465].
-QUIC hosts MUST NOT send packets if they would increase bytes_in_flight (defined in Appendix B.2) beyond the available congestion window, unless the packet is a probe packet sent after a PTO timer expires, as described in Section 5.2.
-Implementations MAY use other congestion control algorithms, such as Cubic [RFC8312], and endpoints MAY use different algorithms from one another. The signals QUIC provides for congestion control are generic and are designed to support different algorithms.
-If a path has been verified to support ECN, QUIC treats a Congestion Experienced codepoint in the IP header as a signal of congestion. This document specifies an endpoint’s response when its peer receives packets with the Congestion Experienced codepoint. As discussed in [RFC8311], endpoints are permitted to experiment with other response functions.
-QUIC begins every connection in slow start and exits slow start upon loss or upon increase in the ECN-CE counter. QUIC re-enters slow start anytime the congestion window is less than ssthresh, which only occurs after persistent congestion is declared. While in slow start, QUIC increases the congestion window by the number of bytes acknowledged when each acknowledgment is processed.
-Slow start exits to congestion avoidance. Congestion avoidance in NewReno uses an additive increase multiplicative decrease (AIMD) approach that increases the congestion window by one maximum packet size per congestion window acknowledged. When a loss is detected, NewReno halves the congestion window and sets the slow start threshold to the new congestion window.
-Recovery is a period of time beginning with detection of a lost packet or an increase in the ECN-CE counter. Because QUIC does not retransmit packets, it defines the end of recovery as a packet sent after the start of recovery being acknowledged. This is slightly different from TCP’s definition of recovery, which ends when the lost packet that started recovery is acknowledged.
-The recovery period limits congestion window reduction to once per round trip. During recovery, the congestion window remains unchanged irrespective of new losses or increases in the ECN-CE counter.
-During the handshake, some packet protection keys might not be available when a packet arrives. In particular, Handshake and 0-RTT packets cannot be processed until the Initial packets arrive, and 1-RTT packets cannot be processed until the handshake completes. Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might arrive before the peer has packet protection keys to process those packets.
-Probe packets MUST NOT be blocked by the congestion controller. A sender MUST however count these packets as being additionally in flight, since these packets add network load without establishing packet loss. Note that sending probe packets might cause the sender’s bytes in flight to exceed the congestion window until an acknowledgement is received that establishes loss or delivery of packets.
-When an ACK frame is received that establishes loss of all in-flight packets sent over a long enough period of time, the network is considered to be experiencing persistent congestion. Commonly, this can be established by consecutive PTOs, but since the PTO timer is reset when a new ack-eliciting packet is sent, an explicit duration must be used to account for those cases where PTOs do not occur or are substantially delayed. This duration is computed as follows:
+¶ +kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in +Appendix A.2 and Appendix A.3.¶
+The PTO period is the amount of time that a sender ought to wait for an +acknowledgement of a sent packet. This time period includes the estimated +network roundtrip-time (smoothed_rtt), the variation in the estimate (4*rttvar), +and max_ack_delay, to account for the maximum time by which a receiver might +delay sending an acknowledgement. When the PTO is armed for Initial or +Handshake packet number spaces, the max_ack_delay is 0, as specified in +13.2.5 of [QUIC-TRANSPORT].¶
+The PTO value MUST be set to at least kGranularity, to avoid the timer expiring +immediately.¶
+A sender computes its PTO timer every time an ack-eliciting packet is sent. +When ack-eliciting packets are in-flight in multiple packet number spaces, +the timer MUST be set for the packet number space with the earliest timeout, +except for ApplicationData, which MUST be ignored until the handshake +completes; see Section 4.1.1 of [QUIC-TLS]. Not arming the PTO for +ApplicationData prioritizes completing the handshake and prevents the server +from sending a 1-RTT packet on a PTO before before it has the keys to process +a 1-RTT packet.¶
+When a PTO timer expires, the PTO period MUST be set to twice its current +value. This exponential reduction in the sender's rate is important because +consecutive PTOs might be caused by loss of packets or acknowledgements due to +severe congestion. Even when there are ack-eliciting packets in-flight in +multiple packet number spaces, the exponential increase in probe timeout +occurs across all spaces to prevent excess load on the network. For example, +a timeout in the Initial packet number space doubles the length of the timeout +in the Handshake packet number space.¶
+The life of a connection that is experiencing consecutive PTOs is limited by +the endpoint's idle timeout.¶
+The probe timer is not set if the time threshold Section 5.1.2 loss +detection timer is set. The time threshold loss detection timer is expected +to both expire earlier than the PTO and be less likely to spuriously retransmit +data.¶
+The initial probe timeout for a new connection or new path SHOULD be +set to twice the initial RTT. Resumed connections over the same network +SHOULD use the previous connection's final smoothed RTT value as the resumed +connection's initial RTT. If no previous RTT is available, the initial RTT +SHOULD be set to 500ms, resulting in a 1 second initial timeout as recommended +in [RFC6298].¶
+A connection MAY use the delay between sending a PATH_CHALLENGE and receiving a +PATH_RESPONSE to set the initial RTT (see kInitialRtt in +Appendix A.2) for a new path, but the delay SHOULD NOT be +considered an RTT sample.¶
+Until the server has validated the client's address on the path, the amount of +data it can send is limited to three times the amount of data received, +as specified in Section 8.1 of [QUIC-TRANSPORT]. If no data can be sent, +then the PTO alarm MUST NOT be armed until datagrams have been received from +the client.¶
+Since the server could be blocked until more packets are received from the +client, it is the client's responsibility to send packets to unblock the server +until it is certain that the server has finished its address validation +(see Section 8 of [QUIC-TRANSPORT]). That is, the client MUST set the +probe timer if the client has not received an acknowledgement for one of its +Handshake or 1-RTT packets.¶
+Prior to handshake completion, when few to none RTT samples have been +generated, it is possible that the probe timer expiration is due to an +incorrect RTT estimate at the client. To allow the client to improve its RTT +estimate, the new packet that it sends MUST be ack-eliciting. If Handshake +keys are available to the client, it MUST send a Handshake packet, and +otherwise it MUST send an Initial packet in a UDP datagram of at least 1200 +bytes.¶
+Initial packets and Handshake packets could be never acknowledged, but they are +removed from bytes in flight when the Initial and Handshake keys are discarded.¶
+When a PTO timer expires, a sender MUST send at least one ack-eliciting packet +in the packet number space as a probe, unless there is no data available to +send. An endpoint MAY send up to two full-sized datagrams containing +ack-eliciting packets, to avoid an expensive consecutive PTO expiration due +to a single lost datagram or transmit data from multiple packet number spaces.¶
+In addition to sending data in the packet number space for which the timer +expired, the sender SHOULD send ack-eliciting packets from other packet +number spaces with in-flight data, coalescing packets if possible.¶
+When the PTO timer expires, and there is new or previously sent unacknowledged +data, it MUST be sent.¶
+It is possible the sender has no new or previously-sent data to send. +As an example, consider the following sequence of events: new application data +is sent in a STREAM frame, deemed lost, then retransmitted in a new packet, +and then the original transmission is acknowledged. When there is no data to +send, the sender SHOULD send a PING or other ack-eliciting frame in a single +packet, re-arming the PTO timer.¶
+Alternatively, instead of sending an ack-eliciting packet, the sender MAY mark +any packets still in flight as lost. Doing so avoids sending an additional +packet, but increases the risk that loss is declared too aggressively, resulting +in an unnecessary rate reduction by the congestion controller.¶
+Consecutive PTO periods increase exponentially, and as a result, connection +recovery latency increases exponentially as packets continue to be dropped in +the network. Sending two packets on PTO expiration increases resilience to +packet drops, thus reducing the probability of consecutive PTO events.¶
+Probe packets sent on a PTO MUST be ack-eliciting. A probe packet SHOULD carry +new data when possible. A probe packet MAY carry retransmitted unacknowledged +data when new data is unavailable, when flow control does not permit new data to +be sent, or to opportunistically reduce loss recovery delay. Implementations +MAY use alternative strategies for determining the content of probe packets, +including sending new or retransmitted data based on the application's +priorities.¶
+When the PTO timer expires multiple times and new data cannot be sent, +implementations must choose between sending the same payload every time +or sending different payloads. Sending the same payload may be simpler +and ensures the highest priority frames arrive first. Sending different +payloads each time reduces the chances of spurious retransmission.¶
+Delivery or loss of packets in flight is established when an ACK frame is +received that newly acknowledges one or more packets.¶
+A PTO timer expiration event does not indicate packet loss and MUST NOT cause +prior unacknowledged packets to be marked as lost. When an acknowledgement +is received that newly acknowledges packets, loss detection proceeds as +dictated by packet and time threshold mechanisms; see Section 5.1.¶
+A Retry packet causes a client to send another Initial packet, effectively +restarting the connection process. A Retry packet indicates that the Initial +was received, but not processed. A Retry packet cannot be treated as an +acknowledgment, because it does not indicate that a packet was processed or +specify the packet number.¶
+Clients that receive a Retry packet reset congestion control and loss recovery +state, including resetting any pending timers. Other connection state, in +particular cryptographic handshake messages, is retained; see Section 17.2.5 of +[QUIC-TRANSPORT].¶
+The client MAY compute an RTT estimate to the server as the time period from +when the first Initial was sent to when a Retry or a Version Negotiation packet +is received. The client MAY use this value in place of its default for the +initial RTT estimate.¶
+When packet protection keys are discarded (see Section 4.9 of [QUIC-TLS]), all +packets that were sent with those keys can no longer be acknowledged because +their acknowledgements cannot be processed anymore. The sender MUST discard +all recovery state associated with those packets and MUST remove them from +the count of bytes in flight.¶
+Endpoints stop sending and receiving Initial packets once they start exchanging +Handshake packets (see Section 17.2.2.1 of [QUIC-TRANSPORT]). At this point, +recovery state for all in-flight Initial packets is discarded.¶
+When 0-RTT is rejected, recovery state for all in-flight 0-RTT packets is +discarded.¶
+If a server accepts 0-RTT, but does not buffer 0-RTT packets that arrive +before Initial packets, early 0-RTT packets will be declared lost, but that +is expected to be infrequent.¶
+It is expected that keys are discarded after packets encrypted with them would +be acknowledged or declared lost. Initial secrets however might be destroyed +sooner, as soon as handshake keys are available (see Section 4.9.1 of +[QUIC-TLS]).¶
+QUIC's congestion control is based on TCP NewReno [RFC6582]. NewReno is a +congestion window based congestion control. QUIC specifies the congestion +window in bytes rather than packets due to finer control and the ease of +appropriate byte counting [RFC3465].¶
+QUIC hosts MUST NOT send packets if they would increase bytes_in_flight (defined +in Appendix B.2) beyond the available congestion window, unless the +packet is a probe packet sent after a PTO timer expires, as described in +Section 5.2.¶
+Implementations MAY use other congestion control algorithms, such as +Cubic [RFC8312], and endpoints MAY use different algorithms from one another. +The signals QUIC provides for congestion control are generic and are designed +to support different algorithms.¶
+If a path has been verified to support ECN [RFC3168] [RFC8311], QUIC +treats a Congestion Experienced(CE) codepoint in the IP header as a signal of +congestion. This document specifies an endpoint's response when its peer +receives packets with the Congestion Experienced codepoint.¶
+QUIC begins every connection in slow start and exits slow start upon loss or +upon increase in the ECN-CE counter. QUIC re-enters slow start any time the +congestion window is less than ssthresh, which only occurs after persistent +congestion is declared. While in slow start, QUIC increases the congestion +window by the number of bytes acknowledged when each acknowledgment is +processed.¶
+Slow start exits to congestion avoidance. Congestion avoidance in NewReno +uses an additive increase multiplicative decrease (AIMD) approach that +increases the congestion window by one maximum packet size per +congestion window acknowledged. When a loss is detected, NewReno halves +the congestion window and sets the slow start threshold to the new +congestion window.¶
+Recovery is a period of time beginning with detection of a lost packet or an +increase in the ECN-CE counter. Because QUIC does not retransmit packets, +it defines the end of recovery as a packet sent after the start of recovery +being acknowledged. This is slightly different from TCP's definition of +recovery, which ends when the lost packet that started recovery is acknowledged.¶
+The recovery period limits congestion window reduction to once per round trip. +During recovery, the congestion window remains unchanged irrespective of new +losses or increases in the ECN-CE counter.¶
+During the handshake, some packet protection keys might not be +available when a packet arrives. In particular, Handshake and 0-RTT packets +cannot be processed until the Initial packets arrive, and 1-RTT packets +cannot be processed until the handshake completes. Endpoints MAY +ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might arrive before +the peer has packet protection keys to process those packets.¶
+Probe packets MUST NOT be blocked by the congestion controller. A sender MUST +however count these packets as being additionally in flight, since these packets +add network load without establishing packet loss. Note that sending probe +packets might cause the sender's bytes in flight to exceed the congestion window +until an acknowledgement is received that establishes loss or delivery of +packets.¶
+When an ACK frame is received that establishes loss of all in-flight packets +sent over a long enough period of time, the network is considered to be +experiencing persistent congestion. Commonly, this can be established by +consecutive PTOs, but since the PTO timer is reset when a new ack-eliciting +packet is sent, an explicit duration must be used to account for those cases +where PTOs do not occur or are substantially delayed. This duration is computed +as follows:¶
+
(smoothed_rtt + 4 * rttvar + max_ack_delay) *
kPersistentCongestionThreshold
-
-For example, assume:
-smoothed_rtt = 1 rttvar = 0 max_ack_delay = 0 kPersistentCongestionThreshold = 3
-If an ack-eliciting packet is sent at time = 0, the following scenario would illustrate persistent congestion:
-| t=0 | -Send Pkt #1 (App Data) | -
|---|---|
| t=1 | -Send Pkt #2 (PTO 1) | -
| t=3 | -Send Pkt #3 (PTO 2) | -
| t=7 | -Send Pkt #4 (PTO 3) | -
| t=8 | -Recv ACK of Pkt #4 | -
The first three packets are determined to be lost when the ACK of packet 4 is received at t=8. The congestion period is calculated as the time between the oldest and newest lost packets: (3 - 0) = 3. The duration for persistent congestion is equal to: (1 * kPersistentCongestionThreshold) = 3. Because the threshold was reached and because none of the packets between the oldest and the newest packets are acknowledged, the network is considered to have experienced persistent congestion.
-When persistent congestion is established, the sender’s congestion window MUST be reduced to the minimum congestion window (kMinimumWindow). This response of collapsing the congestion window on persistent congestion is functionally similar to a sender’s response on a Retransmission Timeout (RTO) in TCP [RFC5681] after Tail Loss Probes (TLP) [RACK].
-This document does not specify a pacer, but it is RECOMMENDED that a sender pace sending of all in-flight packets based on input from the congestion controller. For example, a pacer might distribute the congestion window over the smoothed RTT when used with a window-based controller, and a pacer might use the rate estimate of a rate-based controller.
-An implementation should take care to architect its congestion controller to work well with a pacer. For instance, a pacer might wrap the congestion controller and control the availability of the congestion window, or a pacer might pace out packets handed to it by the congestion controller. Timely delivery of ACK frames is important for efficient loss recovery. Packets containing only ACK frames should therefore not be paced, to avoid delaying their delivery to the peer.
-Sending multiple packets into the network without any delay between them creates a packet burst that might cause short-term congestion and losses. Implementations MUST either use pacing or limit such bursts to the initial congestion window, which is recommended to be the minimum of 10 * max_datagram_size and max(2* max_datagram_size, 14720)), where max_datagram_size is the current maximum size of a datagram for the connection, not including UDP or IP overhead.
-As an example of a well-known and publicly available implementation of a flow pacer, implementers are referred to the Fair Queue packet scheduler (fq qdisc) in Linux (3.11 onwards).
-When bytes in flight is smaller than the congestion window and sending is not pacing limited, the congestion window is under-utilized. When this occurs, the congestion window SHOULD NOT be increased in either slow start or congestion avoidance. This can happen due to insufficient application data or flow control credit.
-A sender MAY use the pipeACK method described in section 4.3 of [RFC7661] to determine if the congestion window is sufficiently utilized.
-A sender that paces packets (see Section 6.8) might delay sending packets and not fully utilize the congestion window due to this delay. A sender should not consider itself application limited if it would have fully utilized the congestion window without pacing delay.
-A sender MAY implement alternative mechanisms to update its congestion window after periods of under-utilization, such as those proposed for TCP in [RFC7661].
-Congestion control fundamentally involves the consumption of signals – both loss and ECN codepoints – from unauthenticated entities. On-path attackers can spoof or alter these signals. An attacker can cause endpoints to reduce their sending rate by dropping packets, or alter send rate by changing ECN codepoints.
-Packets that carry only ACK frames can be heuristically identified by observing packet size. Acknowledgement patterns may expose information about link characteristics or application behavior. Endpoints can use PADDING frames or bundle acknowledgments with other frames to reduce leaked information.
-A receiver can misreport ECN markings to alter the congestion response of a sender. Suppressing reports of ECN-CE markings could cause a sender to increase their send rate. This increase could result in congestion and loss.
-A sender MAY attempt to detect suppression of reports by marking occasional packets that they send with ECN-CE. If a packet marked with ECN-CE is not reported as having been marked when the packet is acknowledged, the sender SHOULD then disable ECN for that path.
-Reporting additional ECN-CE markings will cause a sender to reduce their sending rate, which is similar in effect to advertising reduced connection flow control limits and so no advantage is gained by doing so.
-Endpoints choose the congestion controller that they use. Though congestion controllers generally treat reports of ECN-CE markings as equivalent to loss [RFC8311], the exact response for each controller could be different. Failure to correctly respond to information about ECN markings is therefore difficult to detect.
-This document has no IANA actions. Yet.
-| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using TLS to Secure QUIC", Internet-Draft draft-ietf-quic-tls, November 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, November 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC8311] | --Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018. | -
We now describe an example implementation of the loss detection mechanisms described in Section 5.
-To correctly implement congestion control, a QUIC sender tracks every ack-eliciting packet until the packet is acknowledged or lost. It is expected that implementations will be able to access this information by packet number and crypto context and store the per-packet fields (Appendix A.1.1) for loss recovery and congestion control.
-After a packet is declared lost, the endpoint can track it for an amount of time comparable to the maximum expected packet reordering, such as 1 RTT. This allows for detection of spurious retransmissions.
-Sent packets are tracked for each packet number space, and ACK processing only applies to a single space.
-For example, assume:¶
+smoothed_rtt = 1 + rttvar = 0 + max_ack_delay = 0 + kPersistentCongestionThreshold = 3¶
+If an ack-eliciting packet is sent at time = 0, the following scenario would +illustrate persistent congestion:¶
+| t=0 | +Send Pkt #1 (App Data) | +
|---|---|
| t=1 | +Send Pkt #2 (PTO 1) | +
| t=3 | +Send Pkt #3 (PTO 2) | +
| t=7 | +Send Pkt #4 (PTO 3) | +
| t=8 | +Recv ACK of Pkt #4 | +
The first three packets are determined to be lost when the acknowlegement of +packet 4 is received at t=8. The congestion period is calculated as the time +between the oldest and newest lost packets: (3 - 0) = 3. The duration for +persistent congestion is equal to: (1 * kPersistentCongestionThreshold) = 3. +Because the threshold was reached and because none of the packets between the +oldest and the newest packets are acknowledged, the network is considered to +have experienced persistent congestion.¶
+When persistent congestion is established, the sender's congestion window MUST +be reduced to the minimum congestion window (kMinimumWindow). This response of +collapsing the congestion window on persistent congestion is functionally +similar to a sender's response on a Retransmission Timeout (RTO) in TCP +[RFC5681] after Tail Loss Probes (TLP) [RACK].¶
+This document does not specify a pacer, but it is RECOMMENDED that a sender pace +sending of all in-flight packets based on input from the congestion +controller. For example, a pacer might distribute the congestion window over +the smoothed RTT when used with a window-based controller, and a pacer might use +the rate estimate of a rate-based controller.¶
+An implementation should take care to architect its congestion controller to +work well with a pacer. For instance, a pacer might wrap the congestion +controller and control the availability of the congestion window, or a pacer +might pace out packets handed to it by the congestion controller. Timely +delivery of ACK frames is important for efficient loss recovery. Packets +containing only ACK frames should therefore not be paced, to avoid delaying +their delivery to the peer.¶
+Sending multiple packets into the network without any delay between them +creates a packet burst that might cause short-term congestion and losses. +Implementations MUST either use pacing or limit such bursts to the initial +congestion window, which is recommended to be the minimum of +10 * max_datagram_size and max(2* max_datagram_size, 14720)), where +max_datagram_size is the current maximum size of a datagram for the connection, +not including UDP or IP overhead.¶
+As an example of a well-known and publicly available implementation of a flow +pacer, implementers are referred to the Fair Queue packet scheduler (fq qdisc) +in Linux (3.11 onwards).¶
+When bytes in flight is smaller than the congestion window and sending is not +pacing limited, the congestion window is under-utilized. When this occurs, +the congestion window SHOULD NOT be increased in either slow start or +congestion avoidance. This can happen due to insufficient application data +or flow control credit.¶
+A sender MAY use the pipeACK method described in section 4.3 of [RFC7661] +to determine if the congestion window is sufficiently utilized.¶
+A sender that paces packets (see Section 6.8) might delay sending packets +and not fully utilize the congestion window due to this delay. A sender +should not consider itself application limited if it would have fully +utilized the congestion window without pacing delay.¶
+A sender MAY implement alternative mechanisms to update its congestion window +after periods of under-utilization, such as those proposed for TCP in +[RFC7661].¶
+Congestion control fundamentally involves the consumption of signals - both +loss and ECN codepoints - from unauthenticated entities. On-path attackers can +spoof or alter these signals. An attacker can cause endpoints to reduce their +sending rate by dropping packets, or alter send rate by changing ECN codepoints.¶
+Packets that carry only ACK frames can be heuristically identified by observing +packet size. Acknowledgement patterns may expose information about link +characteristics or application behavior. Endpoints can use PADDING frames or +bundle acknowledgments with other frames to reduce leaked information.¶
+A receiver can misreport ECN markings to alter the congestion response of a +sender. Suppressing reports of ECN-CE markings could cause a sender to +increase their send rate. This increase could result in congestion and loss.¶
+A sender MAY attempt to detect suppression of reports by marking occasional +packets that they send with ECN-CE. If a packet sent with ECN-CE is not +reported as having been CE marked when the packet is acknowledged, then the +sender SHOULD disable ECN for that path.¶
+Reporting additional ECN-CE markings will cause a sender to reduce their sending +rate, which is similar in effect to advertising reduced connection flow control +limits and so no advantage is gained by doing so.¶
+Endpoints choose the congestion controller that they use. Though congestion +controllers generally treat reports of ECN-CE markings as equivalent to loss +[RFC8311], the exact response for each controller could be different. Failure +to correctly respond to information about ECN markings is therefore difficult to +detect.¶
+This document has no IANA actions. Yet.¶
+We now describe an example implementation of the loss detection mechanisms +described in Section 5.¶
+To correctly implement congestion control, a QUIC sender tracks every +ack-eliciting packet until the packet is acknowledged or lost. +It is expected that implementations will be able to access this information by +packet number and crypto context and store the per-packet fields +(Appendix A.1.1) for loss recovery and congestion control.¶
+After a packet is declared lost, the endpoint can track it for an amount of +time comparable to the maximum expected packet reordering, such as 1 RTT. +This allows for detection of spurious retransmissions.¶
+Sent packets are tracked for each packet number space, and ACK +processing only applies to a single space.¶
+Constants used in loss recovery are based on a combination of RFCs, papers, and common practice. Some may need to be changed or negotiated in order to better suit a variety of environments.
- - -Constants used in loss recovery are based on a combination of RFCs, papers, and +common practice. Some may need to be changed or negotiated in order to better +suit a variety of environments.¶
+
enum kPacketNumberSpace {
Initial,
Handshake,
ApplicationData,
}
-
-Variables required to implement the congestion control mechanisms are described in this section.
- - -Variables required to implement the congestion control mechanisms +are described in this section.¶
+At the beginning of the connection, initialize the loss detection variables as follows:
+At the beginning of the connection, initialize the loss detection variables as +follows:¶
+
loss_detection_timer.reset()
pto_count = 0
@@ -1153,12 +2366,19 @@
largest_acked_packet[pn_space] = infinite
time_of_last_sent_ack_eliciting_packet[pn_space] = 0
loss_time[pn_space] = 0
-
-After a packet is sent, information about the packet is stored. The parameters to OnPacketSent are described in detail above in Appendix A.1.1.
-Pseudocode for OnPacketSent follows:
+¶ +After a packet is sent, information about the packet is stored. The parameters +to OnPacketSent are described in detail above in Appendix A.1.1.¶
+Pseudocode for OnPacketSent follows:¶
+
OnPacketSent(packet_number, pn_space, ack_eliciting,
in_flight, sent_bytes):
@@ -1174,12 +2394,18 @@
OnPacketSentCC(sent_bytes)
sent_packets[pn_space][packet_number].size = sent_bytes
SetLossDetectionTimer()
-
-When an ACK frame is received, it may newly acknowledge any number of packets.
-Pseudocode for OnAckReceived and UpdateRtt follow:
+¶ +When an ACK frame is received, it may newly acknowledge any number of packets.¶
+Pseudocode for OnAckReceived and UpdateRtt follow:¶
+OnAckReceived(ack, pn_space): if (largest_acked_packet[pn_space] == infinite): @@ -1237,25 +2463,46 @@-rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt) smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt -
When a packet is acknowledged for the first time, the following OnPacketAcked function is called. Note that a single ACK frame may newly acknowledge several packets. OnPacketAcked must be called once for each of these newly acknowledged packets.
-OnPacketAcked takes two parameters: acked_packet, which is the struct detailed in Appendix A.1.1, and the packet number space that this ACK frame was sent for.
-Pseudocode for OnPacketAcked follows:
+¶ +When a packet is acknowledged for the first time, the following OnPacketAcked +function is called. Note that a single ACK frame may newly acknowledge several +packets. OnPacketAcked must be called once for each of these newly acknowledged +packets.¶
+OnPacketAcked takes two parameters: acked_packet, which is the struct detailed +in Appendix A.1.1, and the packet number space that this ACK frame was +sent for.¶
+Pseudocode for OnPacketAcked follows:¶
+
OnPacketAcked(acked_packet, pn_space):
if (acked_packet.in_flight):
OnPacketAckedCC(acked_packet)
sent_packets[pn_space].remove(acked_packet.packet_number)
-
-QUIC loss detection uses a single timer for all timeout loss detection. The duration of the timer is based on the timer’s mode, which is set in the packet and timer events further below. The function SetLossDetectionTimer defined below shows how the single timer is set.
-This algorithm may result in the timer being set in the past, particularly if timers wake up late. Timers set in the past SHOULD fire immediately.
-Pseudocode for SetLossDetectionTimer follows:
+¶ +QUIC loss detection uses a single timer for all timeout loss detection. The +duration of the timer is based on the timer's mode, which is set in the packet +and timer events further below. The function SetLossDetectionTimer defined +below shows how the single timer is set.¶
+This algorithm may result in the timer being set in the past, particularly if +timers wake up late. Timers set in the past SHOULD fire immediately.¶
+Pseudocode for SetLossDetectionTimer follows:¶
+GetEarliestTimeAndSpace(times): time = times[Initial] @@ -1303,12 +2550,19 @@-sent_time, _ = GetEarliestTimeAndSpace( time_of_last_sent_ack_eliciting_packet) loss_detection_timer.update(sent_time + timeout) -
When the loss detection timer expires, the timer’s mode determines the action to be performed.
-Pseudocode for OnLossDetectionTimeout follows:
+¶ +When the loss detection timer expires, the timer's mode determines the action +to be performed.¶
+Pseudocode for OnLossDetectionTimeout follows:¶
+OnLossDetectionTimeout(): earliest_loss_time, pn_space = @@ -1336,12 +2590,19 @@-pto_count++ SetLossDetectionTimer() -
DetectLostPackets is called every time an ACK is received and operates on the sent_packets for that packet number space.
-Pseudocode for DetectLostPackets follows:
+¶ +DetectLostPackets is called every time an ACK is received and operates on +the sent_packets for that packet number space.¶
+Pseudocode for DetectLostPackets follows:¶
+DetectLostPackets(pn_space): assert(largest_acked_packet[pn_space] != infinite) @@ -1377,51 +2638,116 @@-// let it decide whether to retransmit immediately. if (!lost_packets.empty()): OnPacketsLost(lost_packets) -
We now describe an example implementation of the congestion controller described in Section 6.
-Constants used in congestion control are based on a combination of RFCs, papers, and common practice. Some may need to be changed or negotiated in order to better suit a variety of environments.
- - -We now describe an example implementation of the congestion controller described +in Section 6.¶
+Constants used in congestion control are based on a combination of RFCs, +papers, and common practice. Some may need to be changed or negotiated +in order to better suit a variety of environments.¶
+Variables required to implement the congestion control mechanisms are described in this section.
- - -Variables required to implement the congestion control mechanisms +are described in this section.¶
+At the beginning of the connection, initialize the congestion control variables as follows:
+At the beginning of the connection, initialize the congestion control +variables as follows:¶
+
congestion_window = kInitialWindow
bytes_in_flight = 0
@@ -1429,19 +2755,33 @@
ssthresh = infinite
for pn_space in [ Initial, Handshake, ApplicationData ]:
ecn_ce_counters[pn_space] = 0
-
-Whenever a packet is sent, and it contains non-ACK frames, the packet increases bytes_in_flight.
+¶ +Whenever a packet is sent, and it contains non-ACK frames, the packet +increases bytes_in_flight.¶
+
OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent
-
-Invoked from loss detection’s OnPacketAcked and is supplied with the acked_packet from sent_packets.
+¶ +Invoked from loss detection's OnPacketAcked and is supplied with the +acked_packet from sent_packets.¶
+
InCongestionRecovery(sent_time):
return sent_time <= congestion_recovery_start_time
@@ -1452,9 +2792,9 @@
if (InCongestionRecovery(acked_packet.time_sent)):
// Do not increase congestion window in recovery period.
return
- if (IsAppLimited()):
+ if (IsAppOrFlowControlLimited()):
// Do not increase congestion_window if application
- // limited.
+ // limited or flow control limited.
return
if (congestion_window < ssthresh):
// Slow start.
@@ -1463,11 +2803,19 @@
// Congestion avoidance.
congestion_window += max_datagram_size * acked_packet.size
/ congestion_window
-
-Invoked from ProcessECN and OnPacketsLost when a new congestion event is detected. May start a new recovery period and reduces the congestion window.
+¶ +Invoked from ProcessECN and OnPacketsLost when a new congestion event is +detected. May start a new recovery period and reduces the congestion +window.¶
+
CongestionEvent(sent_time):
// Start a new congestion event if packet was sent after the
@@ -1477,11 +2825,17 @@
congestion_window *= kLossReductionFactor
congestion_window = max(congestion_window, kMinimumWindow)
ssthresh = congestion_window
-
-Invoked when an ACK frame with an ECN section is received from the peer.
+¶ +Invoked when an ACK frame with an ECN section is received from the peer.¶
+
ProcessECN(ack, pn_space):
// If the ECN-CE counter reported by the peer has increased,
@@ -1489,11 +2843,17 @@
if (ack.ce_counter > ecn_ce_counters[pn_space]):
ecn_ce_counters[pn_space] = ack.ce_counter
CongestionEvent(sent_packets[ack.largest_acked].time_sent)
-
-Invoked from DetectLostPackets when packets are deemed lost.
+¶ +Invoked from DetectLostPackets when packets are deemed lost.¶
+
InPersistentCongestion(largest_lost_packet):
pto = smoothed_rtt + max(4 * rttvar, kGranularity) +
@@ -1515,284 +2875,485 @@
// Collapse congestion window if persistent congestion
if (InPersistentCongestion(largest_lost_packet)):
congestion_window = kMinimumWindow
-
-Issue and pull request numbers are listed with a leading octothorp.
-Issue and pull request numbers are listed with a leading octothorp.¶
+No significant changes.¶
+No significant changes.
-No significant changes.¶
+No significant changes.
-No significant changes.
-No significant changes.
-No significant changes.
-No significant changes.¶
+No significant changes.¶
+No significant changes.¶
+| QUIC | -M. Thomson, Ed. | -
| Internet-Draft | -Mozilla | -
| Intended status: Standards Track | -S. Turner, Ed. | -
| Expires: May 15, 2020 | -sn3rd | -
| - | November 12, 2019 | -
Using TLS to Secure QUIC
- draft-ietf-quic-tls-latest
This document describes how Transport Layer Security (TLS) is used to secure QUIC.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic.
-Working Group information can be found at https://github.com/quicwg; source code and issues list for this draft can be found at https://github.com/quicwg/base-drafts/labels/-tls.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
+/* Limit the width of the author address vcard, so names in right-to-left + script don't end up on the other side of the page. */ - - -| Internet-Draft | +Using TLS to Secure QUIC | +January 2020 | +
| Thomson & Turner | +Expires 16 July 2020 | +[Page] | +
This document describes how Transport Layer Security (TLS) is used to secure +QUIC.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +https://mailarchive.ietf.org/arch/search/?email_list=quic.¶
+Working Group information can be found at https://github.com/quicwg; source +code and issues list for this draft can be found at +https://github.com/quicwg/base-drafts/labels/-tls.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+This document describes how QUIC [QUIC-TRANSPORT] is secured using TLS +[TLS13].¶
+TLS 1.3 provides critical latency improvements for connection establishment over +previous versions. Absent packet loss, most new connections can be established +and secured within a single round trip; on subsequent connections between the +same client and server, the client can often send application data immediately, +that is, using a zero round trip setup.¶
+This document describes how TLS acts as a security component of QUIC.¶
+The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+This document uses the terminology established in [QUIC-TRANSPORT].¶
+For brevity, the acronym TLS is used to refer to TLS 1.3, though a newer version +could be used (see Section 4.2).¶
+TLS provides two endpoints with a way to establish a means of communication over +an untrusted medium (that is, the Internet) that ensures that messages they +exchange cannot be observed, modified, or forged.¶
+Internally, TLS is a layered protocol, with the structure shown in +Figure 1.¶
+
+-------------+------------+--------------+---------+
Handshake | | | Application | |
@@ -642,21 +1259,48 @@
| |
+---------------------------------------------------+
-Figure 1: TLS Layers
-Each Handshake layer message (e.g., Handshake, Alerts, and Application Data) is carried as a series of typed TLS records by the Record layer. Records are individually cryptographically protected and then transmitted over a reliable transport (typically TCP) which provides sequencing and guaranteed delivery.
-The TLS authenticated key exchange occurs between two endpoints: client and server. The client initiates the exchange and the server responds. If the key exchange completes successfully, both client and server will agree on a secret. TLS supports both pre-shared key (PSK) and Diffie-Hellman over either finite fields or elliptic curves ((EC)DHE) key exchanges. PSK is the basis for 0-RTT; the latter provides perfect forward secrecy (PFS) when the (EC)DHE keys are destroyed.
-After completing the TLS handshake, the client will have learned and authenticated an identity for the server and the server is optionally able to learn and authenticate an identity for the client. TLS supports X.509 [RFC5280] certificate-based authentication for both server and client.
-The TLS key exchange is resistant to tampering by attackers and it produces shared secrets that cannot be controlled by either participating peer.
-TLS provides two basic handshake modes of interest to QUIC:
- - +Each Handshake layer message (e.g., Handshake, Alerts, and Application Data) is +carried as a series of typed TLS records by the Record layer. Records are +individually cryptographically protected and then transmitted over a reliable +transport (typically TCP) which provides sequencing and guaranteed delivery.¶
+The TLS authenticated key exchange occurs between two endpoints: client and +server. The client initiates the exchange and the server responds. If the key +exchange completes successfully, both client and server will agree on a secret. +TLS supports both pre-shared key (PSK) and Diffie-Hellman over either finite +fields or elliptic curves ((EC)DHE) key exchanges. PSK is the basis for 0-RTT; +the latter provides perfect forward secrecy (PFS) when the (EC)DHE keys are +destroyed.¶
+After completing the TLS handshake, the client will have learned and +authenticated an identity for the server and the server is optionally able to +learn and authenticate an identity for the client. TLS supports X.509 +[RFC5280] certificate-based authentication for both server and client.¶
+The TLS key exchange is resistant to tampering by attackers and it produces +shared secrets that cannot be controlled by either participating peer.¶
+TLS provides two basic handshake modes of interest to QUIC:¶
A simplified TLS handshake with 0-RTT application data is shown in Figure 2. Note that this omits the EndOfEarlyData message, which is not used in QUIC (see Section 8.3). Likewise, neither ChangeCipherSpec nor KeyUpdate messages are used by QUIC; ChangeCipherSpec is redundant in TLS 1.3 and QUIC has defined its own key update mechanism Section 6.
- - +A simplified TLS handshake with 0-RTT application data is shown in Figure 2. +Note that this omits the EndOfEarlyData message, which is not used in QUIC (see +Section 8.3). Likewise, neither ChangeCipherSpec nor KeyUpdate messages are +used by QUIC; ChangeCipherSpec is redundant in TLS 1.3 and QUIC has defined its +own key update mechanism Section 6.¶
+
Client Server
@@ -675,24 +1319,46 @@
[] Indicates messages protected using Application Data
(1-RTT) Keys
-Figure 2: TLS Handshake with 0-RTT
-Data is protected using a number of encryption levels:
- - +Data is protected using a number of encryption levels:¶
Application Data may appear only in the Early Data and Application Data levels. Handshake and Alert messages may appear in any level.
-The 0-RTT handshake is only possible if the client and server have previously communicated. In the 1-RTT handshake, the client is unable to send protected Application Data until it has received all of the Handshake messages sent by the server.
-QUIC [QUIC-TRANSPORT] assumes responsibility for the confidentiality and integrity protection of packets. For this it uses keys derived from a TLS handshake [TLS13], but instead of carrying TLS records over QUIC (as with TCP), TLS Handshake and Alert messages are carried directly over the QUIC transport, which takes over the responsibilities of the TLS record layer, as shown in Figure 3.
- - +Application Data may appear only in the Early Data and Application Data +levels. Handshake and Alert messages may appear in any level.¶
+The 0-RTT handshake is only possible if the client and server have previously +communicated. In the 1-RTT handshake, the client is unable to send protected +Application Data until it has received all of the Handshake messages sent by the +server.¶
+QUIC [QUIC-TRANSPORT] assumes responsibility for the confidentiality and +integrity protection of packets. For this it uses keys derived from a TLS +handshake [TLS13], but instead of carrying TLS records over QUIC (as with +TCP), TLS Handshake and Alert messages are carried directly over the QUIC +transport, which takes over the responsibilities of the TLS record layer, as +shown in Figure 3.¶
++--------------+--------------+ +-------------+ | TLS | TLS | | QUIC | @@ -709,19 +1375,32 @@-| | +---------------------------------------------+
Figure 3: QUIC Layers
-QUIC also relies on TLS for authentication and negotiation of parameters that are critical to security and performance.
-Rather than a strict layering, these two protocols cooperate: QUIC uses the TLS handshake; TLS uses the reliability, ordered delivery, and record layer provided by QUIC.
-At a high level, there are two main interactions between the TLS and QUIC components:
- - +QUIC also relies on TLS for authentication and negotiation of parameters that +are critical to security and performance.¶
+Rather than a strict layering, these two protocols cooperate: QUIC uses the TLS +handshake; TLS uses the reliability, ordered delivery, and record layer provided +by QUIC.¶
+At a high level, there are two main interactions between the TLS and QUIC +components:¶
Figure 4 shows these interactions in more detail, with the QUIC packet protection being called out specially.
- - +Figure 4 shows these interactions in more detail, with the QUIC packet +protection being called out specially.¶
++------------+ +------------+ | |<---- Handshake Messages ----->| | @@ -740,132 +1419,280 @@-| Protection | +------------+
Figure 4: QUIC and TLS Interactions
-Unlike TLS over TCP, QUIC applications which want to send data do not send it through TLS “application_data” records. Rather, they send it as QUIC STREAM frames or other frame types which are then carried in QUIC packets.
-QUIC carries TLS handshake data in CRYPTO frames, each of which consists of a contiguous block of handshake data identified by an offset and length. Those frames are packaged into QUIC packets and encrypted under the current TLS encryption level. As with TLS over TCP, once TLS handshake data has been delivered to QUIC, it is QUIC’s responsibility to deliver it reliably. Each chunk of data that is produced by TLS is associated with the set of keys that TLS is currently using. If QUIC needs to retransmit that data, it MUST use the same keys even if TLS has already updated to newer keys.
-One important difference between TLS records (used with TCP) and QUIC CRYPTO frames is that in QUIC multiple frames may appear in the same QUIC packet as long as they are associated with the same encryption level. For instance, an implementation might bundle a Handshake message and an ACK for some Handshake data into the same packet.
-Some frames are prohibited in different encryption levels, others cannot be sent. The rules here generalize those of TLS, in that frames associated with establishing the connection can usually appear at any encryption level, whereas those associated with transferring data can only appear in the 0-RTT and 1-RTT encryption levels:
- - +Unlike TLS over TCP, QUIC applications which want to send data do not send it +through TLS "application_data" records. Rather, they send it as QUIC STREAM +frames or other frame types which are then carried in QUIC packets.¶
+QUIC carries TLS handshake data in CRYPTO frames, each of which consists of a +contiguous block of handshake data identified by an offset and length. Those +frames are packaged into QUIC packets and encrypted under the current TLS +encryption level. As with TLS over TCP, once TLS handshake data has been +delivered to QUIC, it is QUIC's responsibility to deliver it reliably. Each +chunk of data that is produced by TLS is associated with the set of keys that +TLS is currently using. If QUIC needs to retransmit that data, it MUST use the +same keys even if TLS has already updated to newer keys.¶
+One important difference between TLS records (used with TCP) and QUIC CRYPTO +frames is that in QUIC multiple frames may appear in the same QUIC packet as +long as they are associated with the same encryption level. For instance, an +implementation might bundle a Handshake message and an ACK for some Handshake +data into the same packet.¶
+Some frames are prohibited in different encryption levels, others cannot be +sent. The rules here generalize those of TLS, in that frames associated with +establishing the connection can usually appear at any encryption level, whereas +those associated with transferring data can only appear in the 0-RTT and 1-RTT +encryption levels:¶
Note that it is not possible to send the following frames in 0-RTT for various reasons: ACK, CRYPTO, NEW_TOKEN, PATH_RESPONSE, and RETIRE_CONNECTION_ID.
-Because packets could be reordered on the wire, QUIC uses the packet type to indicate which level a given packet was encrypted under, as shown in Table 1. When multiple packets of different encryption levels need to be sent, endpoints SHOULD use coalesced packets to send them in the same UDP datagram.
- - -| Packet Type | -Encryption Level | -PN Space | -
|---|---|---|
| Initial | -Initial secrets | -Initial | -
| 0-RTT Protected | -0-RTT | -0/1-RTT | -
| Handshake | -Handshake | -Handshake | -
| Retry | -N/A | -N/A | -
| Version Negotiation | -N/A | -N/A | -
| Short Header | -1-RTT | -0/1-RTT | -
Section 17 of [QUIC-TRANSPORT] shows how packets at the various encryption levels fit into the handshake process.
-As shown in Figure 4, the interface from QUIC to TLS consists of four primary functions:
- - +Note that it is not possible to send the following frames in 0-RTT for various +reasons: ACK, CRYPTO, NEW_TOKEN, PATH_RESPONSE, and RETIRE_CONNECTION_ID.¶
+Because packets could be reordered on the wire, QUIC uses the packet type to +indicate which level a given packet was encrypted under, as shown in +Table 1. When multiple packets of different encryption levels +need to be sent, endpoints SHOULD use coalesced packets to send them in the same +UDP datagram.¶
+| Packet Type | +Encryption Level | +PN Space | +
|---|---|---|
| Initial | +Initial secrets | +Initial | +
| 0-RTT Protected | +0-RTT | +0/1-RTT | +
| Handshake | +Handshake | +Handshake | +
| Retry | +N/A | +N/A | +
| Version Negotiation | +N/A | +N/A | +
| Short Header | +1-RTT | +0/1-RTT | +
Section 17 of [QUIC-TRANSPORT] shows how packets at the various encryption +levels fit into the handshake process.¶
+As shown in Figure 4, the interface from QUIC to TLS consists of four +primary functions:¶
Additional functions might be needed to configure TLS.
-In this document, the TLS handshake is considered complete when the TLS stack has reported that the handshake is complete. This happens when the TLS stack has both sent a Finished message and verified the peer’s Finished message. Verifying the peer’s Finished provides the endpoints with an assurance that previous handshake messages have not been modified. Note that the handshake does not complete at both endpoints simultaneously. Consequently, any requirement that is based on the completion of the handshake depends on the perspective of the endpoint in question.
-In this document, the TLS handshake is considered confirmed at an endpoint when the following two conditions are met: the handshake is complete, and the endpoint has received an acknowledgment for a packet sent with 1-RTT keys. This second condition can be implemented by recording the lowest packet number sent with 1-RTT keys, and the highest value of the Largest Acknowledged field in any received 1-RTT ACK frame: once the latter is higher than or equal to the former, the handshake is confirmed.
-In order to drive the handshake, TLS depends on being able to send and receive handshake messages. There are two basic functions on this interface: one where QUIC requests handshake messages and one where QUIC provides handshake packets.
-Before starting the handshake QUIC provides TLS with the transport parameters (see Section 8.2) that it wishes to carry.
-A QUIC client starts TLS by requesting TLS handshake bytes from TLS. The client acquires handshake bytes before sending its first packet. A QUIC server starts the process by providing TLS with the client’s handshake bytes.
-At any time, the TLS stack at an endpoint will have a current sending encryption level and receiving encryption level. Each encryption level is associated with a different flow of bytes, which is reliably transmitted to the peer in CRYPTO frames. When TLS provides handshake bytes to be sent, they are appended to the current flow and any packet that includes the CRYPTO frame is protected using keys from the corresponding encryption level.
-QUIC takes the unprotected content of TLS handshake records as the content of CRYPTO frames. TLS record protection is not used by QUIC. QUIC assembles CRYPTO frames into QUIC packets, which are protected using QUIC packet protection.
-QUIC is only capable of conveying TLS handshake records in CRYPTO frames. TLS alerts are turned into QUIC CONNECTION_CLOSE error codes; see Section 4.9. TLS application data and other message types cannot be carried by QUIC at any encryption level and is an error if they are received from the TLS stack.
-When an endpoint receives a QUIC packet containing a CRYPTO frame from the network, it proceeds as follows:
- - +Additional functions might be needed to configure TLS.¶
+In this document, the TLS handshake is considered complete when the TLS stack +has reported that the handshake is complete. This happens when the TLS stack +has both sent a Finished message and verified the peer's Finished message. +Verifying the peer's Finished provides the endpoints with an assurance that +previous handshake messages have not been modified. Note that the handshake +does not complete at both endpoints simultaneously. Consequently, any +requirement that is based on the completion of the handshake depends on the +perspective of the endpoint in question.¶
+In this document, the TLS handshake is considered confirmed at an endpoint when +the following two conditions are met: the handshake is complete, and the +endpoint has received an acknowledgment for a packet sent with 1-RTT keys. +This second condition can be implemented by recording the lowest packet number +sent with 1-RTT keys, and the highest value of the Largest Acknowledged field +in any received 1-RTT ACK frame: once the latter is higher than or equal to the +former, the handshake is confirmed.¶
+In order to drive the handshake, TLS depends on being able to send and receive +handshake messages. There are two basic functions on this interface: one where +QUIC requests handshake messages and one where QUIC provides handshake packets.¶
+Before starting the handshake QUIC provides TLS with the transport parameters +(see Section 8.2) that it wishes to carry.¶
+A QUIC client starts TLS by requesting TLS handshake bytes from TLS. The client +acquires handshake bytes before sending its first packet. A QUIC server starts +the process by providing TLS with the client's handshake bytes.¶
+At any time, the TLS stack at an endpoint will have a current sending encryption +level and receiving encryption level. Each encryption level is associated with a +different flow of bytes, which is reliably transmitted to the peer in CRYPTO +frames. When TLS provides handshake bytes to be sent, they are appended to the +current flow and any packet that includes the CRYPTO frame is protected using +keys from the corresponding encryption level.¶
+QUIC takes the unprotected content of TLS handshake records as the content of +CRYPTO frames. TLS record protection is not used by QUIC. QUIC assembles +CRYPTO frames into QUIC packets, which are protected using QUIC packet +protection.¶
+QUIC is only capable of conveying TLS handshake records in CRYPTO frames. TLS +alerts are turned into QUIC CONNECTION_CLOSE error codes; see Section 4.9. +TLS application data and other message types cannot be carried by QUIC at any +encryption level and is an error if they are received from the TLS stack.¶
+When an endpoint receives a QUIC packet containing a CRYPTO frame from the +network, it proceeds as follows:¶
Each time that TLS is provided with new data, new handshake bytes are requested from TLS. TLS might not provide any bytes if the handshake messages it has received are incomplete or it has no data to send.
-Once the TLS handshake is complete, this is indicated to QUIC along with any final handshake bytes that TLS needs to send. TLS also provides QUIC with the transport parameters that the peer advertised during the handshake.
-Once the handshake is complete, TLS becomes passive. TLS can still receive data from its peer and respond in kind, but it will not need to send more data unless specifically requested - either by an application or QUIC. One reason to send data is that the server might wish to provide additional or updated session tickets to a client.
-When the handshake is complete, QUIC only needs to provide TLS with any data that arrives in CRYPTO streams. In the same way that is done during the handshake, new data is requested from TLS after providing received data.
-As keys for new encryption levels become available, TLS provides QUIC with those keys. Separately, as keys at a given encryption level become available to TLS, TLS indicates to QUIC that reading or writing keys at that encryption level are available. These events are not asynchronous; they always occur immediately after TLS is provided with new handshake bytes, or after TLS produces handshake bytes.
-TLS provides QUIC with three items as a new encryption level becomes available:
- - +Each time that TLS is provided with new data, new handshake bytes are requested +from TLS. TLS might not provide any bytes if the handshake messages it has +received are incomplete or it has no data to send.¶
+Once the TLS handshake is complete, this is indicated to QUIC along with any +final handshake bytes that TLS needs to send. TLS also provides QUIC with the +transport parameters that the peer advertised during the handshake.¶
+Once the handshake is complete, TLS becomes passive. TLS can still receive data +from its peer and respond in kind, but it will not need to send more data unless +specifically requested - either by an application or QUIC. One reason to send +data is that the server might wish to provide additional or updated session +tickets to a client.¶
+When the handshake is complete, QUIC only needs to provide TLS with any data +that arrives in CRYPTO streams. In the same way that is done during the +handshake, new data is requested from TLS after providing received data.¶
+As keys for new encryption levels become available, TLS provides QUIC with those +keys. Separately, as keys at a given encryption level become available to TLS, +TLS indicates to QUIC that reading or writing keys at that encryption level are +available. These events are not asynchronous; they always occur immediately +after TLS is provided with new handshake bytes, or after TLS produces handshake +bytes.¶
+TLS provides QUIC with three items as a new encryption level becomes available:¶
These values are based on the values that TLS negotiates and are used by QUIC to generate packet and header protection keys (see Section 5 and Section 5.4).
-If 0-RTT is possible, it is ready after the client sends a TLS ClientHello message or the server receives that message. After providing a QUIC client with the first handshake bytes, the TLS stack might signal the change to 0-RTT keys. On the server, after receiving handshake bytes that contain a ClientHello message, a TLS server might signal that 0-RTT keys are available.
-Although TLS only uses one encryption level at a time, QUIC may use more than one level. For instance, after sending its Finished message (using a CRYPTO frame at the Handshake encryption level) an endpoint can send STREAM data (in 1-RTT encryption). If the Finished message is lost, the endpoint uses the Handshake encryption level to retransmit the lost message. Reordering or loss of packets can mean that QUIC will need to handle packets at multiple encryption levels. During the handshake, this means potentially handling packets at higher and lower encryption levels than the current encryption level used by TLS.
-In particular, server implementations need to be able to read packets at the Handshake encryption level at the same time as the 0-RTT encryption level. A client could interleave ACK frames that are protected with Handshake keys with 0-RTT data and the server needs to process those acknowledgments in order to detect lost Handshake packets.
-QUIC also needs access to keys that might not ordinarily be available to a TLS implementation. For instance, a client might need to acknowledge Handshake packets before it is ready to send CRYPTO frames at that encryption level. TLS therefore needs to provide keys to QUIC before it might produce them for its own use.
-Figure 5 summarizes the exchange between QUIC and TLS for both client and server. Each arrow is tagged with the encryption level used for that transmission.
- - +These values are based on the values that TLS negotiates and are used by QUIC to +generate packet and header protection keys (see Section 5 and +Section 5.4).¶
+If 0-RTT is possible, it is ready after the client sends a TLS ClientHello +message or the server receives that message. After providing a QUIC client with +the first handshake bytes, the TLS stack might signal the change to 0-RTT +keys. On the server, after receiving handshake bytes that contain a ClientHello +message, a TLS server might signal that 0-RTT keys are available.¶
+Although TLS only uses one encryption level at a time, QUIC may use more than +one level. For instance, after sending its Finished message (using a CRYPTO +frame at the Handshake encryption level) an endpoint can send STREAM data (in +1-RTT encryption). If the Finished message is lost, the endpoint uses the +Handshake encryption level to retransmit the lost message. Reordering or loss +of packets can mean that QUIC will need to handle packets at multiple encryption +levels. During the handshake, this means potentially handling packets at higher +and lower encryption levels than the current encryption level used by TLS.¶
+In particular, server implementations need to be able to read packets at the +Handshake encryption level at the same time as the 0-RTT encryption level. A +client could interleave ACK frames that are protected with Handshake keys with +0-RTT data and the server needs to process those acknowledgments in order to +detect lost Handshake packets.¶
+QUIC also needs access to keys that might not ordinarily be available to a TLS +implementation. For instance, a client might need to acknowledge Handshake +packets before it is ready to send CRYPTO frames at that encryption level. TLS +therefore needs to provide keys to QUIC before it might produce them for its own +use.¶
+Figure 5 summarizes the exchange between QUIC and TLS for both +client and server. Each arrow is tagged with the encryption level used for that +transmission.¶
+Client Server @@ -897,95 +1724,311 @@-<--------------- 1-RTT Handshake Received
Figure 5: Interaction Summary between QUIC and TLS
-Figure 5 shows the multiple packets that form a single “flight” of messages being processed individually, to show what incoming messages trigger different actions. New handshake messages are requested after all incoming packets have been processed. This process might vary depending on how QUIC implementations and the packets they receive are structured.
-This document describes how TLS 1.3 [TLS13] is used with QUIC.
-In practice, the TLS handshake will negotiate a version of TLS to use. This could result in a newer version of TLS than 1.3 being negotiated if both endpoints support that version. This is acceptable provided that the features of TLS 1.3 that are used by QUIC are supported by the newer version.
-A badly configured TLS implementation could negotiate TLS 1.2 or another older version of TLS. An endpoint MUST terminate the connection if a version of TLS older than 1.3 is negotiated.
-The first Initial packet from a client contains the start or all of its first cryptographic handshake message, which for TLS is the ClientHello. Servers might need to parse the entire ClientHello (e.g., to access extensions such as Server Name Identification (SNI) or Application Layer Protocol Negotiation (ALPN)) in order to decide whether to accept the new incoming QUIC connection. If the ClientHello spans multiple Initial packets, such servers would need to buffer the first received fragments, which could consume excessive resources if the client’s address has not yet been validated. To avoid this, servers MAY use the Retry feature (see Section 8.1 of [QUIC-TRANSPORT]) to only buffer partial ClientHello messages from clients with a validated address.
-QUIC packet and framing add at least 36 bytes of overhead to the ClientHello message. That overhead increases if the client chooses a connection ID without zero length. Overheads also do not include the token or a connection ID longer than 8 bytes, both of which might be required if a server sends a Retry packet.
-A typical TLS ClientHello can easily fit into a 1200 byte packet. However, in addition to the overheads added by QUIC, there are several variables that could cause this limit to be exceeded. Large session tickets, multiple or large key shares, and long lists of supported ciphers, signature algorithms, versions, QUIC transport parameters, and other negotiable parameters and extensions could cause this message to grow.
-For servers, in addition to connection IDs and tokens, the size of TLS session tickets can have an effect on a client’s ability to connect efficiently. Minimizing the size of these values increases the probability that clients can use them and still fit their ClientHello message in their first Initial packet.
-The TLS implementation does not need to ensure that the ClientHello is sufficiently large. QUIC PADDING frames are added to increase the size of the packet as necessary.
-The requirements for authentication depend on the application protocol that is in use. TLS provides server authentication and permits the server to request client authentication.
-A client MUST authenticate the identity of the server. This typically involves verification that the identity of the server is included in a certificate and that the certificate is issued by a trusted entity (see for example [RFC2818]).
-A server MAY request that the client authenticate during the handshake. A server MAY refuse a connection if the client is unable to authenticate when requested. The requirements for client authentication vary based on application protocol and deployment.
-A server MUST NOT use post-handshake client authentication (as defined in Section 4.6.2 of [TLS13]), because the multiplexing offered by QUIC prevents clients from correlating the certificate request with the application-level event that triggered it (see [HTTP2-TLS13]). More specifically, servers MUST NOT send post-handshake TLS CertificateRequest messages and clients MUST treat receipt of such messages as a connection error of type PROTOCOL_VIOLATION.
-To communicate their willingness to process 0-RTT data, servers send a NewSessionTicket message that contains the “early_data” extension with a max_early_data_size of 0xffffffff; the amount of data which the client can send in 0-RTT is controlled by the “initial_max_data” transport parameter supplied by the server. Servers MUST NOT send the “early_data” extension with a max_early_data_size set to any value other than 0xffffffff. A client MUST treat receipt of a NewSessionTicket that contains an “early_data” extension with any other value as a connection error of type PROTOCOL_VIOLATION.
-A client that wishes to send 0-RTT packets uses the “early_data” extension in the ClientHello message of a subsequent handshake (see Section 4.2.10 of [TLS13]). It then sends the application data in 0-RTT packets.
-A server accepts 0-RTT by sending an early_data extension in the EncryptedExtensions (see Section 4.2.10 of [TLS13]). The server then processes and acknowledges the 0-RTT packets that it receives.
-A server rejects 0-RTT by sending the EncryptedExtensions without an early_data extension. A server will always reject 0-RTT if it sends a TLS HelloRetryRequest. When rejecting 0-RTT, a server MUST NOT process any 0-RTT packets, even if it could. When 0-RTT was rejected, a client SHOULD treat receipt of an acknowledgement for a 0-RTT packet as a connection error of type PROTOCOL_VIOLATION, if it is able to detect the condition.
-When 0-RTT is rejected, all connection characteristics that the client assumed might be incorrect. This includes the choice of application protocol, transport parameters, and any application configuration. The client therefore MUST reset the state of all streams, including application state bound to those streams.
-A client MAY attempt to send 0-RTT again if it receives a Retry or Version Negotiation packet. These packets do not signify rejection of 0-RTT.
-When a server receives a ClientHello with the “early_data” extension, it has to decide whether to accept or reject early data from the client. Some of this decision is made by the TLS stack (e.g., checking that the cipher suite being resumed was included in the ClientHello; see Section 4.2.10 of [TLS13]). Even when the TLS stack has no reason to reject early data, the QUIC stack or the application protocol using QUIC might reject early data because the configuration of the transport or application associated with the resumed session is not compatible with the server’s current configuration.
-QUIC requires additional transport state to be associated with a 0-RTT session ticket. One common way to implement this is using stateless session tickets and storing this state in the session ticket. Application protocols that use QUIC might have similar requirements regarding associating or storing state. This associated state is used for deciding whether early data must be rejected. For example, HTTP/3 ([QUIC-HTTP]) settings determine how early data from the client is interpreted. Other applications using QUIC could have different requirements for determining whether to accept or reject early data.
-In TLS over TCP, the HelloRetryRequest feature (see Section 4.1.4 of [TLS13]) can be used to correct a client’s incorrect KeyShare extension as well as for a stateless round-trip check. From the perspective of QUIC, this just looks like additional messages carried in the Initial encryption level. Although it is in principle possible to use this feature for address verification in QUIC, QUIC implementations SHOULD instead use the Retry feature (see Section 8.1 of [QUIC-TRANSPORT]). HelloRetryRequest is still used to request key shares.
-If TLS experiences an error, it generates an appropriate alert as defined in Section 6 of [TLS13].
-A TLS alert is turned into a QUIC connection error by converting the one-byte alert description into a QUIC error code. The alert description is added to 0x100 to produce a QUIC error code from the range reserved for CRYPTO_ERROR. The resulting value is sent in a QUIC CONNECTION_CLOSE frame.
-The alert level of all TLS alerts is “fatal”; a TLS stack MUST NOT generate alerts at the “warning” level.
-After QUIC moves to a new encryption level, packet protection keys for previous encryption levels can be discarded. This occurs several times during the handshake, as well as when keys are updated; see Section 6.
-Packet protection keys are not discarded immediately when new keys are available. If packets from a lower encryption level contain CRYPTO frames, frames that retransmit that data MUST be sent at the same encryption level. Similarly, an endpoint generates acknowledgements for packets at the same encryption level as the packet being acknowledged. Thus, it is possible that keys for a lower encryption level are needed for a short time after keys for a newer encryption level are available.
-An endpoint cannot discard keys for a given encryption level unless it has both received and acknowledged all CRYPTO frames for that encryption level and when all CRYPTO frames for that encryption level have been acknowledged by its peer. However, this does not guarantee that no further packets will need to be received or sent at that encryption level because a peer might not have received all the acknowledgements necessary to reach the same state.
-Though an endpoint might retain older keys, new data MUST be sent at the highest currently-available encryption level. Only ACK frames and retransmissions of data in CRYPTO frames are sent at a previous encryption level. These packets MAY also include PADDING frames.
-Packets protected with Initial secrets (Section 5.2) are not authenticated, meaning that an attacker could spoof packets with the intent to disrupt a connection. To limit these attacks, Initial packet protection keys can be discarded more aggressively than other keys.
-The successful use of Handshake packets indicates that no more Initial packets need to be exchanged, as these keys can only be produced after receiving all CRYPTO frames from Initial packets. Thus, a client MUST discard Initial keys when it first sends a Handshake packet and a server MUST discard Initial keys when it first successfully processes a Handshake packet. Endpoints MUST NOT send Initial packets after this point.
-This results in abandoning loss recovery state for the Initial encryption level and ignoring any outstanding Initial packets.
-An endpoint MUST NOT discard its handshake keys until the TLS handshake is confirmed (Section 4.1.2). An endpoint SHOULD discard its handshake keys as soon as it has confirmed the handshake. Most application protocols will send data after the handshake, resulting in acknowledgements that allow both endpoints to discard their handshake keys promptly. Endpoints that do not have reason to send immediately after completing the handshake MAY send ack-eliciting frames, such as PING, which will cause the handshake to be confirmed when they are acknowledged.
-0-RTT and 1-RTT packets share the same packet number space, and clients do not send 0-RTT packets after sending a 1-RTT packet (Section 5.6).
-Therefore, a client SHOULD discard 0-RTT keys as soon as it installs 1-RTT keys, since they have no use after that moment.
-Additionally, a server MAY discard 0-RTT keys as soon as it receives a 1-RTT packet. However, due to packet reordering, a 0-RTT packet could arrive after a 1-RTT packet. Servers MAY temporarily retain 0-RTT keys to allow decrypting reordered packets without requiring their contents to be retransmitted with 1-RTT keys. After receiving a 1-RTT packet, servers MUST discard 0-RTT keys within a short time; the RECOMMENDED time period is three times the Probe Timeout (PTO, see [QUIC-RECOVERY]). A server MAY discard 0-RTT keys earlier if it determines that it has received all 0-RTT packets, which can be done by keeping track of missing packet numbers.
-As with TLS over TCP, QUIC protects packets with keys derived from the TLS handshake, using the AEAD algorithm negotiated by TLS.
-QUIC derives packet protection keys in the same way that TLS derives record protection keys.
-Each encryption level has separate secret values for protection of packets sent in each direction. These traffic secrets are derived by TLS (see Section 7.1 of [TLS13]) and are used by QUIC for all encryption levels except the Initial encryption level. The secrets for the Initial encryption level are computed based on the client’s initial Destination Connection ID, as described in Section 5.2.
-The keys used for packet protection are computed from the TLS secrets using the KDF provided by TLS. In TLS 1.3, the HKDF-Expand-Label function described in Section 7.1 of [TLS13] is used, using the hash function from the negotiated cipher suite. Other versions of TLS MUST provide a similar function in order to be used with QUIC.
-The current encryption level secret and the label “quic key” are input to the KDF to produce the AEAD key; the label “quic iv” is used to derive the IV; see Section 5.3. The header protection key uses the “quic hp” label; see Section 5.4. Using these labels provides key separation between QUIC and TLS; see Section 9.5.
-The KDF used for initial secrets is always the HKDF-Expand-Label function from TLS 1.3 (see Section 5.2).
-Initial packets are protected with a secret derived from the Destination Connection ID field from the client’s Initial packet. Specifically:
+Figure 5 shows the multiple packets that form a single "flight" of +messages being processed individually, to show what incoming messages trigger +different actions. New handshake messages are requested after all incoming +packets have been processed. This process might vary depending on how QUIC +implementations and the packets they receive are structured.¶
+This document describes how TLS 1.3 [TLS13] is used with QUIC.¶
+In practice, the TLS handshake will negotiate a version of TLS to use. This +could result in a newer version of TLS than 1.3 being negotiated if both +endpoints support that version. This is acceptable provided that the features +of TLS 1.3 that are used by QUIC are supported by the newer version.¶
+A badly configured TLS implementation could negotiate TLS 1.2 or another older +version of TLS. An endpoint MUST terminate the connection if a version of TLS +older than 1.3 is negotiated.¶
+The first Initial packet from a client contains the start or all of its first +cryptographic handshake message, which for TLS is the ClientHello. Servers +might need to parse the entire ClientHello (e.g., to access extensions such as +Server Name Identification (SNI) or Application Layer Protocol Negotiation +(ALPN)) in order to decide whether to accept the new incoming QUIC connection. +If the ClientHello spans multiple Initial packets, such servers would need to +buffer the first received fragments, which could consume excessive resources if +the client's address has not yet been validated. To avoid this, servers MAY +use the Retry feature (see Section 8.1 of [QUIC-TRANSPORT]) to only buffer +partial ClientHello messages from clients with a validated address.¶
+QUIC packet and framing add at least 36 bytes of overhead to the ClientHello +message. That overhead increases if the client chooses a connection ID without +zero length. Overheads also do not include the token or a connection ID longer +than 8 bytes, both of which might be required if a server sends a Retry packet.¶
+A typical TLS ClientHello can easily fit into a 1200 byte packet. However, in +addition to the overheads added by QUIC, there are several variables that could +cause this limit to be exceeded. Large session tickets, multiple or large key +shares, and long lists of supported ciphers, signature algorithms, versions, +QUIC transport parameters, and other negotiable parameters and extensions could +cause this message to grow.¶
+For servers, in addition to connection IDs and tokens, the size of TLS session +tickets can have an effect on a client's ability to connect efficiently. +Minimizing the size of these values increases the probability that clients can +use them and still fit their ClientHello message in their first Initial packet.¶
+The TLS implementation does not need to ensure that the ClientHello is +sufficiently large. QUIC PADDING frames are added to increase the size of the +packet as necessary.¶
+The requirements for authentication depend on the application protocol that is +in use. TLS provides server authentication and permits the server to request +client authentication.¶
+A client MUST authenticate the identity of the server. This typically involves +verification that the identity of the server is included in a certificate and +that the certificate is issued by a trusted entity (see for example +[RFC2818]).¶
+A server MAY request that the client authenticate during the handshake. A server +MAY refuse a connection if the client is unable to authenticate when requested. +The requirements for client authentication vary based on application protocol +and deployment.¶
+A server MUST NOT use post-handshake client authentication (as defined in +Section 4.6.2 of [TLS13]), because the multiplexing offered by QUIC prevents +clients from correlating the certificate request with the application-level +event that triggered it (see [HTTP2-TLS13]). +More specifically, servers MUST NOT send post-handshake TLS CertificateRequest +messages and clients MUST treat receipt of such messages as a connection error +of type PROTOCOL_VIOLATION.¶
+To communicate their willingness to process 0-RTT data, servers send a +NewSessionTicket message that contains the "early_data" extension with a +max_early_data_size of 0xffffffff; the amount of data which the client can send +in 0-RTT is controlled by the "initial_max_data" transport parameter supplied +by the server. Servers MUST NOT send the "early_data" extension with a +max_early_data_size set to any value other than 0xffffffff. A client MUST +treat receipt of a NewSessionTicket that contains an "early_data" extension +with any other value as a connection error of type PROTOCOL_VIOLATION.¶
+A client that wishes to send 0-RTT packets uses the "early_data" extension in +the ClientHello message of a subsequent handshake (see Section 4.2.10 of +[TLS13]). It then sends the application data in 0-RTT packets.¶
+A server accepts 0-RTT by sending an early_data extension in the +EncryptedExtensions (see Section 4.2.10 of [TLS13]). The server then +processes and acknowledges the 0-RTT packets that it receives.¶
+A server rejects 0-RTT by sending the EncryptedExtensions without an early_data +extension. A server will always reject 0-RTT if it sends a TLS +HelloRetryRequest. When rejecting 0-RTT, a server MUST NOT process any 0-RTT +packets, even if it could. When 0-RTT was rejected, a client SHOULD treat +receipt of an acknowledgement for a 0-RTT packet as a connection error of type +PROTOCOL_VIOLATION, if it is able to detect the condition.¶
+When 0-RTT is rejected, all connection characteristics that the client assumed +might be incorrect. This includes the choice of application protocol, transport +parameters, and any application configuration. The client therefore MUST reset +the state of all streams, including application state bound to those streams.¶
+A client MAY attempt to send 0-RTT again if it receives a Retry or Version +Negotiation packet. These packets do not signify rejection of 0-RTT.¶
+When a server receives a ClientHello with the "early_data" extension, it has to +decide whether to accept or reject early data from the client. Some of this +decision is made by the TLS stack (e.g., checking that the cipher suite being +resumed was included in the ClientHello; see Section 4.2.10 of [TLS13]). Even +when the TLS stack has no reason to reject early data, the QUIC stack or the +application protocol using QUIC might reject early data because the +configuration of the transport or application associated with the resumed +session is not compatible with the server's current configuration.¶
+QUIC requires additional transport state to be associated with a 0-RTT session +ticket. One common way to implement this is using stateless session tickets and +storing this state in the session ticket. Application protocols that use QUIC +might have similar requirements regarding associating or storing state. This +associated state is used for deciding whether early data must be rejected. For +example, HTTP/3 ([QUIC-HTTP]) settings determine how early data from the +client is interpreted. Other applications using QUIC could have different +requirements for determining whether to accept or reject early data.¶
+In TLS over TCP, the HelloRetryRequest feature (see Section 4.1.4 of [TLS13]) +can be used to correct a client's incorrect KeyShare extension as well as for a +stateless round-trip check. From the perspective of QUIC, this just looks like +additional messages carried in the Initial encryption level. Although it is in +principle possible to use this feature for address verification in QUIC, QUIC +implementations SHOULD instead use the Retry feature (see Section 8.1 of +[QUIC-TRANSPORT]). HelloRetryRequest is still used to request key shares.¶
+If TLS experiences an error, it generates an appropriate alert as defined in +Section 6 of [TLS13].¶
+A TLS alert is turned into a QUIC connection error by converting the one-byte +alert description into a QUIC error code. The alert description is added to +0x100 to produce a QUIC error code from the range reserved for CRYPTO_ERROR. +The resulting value is sent in a QUIC CONNECTION_CLOSE frame.¶
+The alert level of all TLS alerts is "fatal"; a TLS stack MUST NOT generate +alerts at the "warning" level.¶
+After QUIC moves to a new encryption level, packet protection keys for previous +encryption levels can be discarded. This occurs several times during the +handshake, as well as when keys are updated; see Section 6.¶
+Packet protection keys are not discarded immediately when new keys are +available. If packets from a lower encryption level contain CRYPTO frames, +frames that retransmit that data MUST be sent at the same encryption level. +Similarly, an endpoint generates acknowledgements for packets at the same +encryption level as the packet being acknowledged. Thus, it is possible that +keys for a lower encryption level are needed for a short time after keys for a +newer encryption level are available.¶
+An endpoint cannot discard keys for a given encryption level unless it has both +received and acknowledged all CRYPTO frames for that encryption level and when +all CRYPTO frames for that encryption level have been acknowledged by its peer. +However, this does not guarantee that no further packets will need to be +received or sent at that encryption level because a peer might not have received +all the acknowledgements necessary to reach the same state.¶
+Though an endpoint might retain older keys, new data MUST be sent at the highest +currently-available encryption level. Only ACK frames and retransmissions of +data in CRYPTO frames are sent at a previous encryption level. These packets +MAY also include PADDING frames.¶
+Packets protected with Initial secrets (Section 5.2) are not +authenticated, meaning that an attacker could spoof packets with the intent to +disrupt a connection. To limit these attacks, Initial packet protection keys +can be discarded more aggressively than other keys.¶
+The successful use of Handshake packets indicates that no more Initial packets +need to be exchanged, as these keys can only be produced after receiving all +CRYPTO frames from Initial packets. Thus, a client MUST discard Initial keys +when it first sends a Handshake packet and a server MUST discard Initial keys +when it first successfully processes a Handshake packet. Endpoints MUST NOT +send Initial packets after this point.¶
+This results in abandoning loss recovery state for the Initial encryption level +and ignoring any outstanding Initial packets.¶
+An endpoint MUST NOT discard its handshake keys until the TLS handshake is +confirmed (Section 4.1.2). An endpoint SHOULD discard its handshake +keys as soon as it has confirmed the handshake. Most application protocols +will send data after the handshake, resulting in acknowledgements that allow +both endpoints to discard their handshake keys promptly. Endpoints that do +not have reason to send immediately after completing the handshake MAY send +ack-eliciting frames, such as PING, which will cause the handshake to be +confirmed when they are acknowledged.¶
+0-RTT and 1-RTT packets share the same packet number space, and clients do not +send 0-RTT packets after sending a 1-RTT packet (Section 5.6).¶
+Therefore, a client SHOULD discard 0-RTT keys as soon as it installs 1-RTT +keys, since they have no use after that moment.¶
+Additionally, a server MAY discard 0-RTT keys as soon as it receives a 1-RTT +packet. However, due to packet reordering, a 0-RTT packet could arrive after +a 1-RTT packet. Servers MAY temporarily retain 0-RTT keys to allow decrypting +reordered packets without requiring their contents to be retransmitted with +1-RTT keys. After receiving a 1-RTT packet, servers MUST discard 0-RTT keys +within a short time; the RECOMMENDED time period is three times the Probe +Timeout (PTO, see [QUIC-RECOVERY]). A server MAY discard 0-RTT keys earlier +if it determines that it has received all 0-RTT packets, which can be done by +keeping track of missing packet numbers.¶
+As with TLS over TCP, QUIC protects packets with keys derived from the TLS +handshake, using the AEAD algorithm negotiated by TLS.¶
+QUIC derives packet protection keys in the same way that TLS derives record +protection keys.¶
+Each encryption level has separate secret values for protection of packets sent +in each direction. These traffic secrets are derived by TLS (see Section 7.1 of +[TLS13]) and are used by QUIC for all encryption levels except the Initial +encryption level. The secrets for the Initial encryption level are computed +based on the client's initial Destination Connection ID, as described in +Section 5.2.¶
+The keys used for packet protection are computed from the TLS secrets using the +KDF provided by TLS. In TLS 1.3, the HKDF-Expand-Label function described in +Section 7.1 of [TLS13] is used, using the hash function from the negotiated +cipher suite. Other versions of TLS MUST provide a similar function in order to +be used with QUIC.¶
+The current encryption level secret and the label "quic key" are input to the +KDF to produce the AEAD key; the label "quic iv" is used to derive the IV; see +Section 5.3. The header protection key uses the "quic hp" label; see +Section 5.4. Using these labels provides key separation between QUIC +and TLS; see Section 9.5.¶
+The KDF used for initial secrets is always the HKDF-Expand-Label function from +TLS 1.3 (see Section 5.2).¶
+Initial packets are protected with a secret derived from the Destination +Connection ID field from the client's Initial packet. Specifically:¶
+initial_salt = 0xc3eef712c72ebb5a11a7d2432bb46365bef9f502 initial_secret = HKDF-Extract(initial_salt, @@ -997,52 +2040,129 @@-server_initial_secret = HKDF-Expand-Label(initial_secret, "server in", "", Hash.length) -
The hash function for HKDF when deriving initial secrets and keys is SHA-256 [SHA].
-The connection ID used with HKDF-Expand-Label is the Destination Connection ID in the Initial packet sent by the client. This will be a randomly-selected value unless the client creates the Initial packet after receiving a Retry packet, where the Destination Connection ID is selected by the server.
-The value of initial_salt is a 20 byte sequence shown in the figure in hexadecimal notation. Future versions of QUIC SHOULD generate a new salt value, thus ensuring that the keys are different for each version of QUIC. This prevents a middlebox that only recognizes one version of QUIC from seeing or modifying the contents of packets from future versions.
-The HKDF-Expand-Label function defined in TLS 1.3 MUST be used for Initial packets even where the TLS versions offered do not include TLS 1.3.
-The secrets used for protecting Initial packets change when a server sends a Retry packet to use the connection ID value selected by the server. The secrets do not change when a client changes the Destination Connection ID it uses in response to an Initial packet from the server.
- - -The hash function for HKDF when deriving initial secrets and keys is SHA-256 +[SHA].¶
+The connection ID used with HKDF-Expand-Label is the Destination Connection ID +in the Initial packet sent by the client. This will be a randomly-selected +value unless the client creates the Initial packet after receiving a Retry +packet, where the Destination Connection ID is selected by the server.¶
+The value of initial_salt is a 20 byte sequence shown in the figure in +hexadecimal notation. Future versions of QUIC SHOULD generate a new salt value, +thus ensuring that the keys are different for each version of QUIC. This +prevents a middlebox that only recognizes one version of QUIC from seeing or +modifying the contents of packets from future versions.¶
+The HKDF-Expand-Label function defined in TLS 1.3 MUST be used for Initial +packets even where the TLS versions offered do not include TLS 1.3.¶
+The secrets used for protecting Initial packets change when a server sends a +Retry packet to use the connection ID value selected by the server. The secrets +do not change when a client changes the Destination Connection ID it uses in +response to an Initial packet from the server.¶
+Appendix A contains test vectors for the initial packet encryption.
-The Authentication Encryption with Associated Data (AEAD) [AEAD] function used for QUIC packet protection is the AEAD that is negotiated for use with the TLS connection. For example, if TLS is using the TLS_AES_128_GCM_SHA256, the AEAD_AES_128_GCM function is used.
-Packets are protected prior to applying header protection (Section 5.4). The unprotected packet header is part of the associated data (A). When removing packet protection, an endpoint first removes the header protection.
-All QUIC packets other than Version Negotiation and Retry packets are protected with an AEAD algorithm [AEAD]. Prior to establishing a shared secret, packets are protected with AEAD_AES_128_GCM and a key derived from the Destination Connection ID in the client’s first Initial packet (see Section 5.2). This provides protection against off-path attackers and robustness against QUIC version unaware middleboxes, but not against on-path attackers.
-QUIC can use any of the ciphersuites defined in [TLS13] with the exception of TLS_AES_128_CCM_8_SHA256. A ciphersuite MUST NOT be negotiated unless a header protection scheme is defined for the ciphersuite. This document defines a header protection scheme for all ciphersuites defined in [TLS13] aside from TLS_AES_128_CCM_8_SHA256. These ciphersuites have a 16-byte authentication tag and produce an output 16 bytes larger than their input.
- - -Appendix A contains test vectors for the initial packet +encryption.¶
+The Authentication Encryption with Associated Data (AEAD) [AEAD] function +used for QUIC packet protection is the AEAD that is negotiated for use with the +TLS connection. For example, if TLS is using the TLS_AES_128_GCM_SHA256, the +AEAD_AES_128_GCM function is used.¶
+Packets are protected prior to applying header protection (Section 5.4). +The unprotected packet header is part of the associated data (A). When removing +packet protection, an endpoint first removes the header protection.¶
+All QUIC packets other than Version Negotiation and Retry packets are protected +with an AEAD algorithm [AEAD]. Prior to establishing a shared secret, packets +are protected with AEAD_AES_128_GCM and a key derived from the Destination +Connection ID in the client's first Initial packet (see Section 5.2). +This provides protection against off-path attackers and robustness against QUIC +version unaware middleboxes, but not against on-path attackers.¶
+QUIC can use any of the ciphersuites defined in [TLS13] with the exception of +TLS_AES_128_CCM_8_SHA256. A ciphersuite MUST NOT be negotiated unless a header +protection scheme is defined for the ciphersuite. This document defines a +header protection scheme for all ciphersuites defined in [TLS13] aside from +TLS_AES_128_CCM_8_SHA256. These ciphersuites have a 16-byte authentication tag +and produce an output 16 bytes larger than their input.¶
+The key and IV for the packet are computed as described in Section 5.1. The nonce, N, is formed by combining the packet protection IV with the packet number. The 62 bits of the reconstructed QUIC packet number in network byte order are left-padded with zeros to the size of the IV. The exclusive OR of the padded packet number and the IV forms the AEAD nonce.
-The associated data, A, for the AEAD is the contents of the QUIC header, starting from the flags byte in either the short or long header, up to and including the unprotected packet number.
-The input plaintext, P, for the AEAD is the payload of the QUIC packet, as described in [QUIC-TRANSPORT].
-The output ciphertext, C, of the AEAD is transmitted in place of P.
-Some AEAD functions have limits for how many packets can be encrypted under the same key and IV (see for example [AEBounds]). This might be lower than the packet number limit. An endpoint MUST initiate a key update (Section 6) prior to exceeding any limit set for the AEAD that is in use.
-Parts of QUIC packet headers, in particular the Packet Number field, are protected using a key that is derived separate to the packet protection key and IV. The key derived using the “quic hp” label is used to provide confidentiality protection for those fields that are not exposed to on-path elements.
-This protection applies to the least-significant bits of the first byte, plus the Packet Number field. The four least-significant bits of the first byte are protected for packets with long headers; the five least significant bits of the first byte are protected for packets with short headers. For both header forms, this covers the reserved bits and the Packet Number Length field; the Key Phase bit is also protected for packets with a short header.
-The same header protection key is used for the duration of the connection, with the value not changing after a key update (see Section 6). This allows header protection to be used to protect the key phase.
-This process does not apply to Retry or Version Negotiation packets, which do not contain a protected payload or any of the fields that are protected by this process.
-Header protection is applied after packet protection is applied (see Section 5.3). The ciphertext of the packet is sampled and used as input to an encryption algorithm. The algorithm used depends on the negotiated AEAD.
-The output of this algorithm is a 5 byte mask which is applied to the protected header fields using exclusive OR. The least significant bits of the first byte of the packet are masked by the least significant bits of the first mask byte, and the packet number is masked with the remaining bytes. Any unused bytes of mask that might result from a shorter packet number encoding are unused.
-Figure 6 shows a sample algorithm for applying header protection. Removing header protection only differs in the order in which the packet number length (pn_length) is determined.
- - +The key and IV for the packet are computed as described in Section 5.1. +The nonce, N, is formed by combining the packet protection IV with the packet +number. The 62 bits of the reconstructed QUIC packet number in network byte +order are left-padded with zeros to the size of the IV. The exclusive OR of the +padded packet number and the IV forms the AEAD nonce.¶
+The associated data, A, for the AEAD is the contents of the QUIC header, +starting from the flags byte in either the short or long header, up to and +including the unprotected packet number.¶
+The input plaintext, P, for the AEAD is the payload of the QUIC packet, as +described in [QUIC-TRANSPORT].¶
+The output ciphertext, C, of the AEAD is transmitted in place of P.¶
+Some AEAD functions have limits for how many packets can be encrypted under the +same key and IV (see for example [AEBounds]). This might be lower than the +packet number limit. An endpoint MUST initiate a key update (Section 6) +prior to exceeding any limit set for the AEAD that is in use.¶
+Parts of QUIC packet headers, in particular the Packet Number field, are +protected using a key that is derived separate to the packet protection key and +IV. The key derived using the "quic hp" label is used to provide +confidentiality protection for those fields that are not exposed to on-path +elements.¶
+This protection applies to the least-significant bits of the first byte, plus +the Packet Number field. The four least-significant bits of the first byte are +protected for packets with long headers; the five least significant bits of the +first byte are protected for packets with short headers. For both header forms, +this covers the reserved bits and the Packet Number Length field; the Key Phase +bit is also protected for packets with a short header.¶
+The same header protection key is used for the duration of the connection, with +the value not changing after a key update (see Section 6). This allows +header protection to be used to protect the key phase.¶
+This process does not apply to Retry or Version Negotiation packets, which do +not contain a protected payload or any of the fields that are protected by this +process.¶
+Header protection is applied after packet protection is applied (see Section 5.3). +The ciphertext of the packet is sampled and used as input to an encryption +algorithm. The algorithm used depends on the negotiated AEAD.¶
+The output of this algorithm is a 5 byte mask which is applied to the protected +header fields using exclusive OR. The least significant bits of the first byte +of the packet are masked by the least significant bits of the first mask byte, +and the packet number is masked with the remaining bytes. Any unused bytes of +mask that might result from a shorter packet number encoding are unused.¶
+Figure 6 shows a sample algorithm for applying header protection. Removing +header protection only differs in the order in which the packet number length +(pn_length) is determined.¶
+mask = header_protection(hp_key, sample) @@ -1057,10 +2177,16 @@-# pn_offset is the start of the Packet Number field. packet[pn_offset:pn_offset+pn_length] ^= mask[1:1+pn_length]
Figure 6: Header Protection Pseudocode
-Figure 7 shows the protected fields of long and short headers marked with an E. Figure 7 also shows the sampled fields.
- - +Figure 7 shows the protected fields of long and short headers marked with +an E. Figure 7 also shows the sampled fields.¶
+Long Header: +-+-+-+-+-+-+-+-+ @@ -1087,23 +2213,58 @@-| Protected Payload Remainder (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: Header Protection and Ciphertext Sample
-Before a TLS ciphersuite can be used with QUIC, a header protection algorithm MUST be specified for the AEAD used with that ciphersuite. This document defines algorithms for AEAD_AES_128_GCM, AEAD_AES_128_CCM, AEAD_AES_256_GCM (all AES AEADs are defined in [AEAD]), and AEAD_CHACHA20_POLY1305 [CHACHA]. Prior to TLS selecting a ciphersuite, AES header protection is used (Section 5.4.3), matching the AEAD_AES_128_GCM packet protection.
-The header protection algorithm uses both the header protection key and a sample of the ciphertext from the packet Payload field.
-The same number of bytes are always sampled, but an allowance needs to be made for the endpoint removing protection, which will not know the length of the Packet Number field. In sampling the packet ciphertext, the Packet Number field is assumed to be 4 bytes long (its maximum possible encoded length).
-An endpoint MUST discard packets that are not long enough to contain a complete sample.
-To ensure that sufficient data is available for sampling, packets are padded so that the combined lengths of the encoded packet number and protected payload is at least 4 bytes longer than the sample required for header protection. The ciphersuites defined in [TLS13] - other than TLS_AES_128_CCM_8_SHA256, for which a header protection scheme is not defined in this document - have 16-byte expansions and 16-byte header protection samples. This results in needing at least 3 bytes of frames in the unprotected payload if the packet number is encoded on a single byte, or 2 bytes of frames for a 2-byte packet number encoding.
-The sampled ciphertext for a packet with a short header can be determined by the following pseudocode:
+Before a TLS ciphersuite can be used with QUIC, a header protection algorithm +MUST be specified for the AEAD used with that ciphersuite. This document +defines algorithms for AEAD_AES_128_GCM, AEAD_AES_128_CCM, AEAD_AES_256_GCM +(all AES AEADs are defined in [AEAD]), and +AEAD_CHACHA20_POLY1305 [CHACHA]. Prior to TLS selecting a +ciphersuite, AES header protection is used (Section 5.4.3), matching the +AEAD_AES_128_GCM packet protection.¶
+The header protection algorithm uses both the header protection key and a sample +of the ciphertext from the packet Payload field.¶
+The same number of bytes are always sampled, but an allowance needs to be made +for the endpoint removing protection, which will not know the length of the +Packet Number field. In sampling the packet ciphertext, the Packet Number field +is assumed to be 4 bytes long (its maximum possible encoded length).¶
+An endpoint MUST discard packets that are not long enough to contain a complete +sample.¶
+To ensure that sufficient data is available for sampling, packets are padded so +that the combined lengths of the encoded packet number and protected payload is +at least 4 bytes longer than the sample required for header protection. The +ciphersuites defined in [TLS13] - other than TLS_AES_128_CCM_8_SHA256, for +which a header protection scheme is not defined in this document - have 16-byte +expansions and 16-byte header protection samples. This results in needing at +least 3 bytes of frames in the unprotected payload if the packet number is +encoded on a single byte, or 2 bytes of frames for a 2-byte packet number +encoding.¶
+The sampled ciphertext for a packet with a short header can be determined by the +following pseudocode:¶
+sample_offset = 1 + len(connection_id) + 4 sample = packet[sample_offset..sample_offset+sample_length] --
For example, for a packet with a short header, an 8 byte connection ID, and protected with AEAD_AES_128_GCM, the sample takes bytes 13 to 28 inclusive (using zero-based indexing).
-A packet with a long header is sampled in the same way, noting that multiple QUIC packets might be included in the same UDP datagram and that each one is handled separately.
+¶ +For example, for a packet with a short header, an 8 byte connection ID, and +protected with AEAD_AES_128_GCM, the sample takes bytes 13 to 28 inclusive +(using zero-based indexing).¶
+A packet with a long header is sampled in the same way, noting that multiple +QUIC packets might be included in the same UDP datagram and that each one is +handled separately.¶
+
sample_offset = 7 + len(destination_connection_id) +
len(source_connection_id) +
@@ -1113,71 +2274,182 @@
len(token)
sample = packet[sample_offset..sample_offset+sample_length]
-
-This section defines the packet protection algorithm for AEAD_AES_128_GCM, AEAD_AES_128_CCM, and AEAD_AES_256_GCM. AEAD_AES_128_GCM and AEAD_AES_128_CCM use 128-bit AES [AES] in electronic code-book (ECB) mode. AEAD_AES_256_GCM uses 256-bit AES in ECB mode.
-This algorithm samples 16 bytes from the packet ciphertext. This value is used as the input to AES-ECB. In pseudocode:
+¶ +This section defines the packet protection algorithm for AEAD_AES_128_GCM, +AEAD_AES_128_CCM, and AEAD_AES_256_GCM. AEAD_AES_128_GCM and +AEAD_AES_128_CCM use 128-bit AES [AES] in +electronic code-book (ECB) mode. AEAD_AES_256_GCM uses +256-bit AES in ECB mode.¶
+This algorithm samples 16 bytes from the packet ciphertext. This value is used +as the input to AES-ECB. In pseudocode:¶
+mask = AES-ECB(hp_key, sample) --
When AEAD_CHACHA20_POLY1305 is in use, header protection uses the raw ChaCha20 function as defined in Section 2.4 of [CHACHA]. This uses a 256-bit key and 16 bytes sampled from the packet protection output.
-The first 4 bytes of the sampled ciphertext are the block counter. A ChaCha20 implementation could take a 32-bit integer in place of a byte sequence, in which case the byte sequence is interpreted as a little-endian value.
-The remaining 12 bytes are used as the nonce. A ChaCha20 implementation might take an array of three 32-bit integers in place of a byte sequence, in which case the nonce bytes are interpreted as a sequence of 32-bit little-endian integers.
-The encryption mask is produced by invoking ChaCha20 to protect 5 zero bytes. In pseudocode:
+¶ +When AEAD_CHACHA20_POLY1305 is in use, header protection uses the raw ChaCha20 +function as defined in Section 2.4 of [CHACHA]. This uses a 256-bit key and +16 bytes sampled from the packet protection output.¶
+The first 4 bytes of the sampled ciphertext are the block counter. A ChaCha20 +implementation could take a 32-bit integer in place of a byte sequence, in +which case the byte sequence is interpreted as a little-endian value.¶
+The remaining 12 bytes are used as the nonce. A ChaCha20 implementation might +take an array of three 32-bit integers in place of a byte sequence, in which +case the nonce bytes are interpreted as a sequence of 32-bit little-endian +integers.¶
+The encryption mask is produced by invoking ChaCha20 to protect 5 zero bytes. In +pseudocode:¶
+
counter = sample[0..3]
nonce = sample[4..15]
mask = ChaCha20(hp_key, counter, nonce, {0,0,0,0,0})
-
-Once an endpoint successfully receives a packet with a given packet number, it MUST discard all packets in the same packet number space with higher packet numbers if they cannot be successfully unprotected with either the same key, or - if there is a key update - the next packet protection key (see Section 6). Similarly, a packet that appears to trigger a key update, but cannot be unprotected successfully MUST be discarded.
-Failure to unprotect a packet does not necessarily indicate the existence of a protocol error in a peer or an attack. The truncated packet number encoding used in QUIC can cause packet numbers to be decoded incorrectly if they are delayed significantly.
-If 0-RTT keys are available (see Section 4.5), the lack of replay protection means that restrictions on their use are necessary to avoid replay attacks on the protocol.
-A client MUST only use 0-RTT keys to protect data that is idempotent. A client MAY wish to apply additional restrictions on what data it sends prior to the completion of the TLS handshake. A client otherwise treats 0-RTT keys as equivalent to 1-RTT keys, except that it MUST NOT send ACKs with 0-RTT keys.
-A client that receives an indication that its 0-RTT data has been accepted by a server can send 0-RTT data until it receives all of the server’s handshake messages. A client SHOULD stop sending 0-RTT data if it receives an indication that 0-RTT data has been rejected.
-A server MUST NOT use 0-RTT keys to protect packets; it uses 1-RTT keys to protect acknowledgements of 0-RTT packets. A client MUST NOT attempt to decrypt 0-RTT packets it receives and instead MUST discard them.
-Once a client has installed 1-RTT keys, it MUST NOT send any more 0-RTT packets.
- - -Once an endpoint successfully receives a packet with a given packet number, it +MUST discard all packets in the same packet number space with higher packet +numbers if they cannot be successfully unprotected with either the same key, or +- if there is a key update - the next packet protection key (see +Section 6). Similarly, a packet that appears to trigger a key update, but +cannot be unprotected successfully MUST be discarded.¶
+Failure to unprotect a packet does not necessarily indicate the existence of a +protocol error in a peer or an attack. The truncated packet number encoding +used in QUIC can cause packet numbers to be decoded incorrectly if they are +delayed significantly.¶
+If 0-RTT keys are available (see Section 4.5), the lack of replay protection +means that restrictions on their use are necessary to avoid replay attacks on +the protocol.¶
+A client MUST only use 0-RTT keys to protect data that is idempotent. A client +MAY wish to apply additional restrictions on what data it sends prior to the +completion of the TLS handshake. A client otherwise treats 0-RTT keys as +equivalent to 1-RTT keys, except that it MUST NOT send ACKs with 0-RTT keys.¶
+A client that receives an indication that its 0-RTT data has been accepted by a +server can send 0-RTT data until it receives all of the server's handshake +messages. A client SHOULD stop sending 0-RTT data if it receives an indication +that 0-RTT data has been rejected.¶
+A server MUST NOT use 0-RTT keys to protect packets; it uses 1-RTT keys to +protect acknowledgements of 0-RTT packets. A client MUST NOT attempt to +decrypt 0-RTT packets it receives and instead MUST discard them.¶
+Once a client has installed 1-RTT keys, it MUST NOT send any more 0-RTT +packets.¶
+Due to reordering and loss, protected packets might be received by an endpoint before the final TLS handshake messages are received. A client will be unable to decrypt 1-RTT packets from the server, whereas a server will be able to decrypt 1-RTT packets from the client.
-Even though 1-RTT keys are available to a server after receiving the first handshake messages from a client, it is missing assurances on the client state:
- - +Due to reordering and loss, protected packets might be received by an endpoint +before the final TLS handshake messages are received. A client will be unable +to decrypt 1-RTT packets from the server, whereas a server will be able to +decrypt 1-RTT packets from the client. Endpoints in either role MUST NOT +decrypt 1-RTT packets from their peer prior to completing the handshake.¶
+Even though 1-RTT keys are available to a server after receiving the first +handshake messages from a client, it is missing assurances on the client state:¶
Therefore, the server’s use of 1-RTT keys is limited before the handshake is complete. A server MUST NOT process data from incoming 1-RTT protected packets before the TLS handshake is complete. Because sending acknowledgments indicates that all frames in a packet have been processed, a server cannot send acknowledgments for 1-RTT packets until the TLS handshake is complete. Received packets protected with 1-RTT keys MAY be stored and later decrypted and used once the handshake is complete.
-The requirement for the server to wait for the client Finished message creates a dependency on that message being delivered. A client can avoid the potential for head-of-line blocking that this implies by sending its 1-RTT packets coalesced with a handshake packet containing a copy of the CRYPTO frame that carries the Finished message, until one of the handshake packets is acknowledged. This enables immediate server processing for those packets.
-A server could receive packets protected with 0-RTT keys prior to receiving a TLS ClientHello. The server MAY retain these packets for later decryption in anticipation of receiving a ClientHello.
-Once the handshake is confirmed (see Section 4.1.2), an endpoint MAY initiate a key update.
-The Key Phase bit indicates which packet protection keys are used to protect the packet. The Key Phase bit is initially set to 0 for the first set of 1-RTT packets and toggled to signal each subsequent key update.
-The Key Phase bit allows a recipient to detect a change in keying material without needing to receive the first packet that triggered the change. An endpoint that notices a changed Key Phase bit updates keys and decrypts the packet that contains the changed value.
-This mechanism replaces the TLS KeyUpdate message. Endpoints MUST NOT send a TLS KeyUpdate message. Endpoints MUST treat the receipt of a TLS KeyUpdate message as a connection error of type 0x10a, equivalent to a fatal TLS alert of unexpected_message (see Section 4.9).
-Figure 8 shows a key update process, where the initial set of keys used (identified with @M) are replaced by updated keys (identified with @N). The value of the Key Phase bit is indicated in brackets [].
- - +Therefore, the server's use of 1-RTT keys MUST be limited to sending data before +the handshake is complete. A server MUST NOT process incoming 1-RTT protected +packets before the TLS handshake is complete. Because sending acknowledgments +indicates that all frames in a packet have been processed, a server cannot send +acknowledgments for 1-RTT packets until the TLS handshake is complete. Received +packets protected with 1-RTT keys MAY be stored and later decrypted and used +once the handshake is complete.¶
+The requirement for the server to wait for the client Finished message creates +a dependency on that message being delivered. A client can avoid the +potential for head-of-line blocking that this implies by sending its 1-RTT +packets coalesced with a handshake packet containing a copy of the CRYPTO frame +that carries the Finished message, until one of the handshake packets is +acknowledged. This enables immediate server processing for those packets.¶
+A server could receive packets protected with 0-RTT keys prior to receiving a +TLS ClientHello. The server MAY retain these packets for later decryption in +anticipation of receiving a ClientHello.¶
+Once the handshake is confirmed (see Section 4.1.2), an endpoint MAY +initiate a key update.¶
+The Key Phase bit indicates which packet protection keys are used to protect the +packet. The Key Phase bit is initially set to 0 for the first set of 1-RTT +packets and toggled to signal each subsequent key update.¶
+The Key Phase bit allows a recipient to detect a change in keying material +without needing to receive the first packet that triggered the change. An +endpoint that notices a changed Key Phase bit updates keys and decrypts the +packet that contains the changed value.¶
+This mechanism replaces the TLS KeyUpdate message. Endpoints MUST NOT send a +TLS KeyUpdate message. Endpoints MUST treat the receipt of a TLS KeyUpdate +message as a connection error of type 0x10a, equivalent to a fatal TLS alert of +unexpected_message (see Section 4.9).¶
+Figure 8 shows a key update process, where the initial set of keys used +(identified with @M) are replaced by updated keys (identified with @N). The +value of the Key Phase bit is indicated in brackets [].¶
+
Initiating Peer Responding Peer
@@ -1199,293 +2471,610 @@
-------->
Key Update Permitted ...
-Figure 8: Key Update
-Endpoints maintain separate read and write secrets for packet protection. An endpoint initiates a key update by updating its packet protection write secret and using that to protect new packets. The endpoint creates a new write secret from the existing write secret as performed in Section 7.2 of [TLS13]. This uses the KDF function provided by TLS with a label of “quic ku”. The corresponding key and IV are created from that secret as defined in Section 5.1. The header protection key is not updated.
-For example, to update write keys with TLS 1.3, HKDF-Expand-Label is used as:
+Endpoints maintain separate read and write secrets for packet protection. An +endpoint initiates a key update by updating its packet protection write secret +and using that to protect new packets. The endpoint creates a new write secret +from the existing write secret as performed in Section 7.2 of [TLS13]. This +uses the KDF function provided by TLS with a label of "quic ku". The +corresponding key and IV are created from that secret as defined in +Section 5.1. The header protection key is not updated.¶
+For example, to update write keys with TLS 1.3, HKDF-Expand-Label is used as:¶
+
secret_<n+1> = HKDF-Expand-Label(secret_<n>, "quic ku",
"", Hash.length)
-
-The endpoint toggles the value of the Key Phase bit and uses the updated key and IV to protect all subsequent packets.
-An endpoint MUST NOT initiate a key update prior to having confirmed the handshake (Section 4.1.2). An endpoint MUST NOT initiate a subsequent key update prior unless it has received an acknowledgment for a packet that was sent protected with keys from the current key phase. This ensures that keys are available to both peers before another key update can be initiated. This can be implemented by tracking the lowest packet number sent with each key phase, and the highest acknowledged packet number in the 1-RTT space: once the latter is higher than or equal to the former, another key update can be initiated.
- - -The endpoint toggles the value of the Key Phase bit and uses the updated key and +IV to protect all subsequent packets.¶
+An endpoint MUST NOT initiate a key update prior to having confirmed the +handshake (Section 4.1.2). An endpoint MUST NOT initiate a subsequent +key update prior unless it has received an acknowledgment for a packet that was +sent protected with keys from the current key phase. This ensures that keys are +available to both peers before another key update can be initiated. This can be +implemented by tracking the lowest packet number sent with each key phase, and +the highest acknowledged packet number in the 1-RTT space: once the latter is +higher than or equal to the former, another key update can be initiated.¶
+The endpoint that initiates a key update also updates the keys that it uses for receiving packets. These keys will be needed to process packets the peer sends after updating.
-An endpoint SHOULD retain old keys so that packets sent by its peer prior to receiving the key update can be processed. Discarding old keys too early can cause delayed packets to be discarded. Discarding packets will be interpreted as packet loss by the peer and could adversely affect performance.
-A peer is permitted to initiate a key update after receiving an acknowledgement of a packet in the current key phase. An endpoint detects a key update when processing a packet with a key phase that differs from the value last used to protect the last packet it sent. To process this packet, the endpoint uses the next packet protection key and IV. See Section 6.3 for considerations about generating these keys.
-If a packet is successfully processed using the next key and IV, then the peer has initiated a key update. The endpoint MUST update its send keys to the corresponding key phase in response, as described in Section 6.1. Sending keys MUST be updated before sending an acknowledgement for the packet that was received with updated keys. By acknowledging the packet that triggered the key update in a packet protected with the updated keys, the endpoint signals that the key update is complete.
-An endpoint can defer sending the packet or acknowledgement according to its normal packet sending behaviour; it is not necessary to immediately generate a packet in response to a key update. The next packet sent by the endpoint will use the updated keys. The next packet that contains an acknowledgement will cause the key update to be completed. If an endpoint detects a second update before it has sent any packets with updated keys containing an acknowledgement for the packet that initiated the key update, it indicates that its peer has updated keys twice without awaiting confirmation. An endpoint MAY treat consecutive key updates as a connection error of type KEY_UPDATE_ERROR.
-An endpoint that receives an acknowledgement that is carried in a packet protected with old keys where any acknowledged packet was protected with newer keys MAY treat that as a connection error of type KEY_UPDATE_ERROR. This indicates that a peer has received and acknowledged a packet that initiates a key update, but has not updated keys in response.
-Endpoints responding to an apparent key update MUST NOT generate a timing side-channel signal that might indicate that the Key Phase bit was invalid (see Section 9.3). Endpoints can use dummy packet protection keys in place of discarded keys when key updates are not yet permitted. Using dummy keys will generate no variation in the timing signal produced by attempting to remove packet protection, and results in all packets with an invalid Key Phase bit being rejected.
-The process of creating new packet protection keys for receiving packets could reveal that a key update has occurred. An endpoint MAY perform this process as part of packet processing, but this creates a timing signal that can be used by an attacker to learn when key updates happen and thus the value of the Key Phase bit in certain packets. Endpoints SHOULD instead defer the creation of the next set of receive packet protection keys until some time after a key update completes, up to three times the PTO; see Section 6.5.
-Once generated, the next set of packet protection keys SHOULD be retained, even if the packet that was received was subsequently discarded. Packets containing apparent key updates are easy to forge and - while the process of key update does not require significant effort - triggering this process could be used by an attacker for DoS.
-For this reason, endpoints MUST be able to retain two sets of packet protection keys for receiving packets: the current and the next. Retaining the previous keys in addition to these might improve performance, but this is not essential.
-An endpoint always sends packets that are protected with the newest keys. Keys used for packet protection can be discarded immediately after switching to newer keys.
-Packets with higher packet numbers MUST be protected with either the same or newer packet protection keys than packets with lower packet numbers. An endpoint that successfully removes protection with old keys when newer keys were used for packets with lower packet numbers MUST treat this as a connection error of type KEY_UPDATE_ERROR.
-For receiving packets during a key update, packets protected with older keys might arrive if they were delayed by the network. Retaining old packet protection keys allows these packets to be successfully processed.
-As packets protected with keys from the next key phase use the same Key Phase value as those protected with keys from the previous key phase, it can be necessary to distinguish between the two. This can be done using packet numbers. A recovered packet number that is lower than any packet number from the current key phase uses the previous packet protection keys; a recovered packet number that is higher than any packet number from the current key phase requires the use of the next packet protection keys.
-Some care is necessary to ensure that any process for selecting between previous, current, and next packet protection keys does not expose a timing side channel that might reveal which keys were used to remove packet protection. See Section 9.4 for more information.
-Alternatively, endpoints can retain only two sets of packet protection keys, swapping previous for next after enough time has passed to allow for reordering in the network. In this case, the Key Phase bit alone can be used to select keys.
-An endpoint MAY allow a period of approximately the Probe Timeout (PTO; see [QUIC-RECOVERY]) after a key update before it creates the next set of packet protection keys. These updated keys MAY replace the previous keys at that time. With the caveat that PTO is a subjective measure - that is, a peer could have a different view of the RTT - this time is expected to be long enough that any reordered packets would be declared lost by a peer even if they were acknowledged and short enough to allow for subsequent key updates.
-Endpoints need to allow for the possibility that a peer might not be able to decrypt packets that initiate a key update during the period when it retains old keys. Endpoints SHOULD wait three times the PTO before initiating a key update after receiving an acknowledgment that confirms that the previous key update was received. Failing to allow sufficient time could lead to packets being discarded.
-An endpoint SHOULD retain old read keys for no more than three times the PTO. After this period, old read keys and their corresponding secrets SHOULD be discarded.
-Key updates MUST be initiated before usage limits on packet protection keys are exceeded. For the cipher suites mentioned in this document, the limits in Section 5.5 of [TLS13] apply. Other cipher suites MUST define usage limits in order to be used with QUIC.
-The KEY_UPDATE_ERROR error code (0xE) is used to signal errors related to key updates.
-Initial packets are not protected with a secret key, so they are subject to potential tampering by an attacker. QUIC provides protection against attackers that cannot read packets, but does not attempt to provide additional protection against attacks where the attacker can observe and inject packets. Some forms of tampering – such as modifying the TLS messages themselves – are detectable, but some – such as modifying ACKs – are not.
-For example, an attacker could inject a packet containing an ACK frame that makes it appear that a packet had not been received or to create a false impression of the state of the connection (e.g., by modifying the ACK Delay). Note that such a packet could cause a legitimate packet to be dropped as a duplicate. Implementations SHOULD use caution in relying on any data which is contained in Initial packets that is not otherwise authenticated.
-It is also possible for the attacker to tamper with data that is carried in Handshake packets, but because that tampering requires modifying TLS handshake messages, that tampering will cause the TLS handshake to fail.
-QUIC uses the TLS handshake for more than just negotiation of cryptographic parameters. The TLS handshake provides preliminary values for QUIC transport parameters and allows a server to perform return routability checks on clients.
-QUIC requires that the cryptographic handshake provide authenticated protocol negotiation. TLS uses Application Layer Protocol Negotiation (ALPN) [RFC7301] to select an application protocol. Unless another mechanism is used for agreeing on an application protocol, endpoints MUST use ALPN for this purpose. When using ALPN, endpoints MUST immediately close a connection (see Section 10.3 in [QUIC-TRANSPORT]) if an application protocol is not negotiated with a no_application_protocol TLS alert (QUIC error code 0x178, see Section 4.9). While [RFC7301] only specifies that servers use this alert, QUIC clients MUST also use it to terminate a connection when ALPN negotiation fails.
-An application-layer protocol MAY restrict the QUIC versions that it can operate over. Servers MUST select an application protocol compatible with the QUIC version that the client has selected. If the server cannot select a compatible combination of application protocol and QUIC version, it MUST abort the connection. A client MUST abort a connection if the server picks an application protocol incompatible with the protocol version being used.
-QUIC transport parameters are carried in a TLS extension. Different versions of QUIC might define a different method for negotiating transport configuration.
-Including transport parameters in the TLS handshake provides integrity protection for these values.
+The endpoint that initiates a key update also updates the keys that it uses for +receiving packets. These keys will be needed to process packets the peer sends +after updating.¶
+An endpoint SHOULD retain old keys so that packets sent by its peer prior to +receiving the key update can be processed. Discarding old keys too early can +cause delayed packets to be discarded. Discarding packets will be interpreted +as packet loss by the peer and could adversely affect performance.¶
+A peer is permitted to initiate a key update after receiving an acknowledgement +of a packet in the current key phase. An endpoint detects a key update when +processing a packet with a key phase that differs from the value last used to +protect the last packet it sent. To process this packet, the endpoint uses the +next packet protection key and IV. See Section 6.3 for +considerations about generating these keys.¶
+If a packet is successfully processed using the next key and IV, then the peer +has initiated a key update. The endpoint MUST update its send keys to the +corresponding key phase in response, as described in Section 6.1. +Sending keys MUST be updated before sending an acknowledgement for the packet +that was received with updated keys. By acknowledging the packet that triggered +the key update in a packet protected with the updated keys, the endpoint signals +that the key update is complete.¶
+An endpoint can defer sending the packet or acknowledgement according to its +normal packet sending behaviour; it is not necessary to immediately generate a +packet in response to a key update. The next packet sent by the endpoint will +use the updated keys. The next packet that contains an acknowledgement will +cause the key update to be completed. If an endpoint detects a second update +before it has sent any packets with updated keys containing an +acknowledgement for the packet that initiated the key update, it indicates that +its peer has updated keys twice without awaiting confirmation. An endpoint MAY +treat consecutive key updates as a connection error of type KEY_UPDATE_ERROR.¶
+An endpoint that receives an acknowledgement that is carried in a packet +protected with old keys where any acknowledged packet was protected with newer +keys MAY treat that as a connection error of type KEY_UPDATE_ERROR. This +indicates that a peer has received and acknowledged a packet that initiates a +key update, but has not updated keys in response.¶
+Endpoints responding to an apparent key update MUST NOT generate a timing +side-channel signal that might indicate that the Key Phase bit was invalid (see +Section 9.3). Endpoints can use dummy packet protection keys in +place of discarded keys when key updates are not yet permitted. Using dummy +keys will generate no variation in the timing signal produced by attempting to +remove packet protection, and results in all packets with an invalid Key Phase +bit being rejected.¶
+The process of creating new packet protection keys for receiving packets could +reveal that a key update has occurred. An endpoint MAY perform this process as +part of packet processing, but this creates a timing signal that can be used by +an attacker to learn when key updates happen and thus the value of the Key Phase +bit in certain packets. Endpoints MAY instead defer the creation of the next +set of receive packet protection keys until some time after a key update +completes, up to three times the PTO; see Section 6.5.¶
+Once generated, the next set of packet protection keys SHOULD be retained, even +if the packet that was received was subsequently discarded. Packets containing +apparent key updates are easy to forge and - while the process of key update +does not require significant effort - triggering this process could be used by +an attacker for DoS.¶
+For this reason, endpoints MUST be able to retain two sets of packet protection +keys for receiving packets: the current and the next. Retaining the previous +keys in addition to these might improve performance, but this is not essential.¶
+An endpoint always sends packets that are protected with the newest keys. Keys +used for packet protection can be discarded immediately after switching to newer +keys.¶
+Packets with higher packet numbers MUST be protected with either the same or +newer packet protection keys than packets with lower packet numbers. An +endpoint that successfully removes protection with old keys when newer keys were +used for packets with lower packet numbers MUST treat this as a connection error +of type KEY_UPDATE_ERROR.¶
+For receiving packets during a key update, packets protected with older keys +might arrive if they were delayed by the network. Retaining old packet +protection keys allows these packets to be successfully processed.¶
+As packets protected with keys from the next key phase use the same Key Phase +value as those protected with keys from the previous key phase, it can be +necessary to distinguish between the two. This can be done using packet +numbers. A recovered packet number that is lower than any packet number from +the current key phase uses the previous packet protection keys; a recovered +packet number that is higher than any packet number from the current key phase +requires the use of the next packet protection keys.¶
+Some care is necessary to ensure that any process for selecting between +previous, current, and next packet protection keys does not expose a timing side +channel that might reveal which keys were used to remove packet protection. See +Section 9.4 for more information.¶
+Alternatively, endpoints can retain only two sets of packet protection keys, +swapping previous for next after enough time has passed to allow for reordering +in the network. In this case, the Key Phase bit alone can be used to select +keys.¶
+An endpoint MAY allow a period of approximately the Probe Timeout (PTO; see +[QUIC-RECOVERY]) after a key update before it creates the next set of packet +protection keys. These updated keys MAY replace the previous keys at that time. +With the caveat that PTO is a subjective measure - that is, a peer could have a +different view of the RTT - this time is expected to be long enough that any +reordered packets would be declared lost by a peer even if they were +acknowledged and short enough to allow for subsequent key updates.¶
+Endpoints need to allow for the possibility that a peer might not be able to +decrypt packets that initiate a key update during the period when it retains old +keys. Endpoints SHOULD wait three times the PTO before initiating a key update +after receiving an acknowledgment that confirms that the previous key update was +received. Failing to allow sufficient time could lead to packets being +discarded.¶
+An endpoint SHOULD retain old read keys for no more than three times the PTO. +After this period, old read keys and their corresponding secrets SHOULD be +discarded.¶
+Key updates MUST be initiated before usage limits on packet protection keys are +exceeded. For the cipher suites mentioned in this document, the limits in +Section 5.5 of [TLS13] apply. Other cipher suites MUST define usage limits +in order to be used with QUIC.¶
+The KEY_UPDATE_ERROR error code (0xE) is used to signal errors related to key +updates.¶
+Initial packets are not protected with a secret key, so they are subject to +potential tampering by an attacker. QUIC provides protection against attackers +that cannot read packets, but does not attempt to provide additional protection +against attacks where the attacker can observe and inject packets. Some forms +of tampering - such as modifying the TLS messages themselves - are detectable, +but some - such as modifying ACKs - are not.¶
+For example, an attacker could inject a packet containing an ACK frame that +makes it appear that a packet had not been received or to create a false +impression of the state of the connection (e.g., by modifying the ACK Delay). +Note that such a packet could cause a legitimate packet to be dropped as a +duplicate. Implementations SHOULD use caution in relying on any data which is +contained in Initial packets that is not otherwise authenticated.¶
+It is also possible for the attacker to tamper with data that is carried in +Handshake packets, but because that tampering requires modifying TLS handshake +messages, that tampering will cause the TLS handshake to fail.¶
+QUIC uses the TLS handshake for more than just negotiation of cryptographic +parameters. The TLS handshake provides preliminary values for QUIC transport +parameters and allows a server to perform return routability checks on clients.¶
+QUIC requires that the cryptographic handshake provide authenticated protocol +negotiation. TLS uses Application Layer Protocol Negotiation (ALPN) +[ALPN] to select an application protocol. Unless another mechanism +is used for agreeing on an application protocol, endpoints MUST use ALPN for +this purpose. When using ALPN, endpoints MUST immediately close a connection +(see Section 10.3 in [QUIC-TRANSPORT]) if an application protocol is not +negotiated with a no_application_protocol TLS alert (QUIC error code 0x178, see +Section 4.9). While [ALPN] only specifies that servers use this alert, +QUIC clients MUST also use it to terminate a connection when ALPN negotiation +fails.¶
+An application protocol MAY restrict the QUIC versions that it can operate over. +Servers MUST select an application protocol compatible with the QUIC version +that the client has selected. The server MUST treat the inability to select a +compatible application protocol as a connection error of type 0x178 +(no_application_protocol). Similarly, a client MUST treat the selection of an +incompatible application protocol by a server as a connection error of type +0x178.¶
+QUIC transport parameters are carried in a TLS extension. Different versions of +QUIC might define a different method for negotiating transport configuration.¶
+Including transport parameters in the TLS handshake provides integrity +protection for these values.¶
+
enum {
quic_transport_parameters(0xffa5), (65535)
} ExtensionType;
-
-The extension_data field of the quic_transport_parameters extension contains a value that is defined by the version of QUIC that is in use.
-The quic_transport_parameters extension is carried in the ClientHello and the EncryptedExtensions messages during the handshake. Endpoints MUST send the quic_transport_parameters extension; endpoints that receive ClientHello or EncryptedExtensions messages without the quic_transport_parameters extension MUST close the connection with an error of type 0x16d (equivalent to a fatal TLS missing_extension alert, see Section 4.9).
-While the transport parameters are technically available prior to the completion of the handshake, they cannot be fully trusted until the handshake completes, and reliance on them should be minimized. However, any tampering with the parameters will cause the handshake to fail.
-Endpoints MUST NOT send this extension in a TLS connection that does not use QUIC (such as the use of TLS with TCP defined in [TLS13]). A fatal unsupported_extension alert MUST be sent by an implementation that supports this extension if the extension is received when the transport is not QUIC.
-The TLS EndOfEarlyData message is not used with QUIC. QUIC does not rely on this message to mark the end of 0-RTT data or to signal the change to Handshake keys.
-Clients MUST NOT send the EndOfEarlyData message. A server MUST treat receipt of a CRYPTO frame in a 0-RTT packet as a connection error of type PROTOCOL_VIOLATION.
-As a result, EndOfEarlyData does not appear in the TLS handshake transcript.
-There are likely to be some real clangers here eventually, but the current set of issues is well captured in the relevant sections of the main text.
-Never assume that because it isn’t in the security considerations section it doesn’t affect security. Most of this document does.
-As described in Section 8 of [TLS13], use of TLS early data comes with an exposure to replay attack. The use of 0-RTT in QUIC is similarly vulnerable to replay attack.
-Endpoints MUST implement and use the replay protections described in [TLS13], however it is recognized that these protections are imperfect. Therefore, additional consideration of the risk of replay is needed.
-QUIC is not vulnerable to replay attack, except via the application protocol information it might carry. The management of QUIC protocol state based on the frame types defined in [QUIC-TRANSPORT] is not vulnerable to replay. Processing of QUIC frames is idempotent and cannot result in invalid connection states if frames are replayed, reordered or lost. QUIC connections do not produce effects that last beyond the lifetime of the connection, except for those produced by the application protocol that QUIC serves.
- - -The extension_data field of the quic_transport_parameters extension contains a
+value that is defined by the version of QUIC that is in use.¶
The quic_transport_parameters extension is carried in the ClientHello and the +EncryptedExtensions messages during the handshake. Endpoints MUST send the +quic_transport_parameters extension; endpoints that receive ClientHello or +EncryptedExtensions messages without the quic_transport_parameters extension +MUST close the connection with an error of type 0x16d (equivalent to a fatal TLS +missing_extension alert, see Section 4.9).¶
+While the transport parameters are technically available prior to the completion +of the handshake, they cannot be fully trusted until the handshake completes, +and reliance on them should be minimized. However, any tampering with the +parameters will cause the handshake to fail.¶
+Endpoints MUST NOT send this extension in a TLS connection that does not use +QUIC (such as the use of TLS with TCP defined in [TLS13]). A fatal +unsupported_extension alert MUST be sent by an implementation that supports this +extension if the extension is received when the transport is not QUIC.¶
+The TLS EndOfEarlyData message is not used with QUIC. QUIC does not rely on +this message to mark the end of 0-RTT data or to signal the change to Handshake +keys.¶
+Clients MUST NOT send the EndOfEarlyData message. A server MUST treat receipt +of a CRYPTO frame in a 0-RTT packet as a connection error of type +PROTOCOL_VIOLATION.¶
+As a result, EndOfEarlyData does not appear in the TLS handshake transcript.¶
+There are likely to be some real clangers here eventually, but the current set +of issues is well captured in the relevant sections of the main text.¶
+Never assume that because it isn't in the security considerations section it +doesn't affect security. Most of this document does.¶
+As described in Section 8 of [TLS13], use of TLS early data comes with an +exposure to replay attack. The use of 0-RTT in QUIC is similarly vulnerable to +replay attack.¶
+Endpoints MUST implement and use the replay protections described in [TLS13], +however it is recognized that these protections are imperfect. Therefore, +additional consideration of the risk of replay is needed.¶
+QUIC is not vulnerable to replay attack, except via the application protocol +information it might carry. The management of QUIC protocol state based on the +frame types defined in [QUIC-TRANSPORT] is not vulnerable to replay. +Processing of QUIC frames is idempotent and cannot result in invalid connection +states if frames are replayed, reordered or lost. QUIC connections do not +produce effects that last beyond the lifetime of the connection, except for +those produced by the application protocol that QUIC serves.¶
+A server that accepts 0-RTT on a connection incurs a higher cost than accepting a connection without 0-RTT. This includes higher processing and computation costs. Servers need to consider the probability of replay and all associated costs when accepting 0-RTT.
-Ultimately, the responsibility for managing the risks of replay attacks with 0-RTT lies with an application protocol. An application protocol that uses QUIC MUST describe how the protocol uses 0-RTT and the measures that are employed to protect against replay attack. An analysis of replay risk needs to consider all QUIC protocol features that carry application semantics.
-Disabling 0-RTT entirely is the most effective defense against replay attack.
-QUIC extensions MUST describe how replay attacks affect their operation, or prohibit their use in 0-RTT. Application protocols MUST either prohibit the use of extensions that carry application semantics in 0-RTT or provide replay mitigation strategies.
-A small ClientHello that results in a large block of handshake messages from a server can be used in packet reflection attacks to amplify the traffic generated by an attacker.
-QUIC includes three defenses against this attack. First, the packet containing a ClientHello MUST be padded to a minimum size. Second, if responding to an unverified source address, the server is forbidden to send more than three UDP datagrams in its first flight (see Section 8.1 of [QUIC-TRANSPORT]). Finally, because acknowledgements of Handshake packets are authenticated, a blind attacker cannot forge them. Put together, these defenses limit the level of amplification.
-[NAN] analyzes authenticated encryption algorithms which provide nonce privacy, referred to as “Hide Nonce” (HN) transforms. The general header protection construction in this document is one of those algorithms (HN1). Header protection uses the output of the packet protection AEAD to derive sample, and then encrypts the header field using a pseudorandom function (PRF) as follows:
+A server that accepts 0-RTT on a connection incurs a higher cost than accepting +a connection without 0-RTT. This includes higher processing and computation +costs. Servers need to consider the probability of replay and all associated +costs when accepting 0-RTT.¶
+Ultimately, the responsibility for managing the risks of replay attacks with +0-RTT lies with an application protocol. An application protocol that uses QUIC +MUST describe how the protocol uses 0-RTT and the measures that are employed to +protect against replay attack. An analysis of replay risk needs to consider +all QUIC protocol features that carry application semantics.¶
+Disabling 0-RTT entirely is the most effective defense against replay attack.¶
+QUIC extensions MUST describe how replay attacks affect their operation, or +prohibit their use in 0-RTT. Application protocols MUST either prohibit the use +of extensions that carry application semantics in 0-RTT or provide replay +mitigation strategies.¶
+A small ClientHello that results in a large block of handshake messages from a +server can be used in packet reflection attacks to amplify the traffic generated +by an attacker.¶
+QUIC includes three defenses against this attack. First, the packet containing a +ClientHello MUST be padded to a minimum size. Second, if responding to an +unverified source address, the server is forbidden to send more than three UDP +datagrams in its first flight (see Section 8.1 of [QUIC-TRANSPORT]). Finally, +because acknowledgements of Handshake packets are authenticated, a blind +attacker cannot forge them. Put together, these defenses limit the level of +amplification.¶
+[NAN] analyzes authenticated encryption
+algorithms which provide nonce privacy, referred to as "Hide Nonce" (HN)
+transforms. The general header protection construction in this document is
+one of those algorithms (HN1). Header protection uses the output of the packet
+protection AEAD to derive sample, and then encrypts the header field using
+a pseudorandom function (PRF) as follows:¶
protected_field = field XOR PRF(hp_key, sample) --
The header protection variants in this document use a pseudorandom permutation (PRP) in place of a generic PRF. However, since all PRPs are also PRFs [IMC], these variants do not deviate from the HN1 construction.
-As hp_key is distinct from the packet protection key, it follows that header protection achieves AE2 security as defined in [NAN] and therefore guarantees privacy of field, the protected packet header. Future header protection variants based on this construction MUST use a PRF to ensure equivalent security guarantees.
-Use of the same key and ciphertext sample more than once risks compromising header protection. Protecting two different headers with the same key and ciphertext sample reveals the exclusive OR of the protected fields. Assuming that the AEAD acts as a PRF, if L bits are sampled, the odds of two ciphertext samples being identical approach 2^(-L/2), that is, the birthday bound. For the algorithms described in this document, that probability is one in 2^64.
- - -The header protection variants in this document use a pseudorandom permutation +(PRP) in place of a generic PRF. However, since all PRPs are also PRFs [IMC], +these variants do not deviate from the HN1 construction.¶
+As hp_key is distinct from the packet protection key, it follows that header
+protection achieves AE2 security as defined in [NAN] and therefore guarantees
+privacy of field, the protected packet header. Future header protection
+variants based on this construction MUST use a PRF to ensure equivalent
+security guarantees.¶
Use of the same key and ciphertext sample more than once risks compromising +header protection. Protecting two different headers with the same key and +ciphertext sample reveals the exclusive OR of the protected fields. Assuming +that the AEAD acts as a PRF, if L bits are sampled, the odds of two ciphertext +samples being identical approach 2^(-L/2), that is, the birthday bound. For the +algorithms described in this document, that probability is one in 2^64.¶
+To prevent an attacker from modifying packet headers, the header is transitively authenticated using packet protection; the entire packet header is part of the authenticated additional data. Protected fields that are falsified or modified can only be detected once the packet protection is removed.
-An attacker could guess values for packet numbers or Key Phase and have an endpoint confirm guesses through timing side channels. Similarly, guesses for the packet number length can be trialed and exposed. If the recipient of a packet discards packets with duplicate packet numbers without attempting to remove packet protection they could reveal through timing side-channels that the packet number matches a received packet. For authentication to be free from side-channels, the entire process of header protection removal, packet number recovery, and packet protection removal MUST be applied together without timing and other side-channels.
-For the sending of packets, construction and protection of packet payloads and packet numbers MUST be free from side-channels that would reveal the packet number or its encoded size.
-During a key update, the time taken to generate new keys could reveal through timing side-channels that a key update has occurred. Alternatively, where an attacker injects packets this side-channel could reveal the value of the Key Phase on injected packets. After receiving a key update, an endpoint SHOULD generate and save the next set of receive packet protection keys, as described in Section 6.3. By generating new keys before a key update is received, receipt of packets will not create timing signals that leak the value of the Key Phase.
-This depends on not doing this key generation during packet processing and it can require that endpoints maintain three sets of packet protection keys for receiving: for the previous key phase, for the current key phase, and for the next key phase. Endpoints can instead choose to defer generation of the next receive packet protection keys until they discard old keys so that only two sets of receive keys need to be retained at any point in time.
-In using TLS, the central key schedule of TLS is used. As a result of the TLS handshake messages being integrated into the calculation of secrets, the inclusion of the QUIC transport parameters extension ensures that handshake and 1-RTT keys are not the same as those that might be produced by a server running TLS over TCP. To avoid the possibility of cross-protocol key synchronization, additional measures are provided to improve key separation.
-The QUIC packet protection keys and IVs are derived using a different label than the equivalent keys in TLS.
-To preserve this separation, a new version of QUIC SHOULD define new labels for key derivation for packet protection key and IV, plus the header protection keys. This version of QUIC uses the string “quic”. Other versions can use a version-specific label in place of that string.
-The initial secrets use a key that is specific to the negotiated QUIC version. New QUIC versions SHOULD define a new salt value used in calculating initial secrets.
-This document does not create any new IANA registries, but it registers the values in the following registries:
- - +To prevent an attacker from modifying packet headers, the header is transitively +authenticated using packet protection; the entire packet header is part of the +authenticated additional data. Protected fields that are falsified or modified +can only be detected once the packet protection is removed.¶
+An attacker could guess values for packet numbers or Key Phase and have an +endpoint confirm guesses through timing side channels. Similarly, guesses for +the packet number length can be trialed and exposed. If the recipient of a +packet discards packets with duplicate packet numbers without attempting to +remove packet protection they could reveal through timing side-channels that the +packet number matches a received packet. For authentication to be free from +side-channels, the entire process of header protection removal, packet number +recovery, and packet protection removal MUST be applied together without timing +and other side-channels.¶
+For the sending of packets, construction and protection of packet payloads and +packet numbers MUST be free from side-channels that would reveal the packet +number or its encoded size.¶
+During a key update, the time taken to generate new keys could reveal through +timing side-channels that a key update has occurred. Alternatively, where an +attacker injects packets this side-channel could reveal the value of the Key +Phase on injected packets. After receiving a key update, an endpoint SHOULD +generate and save the next set of receive packet protection keys, as described +in Section 6.3. By generating new keys before a key update is +received, receipt of packets will not create timing signals that leak the value +of the Key Phase.¶
+This depends on not doing this key generation during packet processing and it +can require that endpoints maintain three sets of packet protection keys for +receiving: for the previous key phase, for the current key phase, and for the +next key phase. Endpoints can instead choose to defer generation of the next +receive packet protection keys until they discard old keys so that only two sets +of receive keys need to be retained at any point in time.¶
+In using TLS, the central key schedule of TLS is used. As a result of the TLS +handshake messages being integrated into the calculation of secrets, the +inclusion of the QUIC transport parameters extension ensures that handshake and +1-RTT keys are not the same as those that might be produced by a server running +TLS over TCP. To avoid the possibility of cross-protocol key synchronization, +additional measures are provided to improve key separation.¶
+The QUIC packet protection keys and IVs are derived using a different label than +the equivalent keys in TLS.¶
+To preserve this separation, a new version of QUIC SHOULD define new labels for +key derivation for packet protection key and IV, plus the header protection +keys. This version of QUIC uses the string "quic". Other versions can use a +version-specific label in place of that string.¶
+The initial secrets use a key that is specific to the negotiated QUIC version. +New QUIC versions SHOULD define a new salt value used in calculating initial +secrets.¶
+This document does not create any new IANA registries, but it registers the +values in the following registries:¶
| [AEAD] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
| [AES] | -"Advanced encryption standard (AES)", National Institute of Standards and Technology report, DOI 10.6028/nist.fips.197, November 2001. | -
| [CHACHA] | --Nir, Y. and A. Langley, "ChaCha20 and Poly1305 for IETF Protocols", RFC 8439, DOI 10.17487/RFC8439, June 2018. | -
| [QUIC-RECOVERY] | --Iyengar, J. and I. Swett, "QUIC Loss Detection and Congestion Control", Internet-Draft draft-ietf-quic-recovery, November 2019. | -
| [QUIC-TRANSPORT] | --Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport, November 2019. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [SHA] | --Dang, Q., "Secure Hash Standard", National Institute of Standards and Technology report, DOI 10.6028/nist.fips.180-4, July 2015. | -
| [TLS-REGISTRIES] | --Salowey, J. and S. Turner, "IANA Registry Updates for TLS and DTLS", RFC 8447, DOI 10.17487/RFC8447, August 2018. | -
| [TLS13] | --Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018. | -
| [AEBounds] | --Luykx, A. and K. Paterson, "Limits on Authenticated Encryption Use in TLS", March 2016. | -
| [HTTP2-TLS13] | --Benjamin, D., "Using TLS 1.3 with HTTP/2", Internet-Draft draft-ietf-httpbis-http2-tls13-03, October 2019. | -
| [IMC] | --Katz, J. and Y. Lindell, "Introduction to Modern Cryptography, Second Edition", ISBN 978-1466570269, November 2014. | -
| [NAN] | --Bellare, M., Ng, R. and B. Tackmann, "Nonces Are Noticed: AEAD Revisited", Advances in Cryptology – CRYPTO 2019 pp. 235-265, DOI 10.1007/978-3-030-26948-7_9, 2019. | -
| [QUIC-HTTP] | --Bishop, M., "Hypertext Transfer Protocol (HTTP) over QUIC", Internet-Draft draft-ietf-quic-http, November 2019. | -
| [RFC2818] | --Rescorla, E., "HTTP Over TLS", RFC 2818, DOI 10.17487/RFC2818, May 2000. | -
| [RFC5280] | --Cooper, D., Santesson, S., Farrell, S., Boeyen, S., Housley, R. and W. Polk, "Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile", RFC 5280, DOI 10.17487/RFC5280, May 2008. | -
This section shows examples of packet protection for Initial packets so that implementations can be verified incrementally. These packets use an 8-byte client-chosen Destination Connection ID of 0x8394c8f03e515708. Values for both server and client packet protection are shown together with values in hexadecimal.
-The labels generated by the HKDF-Expand-Label function are:
- - -The initial secret is common:
+This section shows examples of packet protection for Initial packets so that +implementations can be verified incrementally. These packets use an 8-byte +client-chosen Destination Connection ID of 0x8394c8f03e515708. Values for both +server and client packet protection are shown together with values in +hexadecimal.¶
+The labels generated by the HKDF-Expand-Label function are:¶
+The initial secret is common:¶
+
initial_secret = HKDF-Extract(initial_salt, cid)
= 524e374c6da8cf8b496f4bcb69678350
7aafee6198b202b4bc823ebf7514a423
-
-The secrets for protecting client packets are:
+¶ +The secrets for protecting client packets are:¶
+
client_initial_secret
= HKDF-Expand-Label(initial_secret, "client in", _, 32)
@@ -1500,8 +3089,10 @@
hp = HKDF-Expand-Label(client_initial_secret, "quic hp", _, 16)
= a980b8b4fb7d9fbc13e814c23164253d
-
-The secrets for protecting server packets are:
+¶ +The secrets for protecting server packets are:¶
+
server_initial_secret
= HKDF-Expand-Label(initial_secret, "server in", _, 32)
@@ -1516,11 +3107,19 @@
hp = HKDF-Expand-Label(server_initial_secret, "quic hp", _, 16)
= a8ed82e6664f865aedf6106943f95fb8
-
-The client sends an Initial packet. The unprotected payload of this packet contains the following CRYPTO frame, plus enough PADDING frames to make a 1162 byte payload:
+¶ +The client sends an Initial packet. The unprotected payload of this packet +contains the following CRYPTO frame, plus enough PADDING frames to make a 1162 +byte payload:¶
+060040c4010000c003036660261ff947 cea49cce6cfad687f457cf1b14531ba1 4131a0e8f309a1d0b9c4000006130113 031302010000910000000b0009000006 @@ -1529,12 +3128,19 @@-2aedce005ff183d7bb14952072366470 37002b0003020304000d0020001e0403 05030603020308040805080604010501 060102010402050206020202002d0002 0101001c00024001 -
The unprotected header includes the connection ID and a 4 byte packet number encoding for a packet number of 2:
+¶ +The unprotected header includes the connection ID and a 4 byte packet number +encoding for a packet number of 2:¶
+-c3ff000017088394c8f03e5157080000449e00000002 --
Protecting the payload produces output that is sampled for header protection. Because the header uses a 4 byte packet number encoding, the first 16 bytes of the protected payload is sampled, then applied to the header:
+c3ff000019088394c8f03e5157080000449e00000002 +¶ +Protecting the payload produces output that is sampled for header protection. +Because the header uses a 4 byte packet number encoding, the first 16 bytes of +the protected payload is sampled, then applied to the header:¶
+sample = 535064a4268a0d9d7b1c9d250ae35516 @@ -1543,13 +3149,15 @@-header[0] ^= mask[0] & 0x0f = c0 -header[17..20] ^= mask[1..4] +header[18..21] ^= mask[1..4] = 3b343aa8 -header = c0ff000017088394c8f03e5157080000449e3b343aa8 -
The resulting protected packet is:
+header = c0ff000019088394c8f03e5157080000449e3b343aa8 +¶ +The resulting protected packet is:¶
+-c0ff000017088394c8f03e5157080000 449e3b343aa8535064a4268a0d9d7b1c +c0ff000019088394c8f03e5157080000 449e3b343aa8535064a4268a0d9d7b1c 9d250ae355162276e9b1e3011ef6bbc0 ab48ad5bcc2681e953857ca62becd752 4daac473e68d7405fbba4e9ee616c870 38bdbe908c06d9605d9ac49030359eec b1d05a14e117db8cede2bb09d0dbbfee 271cb374d8f10abec82d0f59a1dee29f @@ -1586,280 +3194,465 @@-cd191a58829104d6075c5594f627ca50 6bf181daec940f4a4f3af0074eee89da acde6758312622d4fa675b39f728e062 d2bee680d8f41a597c262648bb18bcfc 13c8b3d97b1a77b2ac3af745d61a34cc 4709865bac824a94bb19058015e4e42d -c9be6c7803567321829dd85853396269 -
The server sends the following payload in response, including an ACK frame, a CRYPTO frame, and no PADDING frames:
+aebe13f98ec51170a4aad0a8324bb768 +¶ +The server sends the following payload in response, including an ACK frame, a +CRYPTO frame, and no PADDING frames:¶
+0d0000000018410a020000560303eefc e7f7b37ba1d1632e96677825ddf73988 cfc79825df566dc5430b9a045a120013 0100002e00330024001d00209d3c940d 89690b84d08a60993c144eca684d1081 287c834d5311bcf32bb9da1a002b0002 0304 --
The header from the server includes a new connection ID and a 2-byte packet number encoding for a packet number of 1:
+¶ +The header from the server includes a new connection ID and a 2-byte packet +number encoding for a packet number of 1:¶
+-c1ff0000170008f067a5502a4262b50040740001 --
As a result, after protection, the header protection sample is taken starting from the third protected octet:
+c1ff0000190008f067a5502a4262b50040740001 +¶ +As a result, after protection, the header protection sample is taken starting +from the third protected octet:¶
+sample = 7002596f99ae67abf65a5852f54f58c3 mask = 38168a0c25 -header = c9ff0000170008f067a5502a4262b5004074168b --
The final protected packet is then:
+header = c9ff0000190008f067a5502a4262b5004074168b +¶ +The final protected packet is then:¶
+-c9ff0000170008f067a5502a4262b500 4074168bf22b7002596f99ae67abf65a +c9ff0000190008f067a5502a4262b500 4074168bf22b7002596f99ae67abf65a 5852f54f58c37c808682e2e40492d8a3 899fb04fc0afe9aabc8767b18a0aa493 537426373b48d502214dd856d63b78ce e37bc664b3fe86d487ac7a77c53038a3 -cd32f0b5004d9f5754c4f7f2d1f35cf3 f7116351c92b9cf9bb6d091ddfc8b32d -432348a2c413 --
Issue and pull request numbers are listed with a leading octothorp.
-Issue and pull request numbers are listed with a leading octothorp.¶
+Key update text update (#3050):¶
Change header protection¶
+TLS provides an AEAD and KDF function (#2046)¶
No significant changes.
-No significant changes.
-Changes to integration of the TLS handshake (#829, #1018, #1094, #1165, #1190, +#1233, #1242, #1252, #1450)¶
+No significant changes.¶
+No significant changes.¶
+This document has benefited from input from Dragana Damjanovic, Christian Huitema, Jana Iyengar, Adam Langley, Roberto Peon, Eric Rescorla, Ian Swett, and many others.
-Ryan Hamilton was originally an author of this specification.
-This document has benefited from input from Dragana Damjanovic, Christian +Huitema, Jana Iyengar, Adam Langley, Roberto Peon, Eric Rescorla, Ian Swett, and +many others.¶
+Ryan Hamilton was originally an author of this specification.¶
+| QUIC | -J. Iyengar, Ed. | -
| Internet-Draft | -Fastly | -
| Intended status: Standards Track | -M. Thomson, Ed. | -
| Expires: May 15, 2020 | -Mozilla | -
| - | November 12, 2019 | -
QUIC: A UDP-Based Multiplexed and Secure Transport
- draft-ietf-quic-transport-latest
This document defines the core of the QUIC transport protocol. Accompanying documents describe QUIC’s loss detection and congestion control and the use of TLS for key negotiation.
-Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at <https://mailarchive.ietf.org/arch/search/?email_list=quic>.
-Working Group information can be found at <https://github.com/quicwg>; source code and issues list for this draft can be found at <https://github.com/quicwg/base-drafts/labels/-transport>.
-This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
-Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.
-Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
-This Internet-Draft will expire on May 15, 2020.
-Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved.
-This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
+ /* avoid overwriting the top border line with the ToC header */ + #toc { + padding-top: 1px; + } - - -| Internet-Draft | +QUIC Transport Protocol | +January 2020 | +
| Iyengar & Thomson | +Expires 16 July 2020 | +[Page] | +
This document defines the core of the QUIC transport protocol. Accompanying +documents describe QUIC's loss detection and congestion control and the use of +TLS for key negotiation.¶
+Discussion of this draft takes place on the QUIC working group mailing list +(quic@ietf.org), which is archived at +<https://mailarchive.ietf.org/arch/search/?email_list=quic>.¶
+Working Group information can be found at <https://github.com/quicwg>; source +code and issues list for this draft can be found at +<https://github.com/quicwg/base-drafts/labels/-transport>.¶
++ This Internet-Draft is submitted in full conformance with the + provisions of BCP 78 and BCP 79.¶
++ Internet-Drafts are working documents of the Internet Engineering Task + Force (IETF). Note that other groups may also distribute working + documents as Internet-Drafts. The list of current Internet-Drafts is + at https://datatracker.ietf.org/drafts/current/.¶
++ Internet-Drafts are draft documents valid for a maximum of six months + and may be updated, replaced, or obsoleted by other documents at any + time. It is inappropriate to use Internet-Drafts as reference + material or to cite them other than as "work in progress."¶
++ This Internet-Draft will expire on 16 July 2020.¶
++ Copyright (c) 2020 IETF Trust and the persons identified as the + document authors. All rights reserved.¶
++ This document is subject to BCP 78 and the IETF Trust's Legal + Provisions Relating to IETF Documents + (https://trustee.ietf.org/license-info) in effect on the date of + publication of this document. Please review these documents + carefully, as they describe your rights and restrictions with + respect to this document. Code Components extracted from this + document must include Simplified BSD License text as described in + Section 4.e of the Trust Legal Provisions and are provided without + warranty as described in the Simplified BSD License.¶
+QUIC is a multiplexed and secure general-purpose transport protocol that +provides:¶
QUIC uses UDP as a substrate to avoid requiring changes to legacy client operating systems and middleboxes. QUIC authenticates all of its headers and encrypts most of the data it exchanges, including its signaling, to avoid incurring a dependency on middleboxes.
-This document describes the core QUIC protocol and is structured as follows.
- - +QUIC uses UDP as a substrate to avoid requiring changes to legacy client +operating systems and middleboxes. QUIC authenticates all of its headers and +encrypts most of the data it exchanges, including its signaling, to avoid +incurring a dependency on middleboxes.¶
+This document describes the core QUIC protocol and is structured as follows:¶
Streams are the basic service abstraction that QUIC provides.¶
+Connections are the context in which QUIC endpoints communicate.¶
+Packets and frames are the basic unit used by QUIC to communicate.¶
+Finally, encoding details of QUIC protocol elements are described in:¶
+Accompanying documents describe QUIC’s loss detection and congestion control [QUIC-RECOVERY], and the use of TLS for key negotiation [QUIC-TLS].
-This document defines QUIC version 1, which conforms to the protocol invariants in [QUIC-INVARIANTS].
-The keywords “MUST”, “MUST NOT”, “REQUIRED”, “SHALL”, “SHALL NOT”, “SHOULD”, “SHOULD NOT”, “RECOMMENDED”, “NOT RECOMMENDED”, “MAY”, and “OPTIONAL” in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
-Commonly used terms in the document are described below.
- - -Accompanying documents describe QUIC's loss detection and congestion control +[QUIC-RECOVERY], and the use of TLS for key negotiation [QUIC-TLS].¶
+This document defines QUIC version 1, which conforms to the protocol invariants +in [QUIC-INVARIANTS].¶
+The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", +"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this +document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] +when, and only when, they appear in all capitals, as shown here.¶
+Commonly used terms in the document are described below.¶
+Packet and frame diagrams in this document use the format described in Section 3.1 of [RFC2360], with the following additional conventions:
- - -Packet and frame diagrams in this document use the format described in Section +3.1 of [RFC2360], with the following additional conventions:¶
+Streams in QUIC provide a lightweight, ordered byte-stream abstraction to an application. Streams can be unidirectional or bidirectional. An alternative view of QUIC unidirectional streams is a “message” abstraction of practically unlimited length.
-Streams can be created by sending data. Other processes associated with stream management - ending, cancelling, and managing flow control - are all designed to impose minimal overheads. For instance, a single STREAM frame (Section 19.8) can open, carry data for, and close a stream. Streams can also be long-lived and can last the entire duration of a connection.
-Streams can be created by either endpoint, can concurrently send data interleaved with other streams, and can be cancelled. QUIC does not provide any means of ensuring ordering between bytes on different streams.
-QUIC allows for an arbitrary number of streams to operate concurrently and for an arbitrary amount of data to be sent on any stream, subject to flow control constraints (see Section 4) and stream limits.
-Streams can be unidirectional or bidirectional. Unidirectional streams carry data in one direction: from the initiator of the stream to its peer. Bidirectional streams allow for data to be sent in both directions.
-Streams are identified within a connection by a numeric value, referred to as the stream ID. A stream ID is a 62-bit integer (0 to 2^62-1) that is unique for all streams on a connection. Stream IDs are encoded as variable-length integers (see Section 16). A QUIC endpoint MUST NOT reuse a stream ID within a connection.
-The least significant bit (0x1) of the stream ID identifies the initiator of the stream. Client-initiated streams have even-numbered stream IDs (with the bit set to 0), and server-initiated streams have odd-numbered stream IDs (with the bit set to 1).
-The second least significant bit (0x2) of the stream ID distinguishes between bidirectional streams (with the bit set to 0) and unidirectional streams (with the bit set to 1).
-The least significant two bits from a stream ID therefore identify a stream as one of four types, as summarized in Table 1.
- - -| Bits | -Stream Type | -
|---|---|
| 0x0 | -Client-Initiated, Bidirectional | -
| 0x1 | -Server-Initiated, Bidirectional | -
| 0x2 | -Client-Initiated, Unidirectional | -
| 0x3 | -Server-Initiated, Unidirectional | -
Within each type, streams are created with numerically increasing stream IDs. A stream ID that is used out of order results in all streams of that type with lower-numbered stream IDs also being opened.
-The first bidirectional stream opened by the client has a stream ID of 0.
-STREAM frames (Section 19.8) encapsulate data sent by an application. An endpoint uses the Stream ID and Offset fields in STREAM frames to place data in order.
-Endpoints MUST be able to deliver stream data to an application as an ordered byte-stream. Delivering an ordered byte-stream requires that an endpoint buffer any data that is received out of order, up to the advertised flow control limit.
-QUIC makes no specific allowances for delivery of stream data out of order. However, implementations MAY choose to offer the ability to deliver data out of order to a receiving application.
-An endpoint could receive data for a stream at the same stream offset multiple times. Data that has already been received can be discarded. The data at a given offset MUST NOT change if it is sent multiple times; an endpoint MAY treat receipt of different data at the same offset within a stream as a connection error of type PROTOCOL_VIOLATION.
-Streams are an ordered byte-stream abstraction with no other structure visible to QUIC. STREAM frame boundaries are not expected to be preserved when data is transmitted, retransmitted after packet loss, or delivered to the application at a receiver.
-An endpoint MUST NOT send data on any stream without ensuring that it is within the flow control limits set by its peer. Flow control is described in detail in Section 4.
-Stream multiplexing can have a significant effect on application performance if resources allocated to streams are correctly prioritized.
-QUIC does not provide a mechanism for exchanging prioritization information. Instead, it relies on receiving priority information from the application that uses QUIC.
-A QUIC implementation SHOULD provide ways in which an application can indicate the relative priority of streams. When deciding which streams to dedicate resources to, the implementation SHOULD use the information provided by the application.
-There are certain operations which an application MUST be able to perform when interacting with QUIC streams. This document does not specify an API, but any implementation of this version of QUIC MUST expose the ability to perform the operations described in this section on a QUIC stream.
-On the sending part of a stream, application protocols need to be able to:
- - +Streams in QUIC provide a lightweight, ordered byte-stream abstraction to an +application. Streams can be unidirectional or bidirectional. An alternative +view of QUIC unidirectional streams is a "message" abstraction of practically +unlimited length.¶
+Streams can be created by sending data. Other processes associated with stream +management - ending, cancelling, and managing flow control - are all designed to +impose minimal overheads. For instance, a single STREAM frame (Section 19.8) +can open, carry data for, and close a stream. Streams can also be long-lived and +can last the entire duration of a connection.¶
+Streams can be created by either endpoint, can concurrently send data +interleaved with other streams, and can be cancelled. QUIC does not provide any +means of ensuring ordering between bytes on different streams.¶
+QUIC allows for an arbitrary number of streams to operate concurrently and for +an arbitrary amount of data to be sent on any stream, subject to flow control +constraints (see Section 4) and stream limits.¶
+Streams can be unidirectional or bidirectional. Unidirectional streams carry +data in one direction: from the initiator of the stream to its peer. +Bidirectional streams allow for data to be sent in both directions.¶
+Streams are identified within a connection by a numeric value, referred to as +the stream ID. A stream ID is a 62-bit integer (0 to 2^62-1) that is unique for +all streams on a connection. Stream IDs are encoded as variable-length integers +(see Section 16). A QUIC endpoint MUST NOT reuse a stream ID within a +connection.¶
+The least significant bit (0x1) of the stream ID identifies the initiator of the +stream. Client-initiated streams have even-numbered stream IDs (with the bit +set to 0), and server-initiated streams have odd-numbered stream IDs (with the +bit set to 1).¶
+The second least significant bit (0x2) of the stream ID distinguishes between +bidirectional streams (with the bit set to 0) and unidirectional streams (with +the bit set to 1).¶
+The least significant two bits from a stream ID therefore identify a stream as +one of four types, as summarized in Table 1.¶
+| Bits | +Stream Type | +
|---|---|
| 0x0 | +Client-Initiated, Bidirectional | +
| 0x1 | +Server-Initiated, Bidirectional | +
| 0x2 | +Client-Initiated, Unidirectional | +
| 0x3 | +Server-Initiated, Unidirectional | +
Within each type, streams are created with numerically increasing stream IDs. A +stream ID that is used out of order results in all streams of that type with +lower-numbered stream IDs also being opened.¶
+The first bidirectional stream opened by the client has a stream ID of 0.¶
+STREAM frames (Section 19.8) encapsulate data sent by an application. An +endpoint uses the Stream ID and Offset fields in STREAM frames to place data in +order.¶
+Endpoints MUST be able to deliver stream data to an application as an ordered +byte-stream. Delivering an ordered byte-stream requires that an endpoint buffer +any data that is received out of order, up to the advertised flow control limit.¶
+QUIC makes no specific allowances for delivery of stream data out of +order. However, implementations MAY choose to offer the ability to deliver data +out of order to a receiving application.¶
+An endpoint could receive data for a stream at the same stream offset multiple +times. Data that has already been received can be discarded. The data at a +given offset MUST NOT change if it is sent multiple times; an endpoint MAY treat +receipt of different data at the same offset within a stream as a connection +error of type PROTOCOL_VIOLATION.¶
+Streams are an ordered byte-stream abstraction with no other structure visible +to QUIC. STREAM frame boundaries are not expected to be preserved when +data is transmitted, retransmitted after packet loss, or delivered to the +application at a receiver.¶
+An endpoint MUST NOT send data on any stream without ensuring that it is within +the flow control limits set by its peer. Flow control is described in detail in +Section 4.¶
+Stream multiplexing can have a significant effect on application performance if +resources allocated to streams are correctly prioritized.¶
+QUIC does not provide a mechanism for exchanging prioritization information. +Instead, it relies on receiving priority information from the application that +uses QUIC.¶
+A QUIC implementation SHOULD provide ways in which an application can indicate +the relative priority of streams. When deciding which streams to dedicate +resources to, the implementation SHOULD use the information provided by the +application.¶
+There are certain operations which an application MUST be able to perform when +interacting with QUIC streams. This document does not specify an API, but +any implementation of this version of QUIC MUST expose the ability to perform +the operations described in this section on a QUIC stream.¶
+On the sending part of a stream, application protocols need to be able to:¶
On the receiving part of a stream, application protocols need to be able to:
- - +On the receiving part of a stream, application protocols need to be able to:¶
Applications also need to be informed of state changes on streams, including when the peer has opened or reset a stream, when a peer aborts reading on a stream, when new data is available, and when data can or cannot be written to the stream due to flow control.
-This section describes streams in terms of their send or receive components. Two state machines are described: one for the streams on which an endpoint transmits data (Section 3.1), and another for streams on which an endpoint receives data (Section 3.2).
-Unidirectional streams use the applicable state machine directly. Bidirectional streams use both state machines. For the most part, the use of these state machines is the same whether the stream is unidirectional or bidirectional. The conditions for opening a stream are slightly more complex for a bidirectional stream because the opening of either send or receive sides causes the stream to open in both directions.
-An endpoint MUST open streams of the same type in increasing order of stream ID.
- - -Applications also need to be informed of state changes on streams, including +when the peer has opened or reset a stream, when a peer aborts reading on a +stream, when new data is available, and when data can or cannot be written to +the stream due to flow control.¶
+This section describes streams in terms of their send or receive components. +Two state machines are described: one for the streams on which an endpoint +transmits data (Section 3.1), and another for streams on which an +endpoint receives data (Section 3.2).¶
+Unidirectional streams use the applicable state machine directly. Bidirectional +streams use both state machines. For the most part, the use of these state +machines is the same whether the stream is unidirectional or bidirectional. The +conditions for opening a stream are slightly more complex for a bidirectional +stream because the opening of either send or receive sides causes the stream +to open in both directions.¶
+An endpoint MUST open streams of the same type in increasing order of stream ID.¶
+Figure 1 shows the states for the part of a stream that sends data to a peer.
- - +Figure 1 shows the states for the part of a stream that sends +data to a peer.¶
+
o
| Create Stream (Sending)
@@ -1170,22 +2033,67 @@
| Recvd | | Recvd |
+-------+ +-------+
-Figure 1: States for Sending Parts of Streams
-The sending part of stream that the endpoint initiates (types 0 and 2 for clients, 1 and 3 for servers) is opened by the application. The “Ready” state represents a newly created stream that is able to accept data from the application. Stream data might be buffered in this state in preparation for sending.
-Sending the first STREAM or STREAM_DATA_BLOCKED frame causes a sending part of a stream to enter the “Send” state. An implementation might choose to defer allocating a stream ID to a stream until it sends the first STREAM frame and enters this state, which can allow for better stream prioritization.
-The sending part of a bidirectional stream initiated by a peer (type 0 for a server, type 1 for a client) enters the “Ready” state then immediately transitions to the “Send” state if the receiving part enters the “Recv” state (Section 3.2).
-In the “Send” state, an endpoint transmits - and retransmits as necessary - stream data in STREAM frames. The endpoint respects the flow control limits set by its peer, and continues to accept and process MAX_STREAM_DATA frames. An endpoint in the “Send” state generates STREAM_DATA_BLOCKED frames if it is blocked from sending by stream or connection flow control limits Section 4.1.
-After the application indicates that all stream data has been sent and a STREAM frame containing the FIN bit is sent, the sending part of the stream enters the “Data Sent” state. From this state, the endpoint only retransmits stream data as necessary. The endpoint does not need to check flow control limits or send STREAM_DATA_BLOCKED frames for a stream in this state. MAX_STREAM_DATA frames might be received until the peer receives the final stream offset. The endpoint can safely ignore any MAX_STREAM_DATA frames it receives from its peer for a stream in this state.
-Once all stream data has been successfully acknowledged, the sending part of the stream enters the “Data Recvd” state, which is a terminal state.
-From any of the “Ready”, “Send”, or “Data Sent” states, an application can signal that it wishes to abandon transmission of stream data. Alternatively, an endpoint might receive a STOP_SENDING frame from its peer. In either case, the endpoint sends a RESET_STREAM frame, which causes the stream to enter the “Reset Sent” state.
-An endpoint MAY send a RESET_STREAM as the first frame that mentions a stream; this causes the sending part of that stream to open and then immediately transition to the “Reset Sent” state.
-Once a packet containing a RESET_STREAM has been acknowledged, the sending part of the stream enters the “Reset Recvd” state, which is a terminal state.
-Figure 2 shows the states for the part of a stream that receives data from a peer. The states for a receiving part of a stream mirror only some of the states of the sending part of the stream at the peer. The receiving part of a stream does not track states on the sending part that cannot be observed, such as the “Ready” state. Instead, the receiving part of a stream tracks the delivery of data to the application, some of which cannot be observed by the sender.
- - +The sending part of stream that the endpoint initiates (types 0 +and 2 for clients, 1 and 3 for servers) is opened by the application. The +"Ready" state represents a newly created stream that is able to accept data from +the application. Stream data might be buffered in this state in preparation for +sending.¶
+Sending the first STREAM or STREAM_DATA_BLOCKED frame causes a sending part of a +stream to enter the "Send" state. An implementation might choose to defer +allocating a stream ID to a stream until it sends the first STREAM frame and +enters this state, which can allow for better stream prioritization.¶
+The sending part of a bidirectional stream initiated by a peer (type 0 for a +server, type 1 for a client) enters the "Ready" state then immediately +transitions to the "Send" state if the receiving part enters the "Recv" state +(Section 3.2).¶
+In the "Send" state, an endpoint transmits - and retransmits as necessary - +stream data in STREAM frames. The endpoint respects the flow control limits set +by its peer, and continues to accept and process MAX_STREAM_DATA frames. An +endpoint in the "Send" state generates STREAM_DATA_BLOCKED frames if it is +blocked from sending by stream or connection flow control limits +Section 4.1.¶
+After the application indicates that all stream data has been sent and a STREAM +frame containing the FIN bit is sent, the sending part of the stream enters the +"Data Sent" state. From this state, the endpoint only retransmits stream data +as necessary. The endpoint does not need to check flow control limits or send +STREAM_DATA_BLOCKED frames for a stream in this state. MAX_STREAM_DATA frames +might be received until the peer receives the final stream offset. The endpoint +can safely ignore any MAX_STREAM_DATA frames it receives from its peer for a +stream in this state.¶
+Once all stream data has been successfully acknowledged, the sending part of the +stream enters the "Data Recvd" state, which is a terminal state.¶
+From any of the "Ready", "Send", or "Data Sent" states, an application can +signal that it wishes to abandon transmission of stream data. Alternatively, an +endpoint might receive a STOP_SENDING frame from its peer. In either case, the +endpoint sends a RESET_STREAM frame, which causes the stream to enter the "Reset +Sent" state.¶
+An endpoint MAY send a RESET_STREAM as the first frame that mentions a stream; +this causes the sending part of that stream to open and then immediately +transition to the "Reset Sent" state.¶
+Once a packet containing a RESET_STREAM has been acknowledged, the sending part +of the stream enters the "Reset Recvd" state, which is a terminal state.¶
+Figure 2 shows the states for the part of a stream that +receives data from a peer. The states for a receiving part of a stream mirror +only some of the states of the sending part of the stream at the peer. The +receiving part of a stream does not track states on the sending part that cannot +be observed, such as the "Ready" state. Instead, the receiving part of a stream +tracks the delivery of data to the application, some of which cannot be observed +by the sender.¶
+
o
| Recv STREAM / STREAM_DATA_BLOCKED / RESET_STREAM
@@ -1219,317 +2127,885 @@
| Read | | Read |
+-------+ +-------+
-Figure 2: States for Receiving Parts of Streams
-The receiving part of a stream initiated by a peer (types 1 and 3 for a client, or 0 and 2 for a server) is created when the first STREAM, STREAM_DATA_BLOCKED, or RESET_STREAM is received for that stream. For bidirectional streams initiated by a peer, receipt of a MAX_STREAM_DATA or STOP_SENDING frame for the sending part of the stream also creates the receiving part. The initial state for the receiving part of stream is “Recv”.
-The receiving part of a stream enters the “Recv” state when the sending part of a bidirectional stream initiated by the endpoint (type 0 for a client, type 1 for a server) enters the “Ready” state.
-An endpoint opens a bidirectional stream when a MAX_STREAM_DATA or STOP_SENDING frame is received from the peer for that stream. Receiving a MAX_STREAM_DATA frame for an unopened stream indicates that the remote peer has opened the stream and is providing flow control credit. Receiving a STOP_SENDING frame for an unopened stream indicates that the remote peer no longer wishes to receive data on this stream. Either frame might arrive before a STREAM or STREAM_DATA_BLOCKED frame if packets are lost or reordered.
-Before a stream is created, all streams of the same type with lower-numbered stream IDs MUST be created. This ensures that the creation order for streams is consistent on both endpoints.
-In the “Recv” state, the endpoint receives STREAM and STREAM_DATA_BLOCKED frames. Incoming data is buffered and can be reassembled into the correct order for delivery to the application. As data is consumed by the application and buffer space becomes available, the endpoint sends MAX_STREAM_DATA frames to allow the peer to send more data.
-When a STREAM frame with a FIN bit is received, the final size of the stream is known (see Section 4.4). The receiving part of the stream then enters the “Size Known” state. In this state, the endpoint no longer needs to send MAX_STREAM_DATA frames, it only receives any retransmissions of stream data.
-Once all data for the stream has been received, the receiving part enters the “Data Recvd” state. This might happen as a result of receiving the same STREAM frame that causes the transition to “Size Known”. After all data has been received, any STREAM or STREAM_DATA_BLOCKED frames for the stream can be discarded.
-The “Data Recvd” state persists until stream data has been delivered to the application. Once stream data has been delivered, the stream enters the “Data Read” state, which is a terminal state.
-Receiving a RESET_STREAM frame in the “Recv” or “Size Known” states causes the stream to enter the “Reset Recvd” state. This might cause the delivery of stream data to the application to be interrupted.
-It is possible that all stream data is received when a RESET_STREAM is received (that is, from the “Data Recvd” state). Similarly, it is possible for remaining stream data to arrive after receiving a RESET_STREAM frame (the “Reset Recvd” state). An implementation is free to manage this situation as it chooses.
-Sending RESET_STREAM means that an endpoint cannot guarantee delivery of stream data; however there is no requirement that stream data not be delivered if a RESET_STREAM is received. An implementation MAY interrupt delivery of stream data, discard any data that was not consumed, and signal the receipt of the RESET_STREAM. A RESET_STREAM signal might be suppressed or withheld if stream data is completely received and is buffered to be read by the application. If the RESET_STREAM is suppressed, the receiving part of the stream remains in “Data Recvd”.
-Once the application receives the signal indicating that the stream was reset, the receiving part of the stream transitions to the “Reset Read” state, which is a terminal state.
-The sender of a stream sends just three frame types that affect the state of a stream at either sender or receiver: STREAM (Section 19.8), STREAM_DATA_BLOCKED (Section 19.13), and RESET_STREAM (Section 19.4).
-A sender MUST NOT send any of these frames from a terminal state (“Data Recvd” or “Reset Recvd”). A sender MUST NOT send STREAM or STREAM_DATA_BLOCKED after sending a RESET_STREAM; that is, in the terminal states and in the “Reset Sent” state. A receiver could receive any of these three frames in any state, due to the possibility of delayed delivery of packets carrying them.
-The receiver of a stream sends MAX_STREAM_DATA (Section 19.10) and STOP_SENDING frames (Section 19.5).
-The receiver only sends MAX_STREAM_DATA in the “Recv” state. A receiver can send STOP_SENDING in any state where it has not received a RESET_STREAM frame; that is states other than “Reset Recvd” or “Reset Read”. However there is little value in sending a STOP_SENDING frame in the “Data Recvd” state, since all stream data has been received. A sender could receive either of these two frames in any state as a result of delayed delivery of packets.
-A bidirectional stream is composed of sending and receiving parts. Implementations may represent states of the bidirectional stream as composites of sending and receiving stream states. The simplest model presents the stream as “open” when either sending or receiving parts are in a non-terminal state and “closed” when both sending and receiving streams are in terminal states.
-Table 2 shows a more complex mapping of bidirectional stream states that loosely correspond to the stream states in HTTP/2 [HTTP2]. This shows that multiple states on sending or receiving parts of streams are mapped to the same composite state. Note that this is just one possibility for such a mapping; this mapping requires that data is acknowledged before the transition to a “closed” or “half-closed” state.
- - -| Sending Part | -Receiving Part | -Composite State | -
|---|---|---|
| No Stream/Ready | -No Stream/Recv *1 | -idle | -
| Ready/Send/Data Sent | -Recv/Size Known | -open | -
| Ready/Send/Data Sent | -Data Recvd/Data Read | -half-closed (remote) | -
| Ready/Send/Data Sent | -Reset Recvd/Reset Read | -half-closed (remote) | -
| Data Recvd | -Recv/Size Known | -half-closed (local) | -
| Reset Sent/Reset Recvd | -Recv/Size Known | -half-closed (local) | -
| Reset Sent/Reset Recvd | -Data Recvd/Data Read | -closed | -
| Reset Sent/Reset Recvd | -Reset Recvd/Reset Read | -closed | -
| Data Recvd | -Data Recvd/Data Read | -closed | -
| Data Recvd | -Reset Recvd/Reset Read | -closed | -
The receiving part of a stream initiated by a peer (types 1 and 3 for a client, +or 0 and 2 for a server) is created when the first STREAM, STREAM_DATA_BLOCKED, +or RESET_STREAM is received for that stream. For bidirectional streams +initiated by a peer, receipt of a MAX_STREAM_DATA or STOP_SENDING frame for the +sending part of the stream also creates the receiving part. The initial state +for the receiving part of stream is "Recv".¶
+The receiving part of a stream enters the "Recv" state when the sending part of +a bidirectional stream initiated by the endpoint (type 0 for a client, type 1 +for a server) enters the "Ready" state.¶
+An endpoint opens a bidirectional stream when a MAX_STREAM_DATA or STOP_SENDING +frame is received from the peer for that stream. Receiving a MAX_STREAM_DATA +frame for an unopened stream indicates that the remote peer has opened the +stream and is providing flow control credit. Receiving a STOP_SENDING frame for +an unopened stream indicates that the remote peer no longer wishes to receive +data on this stream. Either frame might arrive before a STREAM or +STREAM_DATA_BLOCKED frame if packets are lost or reordered.¶
+Before a stream is created, all streams of the same type with lower-numbered +stream IDs MUST be created. This ensures that the creation order for streams is +consistent on both endpoints.¶
+In the "Recv" state, the endpoint receives STREAM and STREAM_DATA_BLOCKED +frames. Incoming data is buffered and can be reassembled into the correct order +for delivery to the application. As data is consumed by the application and +buffer space becomes available, the endpoint sends MAX_STREAM_DATA frames to +allow the peer to send more data.¶
+When a STREAM frame with a FIN bit is received, the final size of the stream is +known (see Section 4.4). The receiving part of the stream then enters the +"Size Known" state. In this state, the endpoint no longer needs to send +MAX_STREAM_DATA frames, it only receives any retransmissions of stream data.¶
+Once all data for the stream has been received, the receiving part enters the +"Data Recvd" state. This might happen as a result of receiving the same STREAM +frame that causes the transition to "Size Known". After all data has been +received, any STREAM or STREAM_DATA_BLOCKED frames for the stream can be +discarded.¶
+The "Data Recvd" state persists until stream data has been delivered to the +application. Once stream data has been delivered, the stream enters the "Data +Read" state, which is a terminal state.¶
+Receiving a RESET_STREAM frame in the "Recv" or "Size Known" states causes the +stream to enter the "Reset Recvd" state. This might cause the delivery of +stream data to the application to be interrupted.¶
+It is possible that all stream data is received when a RESET_STREAM is received +(that is, from the "Data Recvd" state). Similarly, it is possible for remaining +stream data to arrive after receiving a RESET_STREAM frame (the "Reset Recvd" +state). An implementation is free to manage this situation as it chooses.¶
+Sending RESET_STREAM means that an endpoint cannot guarantee delivery of stream +data; however there is no requirement that stream data not be delivered if a +RESET_STREAM is received. An implementation MAY interrupt delivery of stream +data, discard any data that was not consumed, and signal the receipt of the +RESET_STREAM. A RESET_STREAM signal might be suppressed or withheld if stream +data is completely received and is buffered to be read by the application. If +the RESET_STREAM is suppressed, the receiving part of the stream remains in +"Data Recvd".¶
+Once the application receives the signal indicating that the stream +was reset, the receiving part of the stream transitions to the "Reset Read" +state, which is a terminal state.¶
+The sender of a stream sends just three frame types that affect the state of a +stream at either sender or receiver: STREAM (Section 19.8), +STREAM_DATA_BLOCKED (Section 19.13), and RESET_STREAM +(Section 19.4).¶
+A sender MUST NOT send any of these frames from a terminal state ("Data Recvd" +or "Reset Recvd"). A sender MUST NOT send STREAM or STREAM_DATA_BLOCKED after +sending a RESET_STREAM; that is, in the terminal states and in the "Reset Sent" +state. A receiver could receive any of these three frames in any state, due to +the possibility of delayed delivery of packets carrying them.¶
+The receiver of a stream sends MAX_STREAM_DATA (Section 19.10) and +STOP_SENDING frames (Section 19.5).¶
+The receiver only sends MAX_STREAM_DATA in the "Recv" state. A receiver can +send STOP_SENDING in any state where it has not received a RESET_STREAM frame; +that is states other than "Reset Recvd" or "Reset Read". However there is +little value in sending a STOP_SENDING frame in the "Data Recvd" state, since +all stream data has been received. A sender could receive either of these two +frames in any state as a result of delayed delivery of packets.¶
+A bidirectional stream is composed of sending and receiving parts. +Implementations may represent states of the bidirectional stream as composites +of sending and receiving stream states. The simplest model presents the stream +as "open" when either sending or receiving parts are in a non-terminal state and +"closed" when both sending and receiving streams are in terminal states.¶
+Table 2 shows a more complex mapping of bidirectional stream +states that loosely correspond to the stream states in HTTP/2 +[HTTP2]. This shows that multiple states on sending or receiving +parts of streams are mapped to the same composite state. Note that this is just +one possibility for such a mapping; this mapping requires that data is +acknowledged before the transition to a "closed" or "half-closed" state.¶
+| Sending Part | +Receiving Part | +Composite State | +
|---|---|---|
| No Stream/Ready | +No Stream/Recv *1 | +idle | +
| Ready/Send/Data Sent | +Recv/Size Known | +open | +
| Ready/Send/Data Sent | +Data Recvd/Data Read | +half-closed (remote) | +
| Ready/Send/Data Sent | +Reset Recvd/Reset Read | +half-closed (remote) | +
| Data Recvd | +Recv/Size Known | +half-closed (local) | +
| Reset Sent/Reset Recvd | +Recv/Size Known | +half-closed (local) | +
| Reset Sent/Reset Recvd | +Data Recvd/Data Read | +closed | +
| Reset Sent/Reset Recvd | +Reset Recvd/Reset Read | +closed | +
| Data Recvd | +Data Recvd/Data Read | +closed | +
| Data Recvd | +Reset Recvd/Reset Read | +closed | +
If an application is no longer interested in the data it is receiving on a stream, it can abort reading the stream and specify an application error code.
-If the stream is in the “Recv” or “Size Known” states, the transport SHOULD signal this by sending a STOP_SENDING frame to prompt closure of the stream in the opposite direction. This typically indicates that the receiving application is no longer reading data it receives from the stream, but it is not a guarantee that incoming data will be ignored.
-STREAM frames received after sending STOP_SENDING are still counted toward connection and stream flow control, even though these frames can be discarded upon receipt.
-A STOP_SENDING frame requests that the receiving endpoint send a RESET_STREAM frame. An endpoint that receives a STOP_SENDING frame MUST send a RESET_STREAM frame if the stream is in the Ready or Send state. If the stream is in the Data Sent state and any outstanding data is declared lost, an endpoint SHOULD send a RESET_STREAM frame in lieu of a retransmission.
-An endpoint SHOULD copy the error code from the STOP_SENDING frame to the RESET_STREAM frame it sends, but MAY use any application error code. The endpoint that sends a STOP_SENDING frame MAY ignore the error code carried in any RESET_STREAM frame it receives.
-If the STOP_SENDING frame is received on a stream that is already in the “Data Sent” state, an endpoint that wishes to cease retransmission of previously-sent STREAM frames on that stream MUST first send a RESET_STREAM frame.
-STOP_SENDING SHOULD only be sent for a stream that has not been reset by the peer. STOP_SENDING is most useful for streams in the “Recv” or “Size Known” states.
-An endpoint is expected to send another STOP_SENDING frame if a packet containing a previous STOP_SENDING is lost. However, once either all stream data or a RESET_STREAM frame has been received for the stream - that is, the stream is in any state other than “Recv” or “Size Known” - sending a STOP_SENDING frame is unnecessary.
-An endpoint that wishes to terminate both directions of a bidirectional stream can terminate one direction by sending a RESET_STREAM, and it can encourage prompt termination in the opposite direction by sending a STOP_SENDING frame.
-It is necessary to limit the amount of data that a receiver could buffer, to prevent a fast sender from overwhelming a slow receiver, or to prevent a malicious sender from consuming a large amount of memory at a receiver. To enable a receiver to limit memory commitment to a connection and to apply back pressure on the sender, streams are flow controlled both individually and as an aggregate. A QUIC receiver controls the maximum amount of data the sender can send on a stream at any time, as described in Section 4.1 and Section 4.2
-Similarly, to limit concurrency within a connection, a QUIC endpoint controls the maximum cumulative number of streams that its peer can initiate, as described in Section 4.5.
-Data sent in CRYPTO frames is not flow controlled in the same way as stream data. QUIC relies on the cryptographic protocol implementation to avoid excessive buffering of data; see [QUIC-TLS]. The implementation SHOULD provide an interface to QUIC to tell it about its buffering limits so that there is not excessive buffering at multiple layers.
-QUIC employs a credit-based flow-control scheme similar to that in HTTP/2 [HTTP2], where a receiver advertises the number of bytes it is prepared to receive on a given stream and for the entire connection. This leads to two levels of data flow control in QUIC:
- - +If an application is no longer interested in the data it is receiving on a +stream, it can abort reading the stream and specify an application error code.¶
+If the stream is in the "Recv" or "Size Known" states, the transport SHOULD +signal this by sending a STOP_SENDING frame to prompt closure of the stream in +the opposite direction. This typically indicates that the receiving application +is no longer reading data it receives from the stream, but it is not a guarantee +that incoming data will be ignored.¶
+STREAM frames received after sending STOP_SENDING are still counted toward +connection and stream flow control, even though these frames can be discarded +upon receipt.¶
+A STOP_SENDING frame requests that the receiving endpoint send a RESET_STREAM +frame. An endpoint that receives a STOP_SENDING frame MUST send a RESET_STREAM +frame if the stream is in the Ready or Send state. If the stream is in the Data +Sent state and any outstanding data is declared lost, an endpoint SHOULD send a +RESET_STREAM frame in lieu of a retransmission.¶
+An endpoint SHOULD copy the error code from the STOP_SENDING frame to the +RESET_STREAM frame it sends, but MAY use any application error code. The +endpoint that sends a STOP_SENDING frame MAY ignore the error code carried in +any RESET_STREAM frame it receives.¶
+If the STOP_SENDING frame is received on a stream that is already in the +"Data Sent" state, an endpoint that wishes to cease retransmission of +previously-sent STREAM frames on that stream MUST first send a RESET_STREAM +frame.¶
+STOP_SENDING SHOULD only be sent for a stream that has not been reset by the +peer. STOP_SENDING is most useful for streams in the "Recv" or "Size Known" +states.¶
+An endpoint is expected to send another STOP_SENDING frame if a packet +containing a previous STOP_SENDING is lost. However, once either all stream +data or a RESET_STREAM frame has been received for the stream - that is, the +stream is in any state other than "Recv" or "Size Known" - sending a +STOP_SENDING frame is unnecessary.¶
+An endpoint that wishes to terminate both directions of a bidirectional stream +can terminate one direction by sending a RESET_STREAM, and it can encourage +prompt termination in the opposite direction by sending a STOP_SENDING frame.¶
+It is necessary to limit the amount of data that a receiver could buffer, to +prevent a fast sender from overwhelming a slow receiver, or to prevent a +malicious sender from consuming a large amount of memory at a receiver. To +enable a receiver to limit memory commitment to a connection and to apply back +pressure on the sender, streams are flow controlled both individually and as an +aggregate. A QUIC receiver controls the maximum amount of data the sender can +send on a stream at any time, as described in Section 4.1 and +Section 4.2¶
+Similarly, to limit concurrency within a connection, a QUIC endpoint controls +the maximum cumulative number of streams that its peer can initiate, as +described in Section 4.5.¶
+Data sent in CRYPTO frames is not flow controlled in the same way as stream +data. QUIC relies on the cryptographic protocol implementation to avoid +excessive buffering of data; see [QUIC-TLS]. The implementation SHOULD +provide an interface to QUIC to tell it about its buffering limits so that there +is not excessive buffering at multiple layers.¶
+QUIC employs a credit-based flow-control scheme similar to that in HTTP/2 +[HTTP2], where a receiver advertises the number of bytes it is prepared to +receive on a given stream and for the entire connection. This leads to two +levels of data flow control in QUIC:¶
A receiver sets initial credits for all streams by sending transport parameters during the handshake (Section 7.3). A receiver sends MAX_STREAM_DATA (Section 19.10) or MAX_DATA (Section 19.9) frames to the sender to advertise additional credit.
-A receiver advertises credit for a stream by sending a MAX_STREAM_DATA frame with the Stream ID field set appropriately. A MAX_STREAM_DATA frame indicates the maximum absolute byte offset of a stream. A receiver could use the current offset of data consumed to determine the flow control offset to be advertised. A receiver MAY send MAX_STREAM_DATA frames in multiple packets in order to make sure that the sender receives an update before running out of flow control credit, even if one of the packets is lost.
-A receiver advertises credit for a connection by sending a MAX_DATA frame, which indicates the maximum of the sum of the absolute byte offsets of all streams. A receiver maintains a cumulative sum of bytes received on all streams, which is used to check for flow control violations. A receiver might use a sum of bytes consumed on all streams to determine the maximum data limit to be advertised.
-A receiver can advertise a larger offset by sending MAX_STREAM_DATA or MAX_DATA frames. Once a receiver advertises an offset, it MAY advertise a smaller offset, but this has no effect.
-A receiver MUST close the connection with a FLOW_CONTROL_ERROR error (Section 11) if the sender violates the advertised connection or stream data limits.
-A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do not increase flow control limits.
-If a sender runs out of flow control credit, it will be unable to send new data and is considered blocked. A sender SHOULD send a STREAM_DATA_BLOCKED or DATA_BLOCKED frame to indicate it has data to write but is blocked by flow control limits. These frames are expected to be sent infrequently in common cases, but they are considered useful for debugging and monitoring purposes.
-A sender SHOULD NOT send multiple STREAM_DATA_BLOCKED or DATA_BLOCKED frames for the same data limit, unless the original frame is determined to be lost. Another STREAM_DATA_BLOCKED or DATA_BLOCKED frame can be sent after the data limit is increased.
-This document leaves when and how many bytes to advertise in a MAX_STREAM_DATA or MAX_DATA frame to implementations, but offers a few considerations. These frames contribute to connection overhead. Therefore frequently sending frames with small changes is undesirable. At the same time, larger increments to limits are necessary to avoid blocking if updates are less frequent, requiring larger resource commitments at the receiver. Thus there is a trade-off between resource commitment and overhead when determining how large a limit is advertised.
-A receiver can use an autotuning mechanism to tune the frequency and amount of advertised additional credit based on a round-trip time estimate and the rate at which the receiving application consumes data, similar to common TCP implementations. As an optimization, sending frames related to flow control only when there are other frames to send or when a peer is blocked ensures that flow control doesn’t cause extra packets to be sent.
-If a sender runs out of flow control credit, it will be unable to send new data and is considered blocked. It is generally considered best to not let the sender become blocked. To avoid blocking a sender, and to reasonably account for the possibility of loss, a receiver should send a MAX_DATA or MAX_STREAM_DATA frame at least two round trips before it expects the sender to get blocked.
-A receiver MUST NOT wait for a STREAM_DATA_BLOCKED or DATA_BLOCKED frame before sending MAX_STREAM_DATA or MAX_DATA, since doing so will mean that a sender will be blocked for at least an entire round trip, and potentially for longer if the peer chooses to not send STREAM_DATA_BLOCKED or DATA_BLOCKED frames.
-Endpoints need to eventually agree on the amount of flow control credit that has been consumed, to avoid either exceeding flow control limits or deadlocking.
-On receipt of a RESET_STREAM frame, an endpoint will tear down state for the matching stream and ignore further data arriving on that stream. Without the offset included in RESET_STREAM, the two endpoints could disagree on the number of bytes that count towards connection flow control.
-To remedy this issue, a RESET_STREAM frame (Section 19.4) includes the final size of data sent on the stream. On receiving a RESET_STREAM frame, a receiver definitively knows how many bytes were sent on that stream before the RESET_STREAM frame, and the receiver MUST use the final size of the stream to account for all bytes sent on the stream in its connection level flow controller.
-RESET_STREAM terminates one direction of a stream abruptly. For a bidirectional stream, RESET_STREAM has no effect on data flow in the opposite direction. Both endpoints MUST maintain flow control state for the stream in the unterminated direction until that direction enters a terminal state, or until one of the endpoints sends CONNECTION_CLOSE.
-The final size is the amount of flow control credit that is consumed by a stream. Assuming that every contiguous byte on the stream was sent once, the final size is the number of bytes sent. More generally, this is one higher than the offset of the byte with the largest offset sent on the stream, or zero if no bytes were sent.
-For a stream that is reset, the final size is carried explicitly in a RESET_STREAM frame. Otherwise, the final size is the offset plus the length of a STREAM frame marked with a FIN flag, or 0 in the case of incoming unidirectional streams.
-An endpoint will know the final size for a stream when the receiving part of the stream enters the “Size Known” or “Reset Recvd” state (Section 3).
-An endpoint MUST NOT send data on a stream at or beyond the final size.
-Once a final size for a stream is known, it cannot change. If a RESET_STREAM or STREAM frame is received indicating a change in the final size for the stream, an endpoint SHOULD respond with a FINAL_SIZE_ERROR error (see Section 11). A receiver SHOULD treat receipt of data at or beyond the final size as a FINAL_SIZE_ERROR error, even after a stream is closed. Generating these errors is not mandatory, but only because requiring that an endpoint generate these errors also means that the endpoint needs to maintain the final size state for closed streams, which could mean a significant state commitment.
-An endpoint limits the cumulative number of incoming streams a peer can open. Only streams with a stream ID less than (max_stream * 4 + initial_stream_id_for_type) can be opened (see Table 5). Initial limits are set in the transport parameters (see Section 18.2) and subsequently limits are advertised using MAX_STREAMS frames (Section 19.11). Separate limits apply to unidirectional and bidirectional streams.
-If a max_streams transport parameter or MAX_STREAMS frame is received with a value greater than 2^60, this would allow a maximum stream ID that cannot be expressed as a variable-length integer (see Section 16). If either is received, the connection MUST be closed immediately with a connection error of type STREAM_LIMIT_ERROR (see Section 10.3).
-Endpoints MUST NOT exceed the limit set by their peer. An endpoint that receives a frame with a stream ID exceeding the limit it has sent MUST treat this as a connection error of type STREAM_LIMIT_ERROR (Section 11).
-Once a receiver advertises a stream limit using the MAX_STREAMS frame, advertising a smaller limit has no effect. A receiver MUST ignore any MAX_STREAMS frame that does not increase the stream limit.
-As with stream and connection flow control, this document leaves when and how many streams to advertise to a peer via MAX_STREAMS to implementations. Implementations might choose to increase limits as streams close to keep the number of streams available to peers roughly consistent.
-An endpoint that is unable to open a new stream due to the peer’s limits SHOULD send a STREAMS_BLOCKED frame (Section 19.14). This signal is considered useful for debugging. An endpoint MUST NOT wait to receive this signal before advertising additional credit, since doing so will mean that the peer will be blocked for at least an entire round trip, and potentially for longer if the peer chooses to not send STREAMS_BLOCKED frames.
-QUIC’s connection establishment combines version negotiation with the cryptographic and transport handshakes to reduce connection establishment latency, as described in Section 7. Once established, a connection may migrate to a different IP or port at either endpoint as described in Section 9. Finally, a connection may be terminated by either endpoint, as described in Section 10.
-Each connection possesses a set of connection identifiers, or connection IDs, each of which can identify the connection. Connection IDs are independently selected by endpoints; each endpoint selects the connection IDs that its peer uses.
-The primary function of a connection ID is to ensure that changes in addressing at lower protocol layers (UDP, IP) don’t cause packets for a QUIC connection to be delivered to the wrong endpoint. Each endpoint selects connection IDs using an implementation-specific (and perhaps deployment-specific) method which will allow packets with that connection ID to be routed back to the endpoint and identified by the endpoint upon receipt.
-Connection IDs MUST NOT contain any information that can be used by an external observer (that is, one that does not cooperate with the issuer) to correlate them with other connection IDs for the same connection. As a trivial example, this means the same connection ID MUST NOT be issued more than once on the same connection.
-Packets with long headers include Source Connection ID and Destination Connection ID fields. These fields are used to set the connection IDs for new connections; see Section 7.2 for details.
-Packets with short headers (Section 17.3) only include the Destination Connection ID and omit the explicit length. The length of the Destination Connection ID field is expected to be known to endpoints. Endpoints using a load balancer that routes based on connection ID could agree with the load balancer on a fixed length for connection IDs, or agree on an encoding scheme. A fixed portion could encode an explicit length, which allows the entire connection ID to vary in length and still be used by the load balancer.
-A Version Negotiation (Section 17.2.1) packet echoes the connection IDs selected by the client, both to ensure correct routing toward the client and to allow the client to validate that the packet is in response to an Initial packet.
-A zero-length connection ID can be used when a connection ID is not needed to route to the correct endpoint. However, multiplexing connections on the same local IP address and port while using zero-length connection IDs will cause failures in the presence of peer connection migration, NAT rebinding, and client port reuse; and therefore MUST NOT be done unless an endpoint is certain that those protocol features are not in use.
-When an endpoint has requested a non-zero-length connection ID, it needs to ensure that the peer has a supply of connection IDs from which to choose for packets sent to the endpoint. These connection IDs are supplied by the endpoint using the NEW_CONNECTION_ID frame (Section 19.15).
-Each Connection ID has an associated sequence number to assist in deduplicating messages. The initial connection ID issued by an endpoint is sent in the Source Connection ID field of the long packet header (Section 17.2) during the handshake. The sequence number of the initial connection ID is 0. If the preferred_address transport parameter is sent, the sequence number of the supplied connection ID is 1.
-Additional connection IDs are communicated to the peer using NEW_CONNECTION_ID frames (Section 19.15). The sequence number on each newly-issued connection ID MUST increase by 1. The connection ID randomly selected by the client in the Initial packet and any connection ID provided by a Retry packet are not assigned sequence numbers unless a server opts to retain them as its initial connection ID.
-When an endpoint issues a connection ID, it MUST accept packets that carry this connection ID for the duration of the connection or until its peer invalidates the connection ID via a RETIRE_CONNECTION_ID frame (Section 19.16). Connection IDs that are issued and not retired are considered active; any active connection ID can be used.
-An endpoint SHOULD ensure that its peer has a sufficient number of available and unused connection IDs. Endpoints store received connection IDs for future use and advertise the number of connection IDs they are willing to store with the active_connection_id_limit transport parameter. An endpoint SHOULD NOT provide more connection IDs than the peer’s limit.
-An endpoint SHOULD supply a new connection ID when it receives a packet with a previously unused connection ID or when the peer retires one, unless providing the new connection ID would exceed the peer’s limit. An endpoint MAY limit the frequency or the total number of connection IDs issued for each connection to avoid the risk of running out of connection IDs; see Section 10.4.2.
-An endpoint that initiates migration and requires non-zero-length connection IDs SHOULD ensure that the pool of connection IDs available to its peer allows the peer to use a new connection ID on migration, as the peer will close the connection if the pool is exhausted.
-An endpoint can change the connection ID it uses for a peer to another available one at any time during the connection. An endpoint consumes connection IDs in response to a migrating peer; see Section 9.5 for more.
-An endpoint maintains a set of connection IDs received from its peer, any of which it can use when sending packets. When the endpoint wishes to remove a connection ID from use, it sends a RETIRE_CONNECTION_ID frame to its peer. Sending a RETIRE_CONNECTION_ID frame indicates that the connection ID will not be used again and requests that the peer replace it with a new connection ID using a NEW_CONNECTION_ID frame.
-As discussed in Section 9.5, each connection ID MUST be used on packets sent from only one local address. An endpoint that migrates away from a local address SHOULD retire all connection IDs used on that address once it no longer plans to use that address.
-An endpoint can cause its peer to retire connection IDs by sending a NEW_CONNECTION_ID frame with an increased Retire Prior To field. Upon receipt, the peer MUST retire the corresponding connection IDs using RETIRE_CONNECTION_ID frames. Failure to retire the connection IDs within approximately one PTO can cause packets to be delayed, lost, or cause the original endpoint to send a stateless reset in response to a connection ID it can no longer route correctly.
-An endpoint MAY discard a connection ID for which retirement has been requested once an interval of no less than 3 PTO has elapsed since an acknowledgement is received for the NEW_CONNECTION_ID frame requesting that retirement. Until then, the endpoint SHOULD be prepared to receive packets that contain the connection ID that it has requested be retired. Subsequent incoming packets using that connection ID could elicit a response with the corresponding stateless reset token.
-Incoming packets are classified on receipt. Packets can either be associated with an existing connection, or - for servers - potentially create a new connection.
-Hosts try to associate a packet with an existing connection. If the packet has a non-zero-length Destination Connection ID corresponding to an existing connection, QUIC processes that packet accordingly. Note that more than one connection ID can be associated with a connection; see Section 5.1.
-If the Destination Connection ID is zero length and the packet matches the local address and port of a connection where the host used zero-length connection IDs, QUIC processes the packet as part of that connection.
-Endpoints can send a Stateless Reset (Section 10.4) for any packets that cannot be attributed to an existing connection. A stateless reset allows a peer to more quickly identify when a connection becomes unusable.
-Packets that are matched to an existing connection are discarded if the packets are inconsistent with the state of that connection. For example, packets are discarded if they indicate a different protocol version than that of the connection, or if the removal of packet protection is unsuccessful once the expected keys are available.
-Invalid packets without packet protection, such as Initial, Retry, or Version Negotiation, MAY be discarded. An endpoint MUST generate a connection error if it commits changes to state before discovering an error.
-Valid packets sent to clients always include a Destination Connection ID that matches a value the client selects. Clients that choose to receive zero-length connection IDs can use the local address and port to identify a connection. Packets that don’t match an existing connection are discarded.
-Due to packet reordering or loss, a client might receive packets for a connection that are encrypted with a key it has not yet computed. The client MAY drop these packets, or MAY buffer them in anticipation of later packets that allow it to compute the key.
-If a client receives a packet that has an unsupported version, it MUST discard that packet.
-If a server receives a packet that has an unsupported version, but the packet is sufficiently large to initiate a new connection for any version supported by the server, it SHOULD send a Version Negotiation packet as described in Section 6.1. Servers MAY rate control these packets to avoid storms of Version Negotiation packets. Otherwise, servers MUST drop packets that specify unsupported versions.
-The first packet for an unsupported version can use different semantics and encodings for any version-specific field. In particular, different packet protection keys might be used for different versions. Servers that do not support a particular version are unlikely to be able to decrypt the payload of the packet. Servers SHOULD NOT attempt to decode or decrypt a packet from an unknown version, but instead send a Version Negotiation packet, provided that the packet is sufficiently long.
-Packets with a supported version, or no version field, are matched to a connection using the connection ID or - for packets with zero-length connection IDs - the local address and port. If the packet doesn’t match an existing connection, the server continues below.
-If the packet is an Initial packet fully conforming with the specification, the server proceeds with the handshake (Section 7). This commits the server to the version that the client selected.
-If a server isn’t currently accepting any new connections, it SHOULD send an Initial packet containing a CONNECTION_CLOSE frame with error code SERVER_BUSY.
-If the packet is a 0-RTT packet, the server MAY buffer a limited number of these packets in anticipation of a late-arriving Initial packet. Clients are not able to send Handshake packets prior to receiving a server response, so servers SHOULD ignore any such packets.
-Servers MUST drop incoming packets under all other circumstances.
-TBD.
-There are certain operations which an application MUST be able to perform when interacting with the QUIC transport. This document does not specify an API, but any implementation of this version of QUIC MUST expose the ability to perform the operations described in this section on a QUIC connection.
-When implementing the client role, applications need to be able to:
- - +A receiver sets initial credits for all streams by sending transport parameters +during the handshake (Section 7.3). A receiver sends +MAX_STREAM_DATA (Section 19.10) or MAX_DATA (Section 19.9) +frames to the sender to advertise additional credit.¶
+A receiver advertises credit for a stream by sending a MAX_STREAM_DATA frame +with the Stream ID field set appropriately. A MAX_STREAM_DATA frame indicates +the maximum absolute byte offset of a stream. A receiver could use the current +offset of data consumed to determine the flow control offset to be advertised. +A receiver MAY send MAX_STREAM_DATA frames in multiple packets in order to make +sure that the sender receives an update before running out of flow control +credit, even if one of the packets is lost.¶
+A receiver advertises credit for a connection by sending a MAX_DATA frame, which +indicates the maximum of the sum of the absolute byte offsets of all streams. A +receiver maintains a cumulative sum of bytes received on all streams, which is +used to check for flow control violations. A receiver might use a sum of bytes +consumed on all streams to determine the maximum data limit to be advertised.¶
+A receiver can advertise a larger offset by sending MAX_STREAM_DATA or MAX_DATA +frames. Once a receiver advertises an offset, it MAY advertise a smaller +offset, but this has no effect.¶
+A receiver MUST close the connection with a FLOW_CONTROL_ERROR error +(Section 11) if the sender violates the advertised connection or stream +data limits.¶
+A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do not increase +flow control limits.¶
+If a sender runs out of flow control credit, it will be unable to send new data +and is considered blocked. A sender SHOULD send a STREAM_DATA_BLOCKED or +DATA_BLOCKED frame to indicate it has data to write but is blocked by flow +control limits. These frames are expected to be sent infrequently in common +cases, but they are considered useful for debugging and monitoring purposes.¶
+A sender SHOULD NOT send multiple STREAM_DATA_BLOCKED or DATA_BLOCKED frames +for the same data limit, unless the original frame is determined to be lost. +Another STREAM_DATA_BLOCKED or DATA_BLOCKED frame can be sent after the data +limit is increased.¶
+This document leaves when and how many bytes to advertise in a MAX_STREAM_DATA +or MAX_DATA frame to implementations, but offers a few considerations. These +frames contribute to connection overhead. Therefore frequently sending frames +with small changes is undesirable. At the same time, larger increments to +limits are necessary to avoid blocking if updates are less frequent, requiring +larger resource commitments at the receiver. Thus there is a trade-off between +resource commitment and overhead when determining how large a limit is +advertised.¶
+A receiver can use an autotuning mechanism to tune the frequency and amount of +advertised additional credit based on a round-trip time estimate and the rate at +which the receiving application consumes data, similar to common TCP +implementations. As an optimization, sending frames related to flow control +only when there are other frames to send or when a peer is blocked ensures that +flow control doesn't cause extra packets to be sent.¶
+If a sender runs out of flow control credit, it will be unable to send new data +and is considered blocked. It is generally considered best to not let the +sender become blocked. To avoid blocking a sender, and to reasonably account +for the possibility of loss, a receiver should send a MAX_DATA or +MAX_STREAM_DATA frame at least two round trips before it expects the sender to +get blocked.¶
+A receiver MUST NOT wait for a STREAM_DATA_BLOCKED or DATA_BLOCKED frame before +sending MAX_STREAM_DATA or MAX_DATA, since doing so will mean that a sender will +be blocked for at least an entire round trip, and potentially for longer if the +peer chooses to not send STREAM_DATA_BLOCKED or DATA_BLOCKED frames.¶
+Endpoints need to eventually agree on the amount of flow control credit that has +been consumed, to avoid either exceeding flow control limits or deadlocking.¶
+On receipt of a RESET_STREAM frame, an endpoint will tear down state for the +matching stream and ignore further data arriving on that stream. Without the +offset included in RESET_STREAM, the two endpoints could disagree on +the number of bytes that count towards connection flow control.¶
+To remedy this issue, a RESET_STREAM frame (Section 19.4) includes the +final size of data sent on the stream. On receiving a RESET_STREAM frame, a +receiver definitively knows how many bytes were sent on that stream before the +RESET_STREAM frame, and the receiver MUST use the final size of the stream to +account for all bytes sent on the stream in its connection level flow +controller.¶
+RESET_STREAM terminates one direction of a stream abruptly. For a bidirectional +stream, RESET_STREAM has no effect on data flow in the opposite direction. Both +endpoints MUST maintain flow control state for the stream in the unterminated +direction until that direction enters a terminal state, or until one of the +endpoints sends CONNECTION_CLOSE.¶
+The final size is the amount of flow control credit that is consumed by a +stream. Assuming that every contiguous byte on the stream was sent once, the +final size is the number of bytes sent. More generally, this is one higher +than the offset of the byte with the largest offset sent on the stream, or zero +if no bytes were sent.¶
+For a stream that is reset, the final size is carried explicitly in a +RESET_STREAM frame. Otherwise, the final size is the offset plus the length of +a STREAM frame marked with a FIN flag, or 0 in the case of incoming +unidirectional streams.¶
+An endpoint will know the final size for a stream when the receiving part of the +stream enters the "Size Known" or "Reset Recvd" state (Section 3).¶
+An endpoint MUST NOT send data on a stream at or beyond the final size.¶
+Once a final size for a stream is known, it cannot change. If a RESET_STREAM or +STREAM frame is received indicating a change in the final size for the stream, +an endpoint SHOULD respond with a FINAL_SIZE_ERROR error (see +Section 11). A receiver SHOULD treat receipt of data at or beyond the +final size as a FINAL_SIZE_ERROR error, even after a stream is closed. +Generating these errors is not mandatory, but only because requiring that an +endpoint generate these errors also means that the endpoint needs to maintain +the final size state for closed streams, which could mean a significant state +commitment.¶
+An endpoint limits the cumulative number of incoming streams a peer can open. +Only streams with a stream ID less than (max_stream * 4 + +initial_stream_id_for_type) can be opened (see Table 5). Initial +limits are set in the transport parameters (see +Section 18.2) and subsequently limits are advertised +using MAX_STREAMS frames (Section 19.11). Separate limits apply to +unidirectional and bidirectional streams.¶
+If a max_streams transport parameter or MAX_STREAMS frame is received with a +value greater than 2^60, this would allow a maximum stream ID that cannot be +expressed as a variable-length integer (see Section 16). +If either is received, the connection MUST be closed immediately with a +connection error of type STREAM_LIMIT_ERROR (see Section 10.3).¶
+Endpoints MUST NOT exceed the limit set by their peer. An endpoint that +receives a frame with a stream ID exceeding the limit it has sent MUST treat +this as a connection error of type STREAM_LIMIT_ERROR (Section 11).¶
+Once a receiver advertises a stream limit using the MAX_STREAMS frame, +advertising a smaller limit has no effect. A receiver MUST ignore any +MAX_STREAMS frame that does not increase the stream limit.¶
+As with stream and connection flow control, this document leaves when and how +many streams to advertise to a peer via MAX_STREAMS to implementations. +Implementations might choose to increase limits as streams close to keep the +number of streams available to peers roughly consistent.¶
+An endpoint that is unable to open a new stream due to the peer's limits SHOULD +send a STREAMS_BLOCKED frame (Section 19.14). This signal is +considered useful for debugging. An endpoint MUST NOT wait to receive this +signal before advertising additional credit, since doing so will mean that the +peer will be blocked for at least an entire round trip, and potentially for +longer if the peer chooses to not send STREAMS_BLOCKED frames.¶
+QUIC's connection establishment combines version negotiation with the +cryptographic and transport handshakes to reduce connection establishment +latency, as described in Section 7. Once established, a connection +may migrate to a different IP or port at either endpoint as +described in Section 9. Finally, a connection may be terminated by either +endpoint, as described in Section 10.¶
+Each connection possesses a set of connection identifiers, or connection IDs, +each of which can identify the connection. Connection IDs are independently +selected by endpoints; each endpoint selects the connection IDs that its peer +uses.¶
+The primary function of a connection ID is to ensure that changes in addressing +at lower protocol layers (UDP, IP) don't cause packets for a QUIC +connection to be delivered to the wrong endpoint. Each endpoint selects +connection IDs using an implementation-specific (and perhaps +deployment-specific) method which will allow packets with that connection ID to +be routed back to the endpoint and identified by the endpoint upon receipt.¶
+Connection IDs MUST NOT contain any information that can be used by an external +observer (that is, one that does not cooperate with the issuer) to correlate +them with other connection IDs for the same connection. As a trivial example, +this means the same connection ID MUST NOT be issued more than once on the same +connection.¶
+Packets with long headers include Source Connection ID and Destination +Connection ID fields. These fields are used to set the connection IDs for new +connections; see Section 7.2 for details.¶
+Packets with short headers (Section 17.3) only include the Destination +Connection ID and omit the explicit length. The length of the Destination +Connection ID field is expected to be known to endpoints. Endpoints using a +load balancer that routes based on connection ID could agree with the load +balancer on a fixed length for connection IDs, or agree on an encoding scheme. +A fixed portion could encode an explicit length, which allows the entire +connection ID to vary in length and still be used by the load balancer.¶
+A Version Negotiation (Section 17.2.1) packet echoes the connection IDs +selected by the client, both to ensure correct routing toward the client and to +allow the client to validate that the packet is in response to an Initial +packet.¶
+A zero-length connection ID can be used when a connection ID is not needed to +route to the correct endpoint. However, multiplexing connections on the same +local IP address and port while using zero-length connection IDs will cause +failures in the presence of peer connection migration, NAT rebinding, and client +port reuse; and therefore MUST NOT be done unless an endpoint is certain that +those protocol features are not in use.¶
+When an endpoint uses a non-zero-length connection ID, it needs to ensure that +the peer has a supply of connection IDs from which to choose for packets sent to +the endpoint. These connection IDs are supplied by the endpoint using the +NEW_CONNECTION_ID frame (Section 19.15).¶
+Each Connection ID has an associated sequence number to assist in deduplicating +messages. The initial connection ID issued by an endpoint is sent in the Source +Connection ID field of the long packet header (Section 17.2) during the +handshake. The sequence number of the initial connection ID is 0. If the +preferred_address transport parameter is sent, the sequence number of the +supplied connection ID is 1.¶
+Additional connection IDs are communicated to the peer using NEW_CONNECTION_ID +frames (Section 19.15). The sequence number on each newly-issued +connection ID MUST increase by 1. The connection ID randomly selected by the +client in the Initial packet and any connection ID provided by a Retry packet +are not assigned sequence numbers unless a server opts to retain them as its +initial connection ID.¶
+When an endpoint issues a connection ID, it MUST accept packets that carry this +connection ID for the duration of the connection or until its peer invalidates +the connection ID via a RETIRE_CONNECTION_ID frame +(Section 19.16). Connection IDs that are issued and not +retired are considered active; any active connection ID can be used.¶
+An endpoint SHOULD ensure that its peer has a sufficient number of available and +unused connection IDs. Endpoints store received connection IDs for future use +and advertise the number of connection IDs they are willing to store with the +active_connection_id_limit transport parameter. An endpoint MUST NOT provide +more connection IDs than the peer's limit. An endpoint that receives more +connection IDs than its advertised active_connection_id_limit MUST close the +connection with an error of type CONNECTION_ID_LIMIT_ERROR.¶
+An endpoint SHOULD supply a new connection ID when the peer retires a connection +ID. If an endpoint provided fewer connection IDs than the peer's +active_connection_id_limit, it MAY supply a new connection ID when it receives +a packet with a previously unused connection ID. An endpoint MAY limit the +frequency or the total number of connection IDs issued for each connection to +avoid the risk of running out of connection IDs; see Section 10.4.2.¶
+An endpoint that initiates migration and requires non-zero-length connection IDs +SHOULD ensure that the pool of connection IDs available to its peer allows the +peer to use a new connection ID on migration, as the peer will close the +connection if the pool is exhausted.¶
+An endpoint can change the connection ID it uses for a peer to another available +one at any time during the connection. An endpoint consumes connection IDs in +response to a migrating peer; see Section 9.5 for more.¶
+An endpoint maintains a set of connection IDs received from its peer, any of +which it can use when sending packets. When the endpoint wishes to remove a +connection ID from use, it sends a RETIRE_CONNECTION_ID frame to its peer. +Sending a RETIRE_CONNECTION_ID frame indicates that the connection ID will not +be used again and requests that the peer replace it with a new connection ID +using a NEW_CONNECTION_ID frame.¶
+As discussed in Section 9.5, each connection ID MUST be used on +packets sent from only one local address. An endpoint that migrates away from a +local address SHOULD retire all connection IDs used on that address once it no +longer plans to use that address.¶
+An endpoint can cause its peer to retire connection IDs by sending a +NEW_CONNECTION_ID frame with an increased Retire Prior To field. Upon receipt, +the peer MUST first retire the corresponding connection IDs using +RETIRE_CONNECTION_ID frames and then add the newly provided connection ID to the +set of active connection IDs. Failure to retire the connection IDs within +approximately one PTO can cause packets to be delayed, lost, or cause the +original endpoint to send a stateless reset in response to a connection ID it +can no longer route correctly.¶
+An endpoint MAY discard a connection ID for which retirement has been requested +once an interval of no less than 3 PTO has elapsed since an acknowledgement is +received for the NEW_CONNECTION_ID frame requesting that retirement. Until +then, the endpoint SHOULD be prepared to receive packets that contain the +connection ID that it has requested be retired. Subsequent incoming packets +using that connection ID could elicit a response with the corresponding +stateless reset token.¶
+Incoming packets are classified on receipt. Packets can either be associated +with an existing connection, or - for servers - potentially create a new +connection.¶
+Endpoints try to associate a packet with an existing connection. If the packet +has a non-zero-length Destination Connection ID corresponding to an existing +connection, QUIC processes that packet accordingly. Note that more than one +connection ID can be associated with a connection; see Section 5.1.¶
+If the Destination Connection ID is zero length and the addressing information +in the packet matches the addressing information the endpoint uses to identify a +connection with a zero-length connection ID, QUIC processes the packet as part +of that connection. An endpoint can use just destination IP and port or both +source and destination addresses for identification, though this makes +connections fragile as described in Section 5.1.¶
+Endpoints can send a Stateless Reset (Section 10.4) for any packets that +cannot be attributed to an existing connection. A stateless reset allows a peer +to more quickly identify when a connection becomes unusable.¶
+Packets that are matched to an existing connection are discarded if the packets +are inconsistent with the state of that connection. For example, packets are +discarded if they indicate a different protocol version than that of the +connection, or if the removal of packet protection is unsuccessful once the +expected keys are available.¶
+Invalid packets without packet protection, such as Initial, Retry, or Version +Negotiation, MAY be discarded. An endpoint MUST generate a connection error if +it commits changes to state before discovering an error.¶
+Valid packets sent to clients always include a Destination Connection ID that +matches a value the client selects. Clients that choose to receive +zero-length connection IDs can use the local address and port to identify a +connection. Packets that don't match an existing connection are discarded.¶
+Due to packet reordering or loss, a client might receive packets for a +connection that are encrypted with a key it has not yet computed. The client MAY +drop these packets, or MAY buffer them in anticipation of later packets that +allow it to compute the key.¶
+If a client receives a packet that has an unsupported version, it MUST discard +that packet.¶
+If a server receives a packet that has an unsupported version, but the packet is +sufficiently large to initiate a new connection for any version supported by the +server, it SHOULD send a Version Negotiation packet as described in +Section 6.1. Servers MAY rate control these packets to avoid storms of Version +Negotiation packets. Otherwise, servers MUST drop packets that specify +unsupported versions.¶
+The first packet for an unsupported version can use different semantics and +encodings for any version-specific field. In particular, different packet +protection keys might be used for different versions. Servers that do not +support a particular version are unlikely to be able to decrypt the payload of +the packet. Servers SHOULD NOT attempt to decode or decrypt a packet from an +unknown version, but instead send a Version Negotiation packet, provided that +the packet is sufficiently long.¶
+Packets with a supported version, or no version field, are matched to a +connection using the connection ID or - for packets with zero-length connection +IDs - the local address and port. If the packet doesn't match an existing +connection, the server continues below.¶
+If the packet is an Initial packet fully conforming with the specification, the +server proceeds with the handshake (Section 7). This commits the server to +the version that the client selected.¶
+If a server isn't currently accepting any new connections, it SHOULD send an +Initial packet containing a CONNECTION_CLOSE frame with error code +SERVER_BUSY.¶
+If the packet is a 0-RTT packet, the server MAY buffer a limited number of these +packets in anticipation of a late-arriving Initial packet. Clients are not able +to send Handshake packets prior to receiving a server response, so servers +SHOULD ignore any such packets.¶
+Servers MUST drop incoming packets under all other circumstances.¶
+A QUIC connection is a stateful interaction between a client and server, the +primary purpose of which is to support the exchange of data by an application +protocol. Streams (Section 2) are the primary means by which an application +protocol exchanges information.¶
+Each connection starts with a handshake phase, during which client and server +establish a shared secret using the cryptographic handshake protocol +[QUIC-TLS] and negotiate the application protocol. The handshake +(Section 7) confirms that both endpoints are willing to communicate +(Section 8.1) and establishes parameters for the connection +(Section 7.3).¶
+An application protocol can also operate in a limited fashion during the +handshake phase. 0-RTT allows application messages to be sent by a client +before receiving any messages from the server. However, 0-RTT lacks certain key +security guarantees. In particular, there is no protection against replay +attacks in 0-RTT; see [QUIC-TLS]. Separately, a server can also send +application data to a client before it receives the final cryptographic +handshake messages that allow it to confirm the identity and liveness of the +client. These capabilities allow an application protocol to offer the option to +trade some security guarantees for reduced latency.¶
+The use of connection IDs (Section 5.1) allows connections to migrate to a +new network path, both as a direct choice of an endpoint and when forced by a +change in a middlebox. Section 9 describes mitigations for the security and +privacy issues associated with migration.¶
+For connections that are no longer needed or desired, there are several ways for +a client and server to terminate a connection (Section 10).¶
+There are certain operations which an application MUST be able to perform when +interacting with the QUIC transport. This document does not specify an API, but +any implementation of this version of QUIC MUST expose the ability to perform +the operations described in this section on a QUIC connection.¶
+When implementing the client role, applications need to be able to:¶
When implementing the server role, applications need to be able to:
- - +When implementing the server role, applications need to be able to:¶
In either role, applications need to be able to:
- - +In either role, applications need to be able to:¶
Version negotiation ensures that client and server agree to a QUIC version that is mutually supported. A server sends a Version Negotiation packet in response to each packet that might initiate a new connection; see Section 5.2 for details.
-The size of the first packet sent by a client will determine whether a server sends a Version Negotiation packet. Clients that support multiple QUIC versions SHOULD pad the first packet they send to the largest of the minimum packet sizes across all versions they support. This ensures that the server responds if there is a mutually supported version.
-If the version selected by the client is not acceptable to the server, the server responds with a Version Negotiation packet (see Section 17.2.1). This includes a list of versions that the server will accept. An endpoint MUST NOT send a Version Negotiation packet in response to receiving a Version Negotiation packet.
-This system allows a server to process packets with unsupported versions without retaining state. Though either the Initial packet or the Version Negotiation packet that is sent in response could be lost, the client will send new packets until it successfully receives a response or it abandons the connection attempt. As a result, the client discards all state for the connection and does not send any more packets on the connection.
-A server MAY limit the number of Version Negotiation packets it sends. For instance, a server that is able to recognize packets as 0-RTT might choose not to send Version Negotiation packets in response to 0-RTT packets with the expectation that it will eventually receive an Initial packet.
-When a client receives a Version Negotiation packet, it MUST abandon the current connection attempt. Version Negotiation packets are designed to allow future versions of QUIC to negotiate the version in use between endpoints. Future versions of QUIC might change how implementations that support multiple versions of QUIC react to Version Negotiation packets when attempting to establish a connection using this version. How to perform version negotiation is left as future work defined by future versions of QUIC. In particular, that future work will need to ensure robustness against version downgrade attacks; see Section 21.10.
-[[RFC editor: please remove this section before publication.]]
-When a draft implementation receives a Version Negotiation packet, it MAY use it to attempt a new connection with one of the versions listed in the packet, instead of abandoning the current connection attempt; see Section 6.2.
-The client MUST check that the Destination and Source Connection ID fields match the Source and Destination Connection ID fields in a packet that the client sent. If this check fails, the packet MUST be discarded.
-Once the Version Negotiation packet is determined to be valid, the client then selects an acceptable protocol version from the list provided by the server. The client then attempts to create a new connection using that version. The new connection MUST use a new random Destination Connection ID different from the one it had previously sent.
-Note that this mechanism does not protect against downgrade attacks and MUST NOT be used outside of draft implementations.
-For a server to use a new version in the future, clients need to correctly handle unsupported versions. To help ensure this, a server SHOULD include a version that is reserved for forcing version negotiation (0x?a?a?a?a as defined in Section 15) when generating a Version Negotiation packet.
-The design of version negotiation permits a server to avoid maintaining state for packets that it rejects in this fashion.
-A client MAY send a packet using a version that is reserved for forcing version negotiation. This can be used to solicit a list of supported versions from a server.
-QUIC relies on a combined cryptographic and transport handshake to minimize connection establishment latency. QUIC uses the CRYPTO frame Section 19.6 to transmit the cryptographic handshake. Version 0x00000001 of QUIC uses TLS as described in [QUIC-TLS]; a different QUIC version number could indicate that a different cryptographic handshake protocol is in use.
-QUIC provides reliable, ordered delivery of the cryptographic handshake data. QUIC packet protection is used to encrypt as much of the handshake protocol as possible. The cryptographic handshake MUST provide the following properties:
- - +Version negotiation ensures that client and server agree to a QUIC version +that is mutually supported. A server sends a Version Negotiation packet in +response to each packet that might initiate a new connection; see +Section 5.2 for details.¶
+The size of the first packet sent by a client will determine whether a server +sends a Version Negotiation packet. Clients that support multiple QUIC versions +SHOULD pad the first packet they send to the largest of the minimum packet sizes +across all versions they support. This ensures that the server responds if there +is a mutually supported version.¶
+If the version selected by the client is not acceptable to the server, the +server responds with a Version Negotiation packet (see Section 17.2.1). +This includes a list of versions that the server will accept. An endpoint MUST +NOT send a Version Negotiation packet in response to receiving a Version +Negotiation packet.¶
+This system allows a server to process packets with unsupported versions without +retaining state. Though either the Initial packet or the Version Negotiation +packet that is sent in response could be lost, the client will send new packets +until it successfully receives a response or it abandons the connection attempt. +As a result, the client discards all state for the connection and does not send +any more packets on the connection.¶
+A server MAY limit the number of Version Negotiation packets it sends. For +instance, a server that is able to recognize packets as 0-RTT might choose not +to send Version Negotiation packets in response to 0-RTT packets with the +expectation that it will eventually receive an Initial packet.¶
+When a client receives a Version Negotiation packet, it MUST abandon the +current connection attempt. Version Negotiation packets are designed to allow +future versions of QUIC to negotiate the version in use between endpoints. +Future versions of QUIC might change how implementations that support multiple +versions of QUIC react to Version Negotiation packets when attempting to +establish a connection using this version. How to perform version negotiation +is left as future work defined by future versions of QUIC. In particular, +that future work will need to ensure robustness against version downgrade +attacks; see Section 21.10.¶
+[[RFC editor: please remove this section before publication.]]¶
+When a draft implementation receives a Version Negotiation packet, it MAY use +it to attempt a new connection with one of the versions listed in the packet, +instead of abandoning the current connection attempt; see Section 6.2.¶
+The client MUST check that the Destination and Source Connection ID fields +match the Source and Destination Connection ID fields in a packet that the +client sent. If this check fails, the packet MUST be discarded.¶
+Once the Version Negotiation packet is determined to be valid, the client then +selects an acceptable protocol version from the list provided by the server. +The client then attempts to create a new connection using that version. The new +connection MUST use a new random Destination Connection ID different from the +one it had previously sent.¶
+Note that this mechanism does not protect against downgrade attacks and +MUST NOT be used outside of draft implementations.¶
+For a server to use a new version in the future, clients need to correctly +handle unsupported versions. To help ensure this, a server SHOULD include a +version that is reserved for forcing version negotiation (0x?a?a?a?a as defined +in Section 15) when generating a Version Negotiation packet.¶
+The design of version negotiation permits a server to avoid maintaining state +for packets that it rejects in this fashion.¶
+A client MAY send a packet using a version that is reserved for forcing version +negotiation. This can be used to solicit a list of supported versions from a +server.¶
+QUIC relies on a combined cryptographic and transport handshake to minimize +connection establishment latency. QUIC uses the CRYPTO frame Section 19.6 +to transmit the cryptographic handshake. Version 0x00000001 of QUIC uses TLS as +described in [QUIC-TLS]; a different QUIC version number could indicate that a +different cryptographic handshake protocol is in use.¶
+QUIC provides reliable, ordered delivery of the cryptographic handshake +data. QUIC packet protection is used to encrypt as much of the handshake +protocol as possible. The cryptographic handshake MUST provide the following +properties:¶
authenticated key exchange, where¶
+An endpoint can verify support for Explicit Congestion Notification (ECN) in the first packets it sends, as described in Section 13.4.2.
-The CRYPTO frame can be sent in different packet number spaces. The sequence numbers used by CRYPTO frames to ensure ordered delivery of cryptographic handshake data start from zero in each packet number space.
-Endpoints MUST explicitly negotiate an application protocol. This avoids situations where there is a disagreement about the protocol that is in use.
-Details of how TLS is integrated with QUIC are provided in [QUIC-TLS], but some examples are provided here. An extension of this exchange to support client address validation is shown in Section 8.1.1.
-Once any address validation exchanges are complete, the cryptographic handshake is used to agree on cryptographic keys. The cryptographic handshake is carried in Initial (Section 17.2.2) and Handshake (Section 17.2.4) packets.
-Figure 3 provides an overview of the 1-RTT handshake. Each line shows a QUIC packet with the packet type and packet number shown first, followed by the frames that are typically contained in those packets. So, for instance the first packet is of type Initial, with packet number 0, and contains a CRYPTO frame carrying the ClientHello.
-Note that multiple QUIC packets – even of different encryption levels – may be coalesced into a single UDP datagram (see Section 12.2), and so this handshake may consist of as few as 4 UDP datagrams, or any number more. For instance, the server’s first flight contains packets from the Initial encryption level (obfuscation), the Handshake level, and “0.5-RTT data” from the server at the 1-RTT encryption level.
- - +An endpoint can verify support for Explicit Congestion Notification (ECN) in the +first packets it sends, as described in Section 13.4.2.¶
+The CRYPTO frame can be sent in different packet number spaces. The sequence +numbers used by CRYPTO frames to ensure ordered delivery of cryptographic +handshake data start from zero in each packet number space.¶
+Endpoints MUST explicitly negotiate an application protocol. This avoids +situations where there is a disagreement about the protocol that is in use.¶
+Details of how TLS is integrated with QUIC are provided in [QUIC-TLS], but +some examples are provided here. An extension of this exchange to support +client address validation is shown in Section 8.1.2.¶
+Once any address validation exchanges are complete, the +cryptographic handshake is used to agree on cryptographic keys. The +cryptographic handshake is carried in Initial (Section 17.2.2) and Handshake +(Section 17.2.4) packets.¶
+Figure 3 provides an overview of the 1-RTT handshake. Each line +shows a QUIC packet with the packet type and packet number shown first, followed +by the frames that are typically contained in those packets. So, for instance +the first packet is of type Initial, with packet number 0, and contains a CRYPTO +frame carrying the ClientHello.¶
+Note that multiple QUIC packets - even of different encryption levels - may be +coalesced into a single UDP datagram (see Section 12.2), and so this +handshake may consist of as few as 4 UDP datagrams, or any number more. For +instance, the server's first flight contains packets from the Initial encryption +level (obfuscation), the Handshake level, and "0.5-RTT data" from the server at +the 1-RTT encryption level.¶
+Client Server @@ -1546,10 +3022,18 @@-1-RTT[1]: STREAM[3, "..."], ACK[0] <- Handshake[1]: ACK[0]
Figure 3: Example 1-RTT Handshake
-Figure 4 shows an example of a connection with a 0-RTT handshake and a single packet of 0-RTT data. Note that as described in Section 12.3, the server acknowledges 0-RTT data at the 1-RTT encryption level, and the client sends 1-RTT packets in the same packet number space.
- - +Figure 4 shows an example of a connection with a 0-RTT handshake +and a single packet of 0-RTT data. Note that as described in Section 12.3, +the server acknowledges 0-RTT data at the 1-RTT encryption level, and the +client sends 1-RTT packets in the same packet number space.¶
+Client Server @@ -1567,84 +3051,289 @@-1-RTT[1]: STREAM[3, "..."], ACK[1] <- Handshake[1]: ACK[0]
Figure 4: Example 0-RTT Handshake
-A connection ID is used to ensure consistent routing of packets, as described in Section 5.1. The long header contains two connection IDs: the Destination Connection ID is chosen by the recipient of the packet and is used to provide consistent routing; the Source Connection ID is used to set the Destination Connection ID used by the peer.
-During the handshake, packets with the long header (Section 17.2) are used to establish the connection ID that each endpoint uses. Each endpoint uses the Source Connection ID field to specify the connection ID that is used in the Destination Connection ID field of packets being sent to them. Upon receiving a packet, each endpoint sets the Destination Connection ID it sends to match the value of the Source Connection ID that they receive.
-When an Initial packet is sent by a client that has not previously received an Initial or Retry packet from the server, it populates the Destination Connection ID field with an unpredictable value. This MUST be at least 8 bytes in length. Until a packet is received from the server, the client MUST use the same value unless it abandons the connection attempt and starts a new one. The initial Destination Connection ID is used to determine packet protection keys for Initial packets.
-The client populates the Source Connection ID field with a value of its choosing and sets the SCID Len field to indicate the length.
-The first flight of 0-RTT packets use the same Destination and Source Connection ID values as the client’s first Initial.
-Upon first receiving an Initial or Retry packet from the server, the client uses the Source Connection ID supplied by the server as the Destination Connection ID for subsequent packets, including any subsequent 0-RTT packets. That means that a client might change the Destination Connection ID twice during connection establishment, once in response to a Retry and once in response to the first Initial packet from the server. Once a client has received an Initial packet from the server, it MUST discard any packet it receives with a different Source Connection ID.
-A client MUST only change the value it sends in the Destination Connection ID in response to the first packet of each type it receives from the server (Retry or Initial); a server MUST set its value based on the Initial packet. Any additional changes are not permitted; if subsequent packets of those types include a different Source Connection ID, they MUST be discarded. This avoids problems that might arise from stateless processing of multiple Initial packets producing different connection IDs.
-The connection ID can change over the lifetime of a connection, especially in response to connection migration (Section 9); see Section 5.1.1 for details.
-During connection establishment, both endpoints make authenticated declarations of their transport parameters. These declarations are made unilaterally by each endpoint. Endpoints are required to comply with the restrictions implied by these parameters; the description of each parameter includes rules for its handling.
-The encoding of the transport parameters is detailed in Section 18.
-QUIC includes the encoded transport parameters in the cryptographic handshake. Once the handshake completes, the transport parameters declared by the peer are available. Each endpoint validates the value provided by its peer.
-Definitions for each of the defined transport parameters are included in Section 18.2.
-An endpoint MUST treat receipt of a transport parameter with an invalid value as a connection error of type TRANSPORT_PARAMETER_ERROR.
-An endpoint MUST NOT send a parameter more than once in a given transport parameters extension. An endpoint SHOULD treat receipt of duplicate transport parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.
-A server MUST include the original_connection_id transport parameter (Section 18.2) if it sent a Retry packet to enable validation of the Retry, as described in Section 17.2.5.
-Both endpoints store the value of the server transport parameters from a connection and apply them to any 0-RTT packets that are sent in subsequent connections to that peer, except for transport parameters that are explicitly excluded. Remembered transport parameters apply to the new connection until the handshake completes and the client starts sending 1-RTT packets. Once the handshake completes, the client uses the transport parameters established in the handshake.
-The definition of new transport parameters (Section 7.3.2) MUST specify whether they MUST, MAY, or MUST NOT be stored for 0-RTT. A client need not store a transport parameter it cannot process.
-A client MUST NOT use remembered values for the following parameters: original_connection_id, preferred_address, stateless_reset_token, ack_delay_exponent and active_connection_id_limit. The client MUST use the server’s new values in the handshake instead, and absent new values from the server, the default value.
-A client that attempts to send 0-RTT data MUST remember all other transport parameters used by the server. The server can remember these transport parameters, or store an integrity-protected copy of the values in the ticket and recover the information when accepting 0-RTT data. A server uses the transport parameters in determining whether to accept 0-RTT data.
-If 0-RTT data is accepted by the server, the server MUST NOT reduce any limits or alter any values that might be violated by the client with its 0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT set values for the following parameters (Section 18.2) that are smaller than the remembered value of the parameters.
- - +A connection ID is used to ensure consistent routing of packets, as described in +Section 5.1. The long header contains two connection IDs: the Destination +Connection ID is chosen by the recipient of the packet and is used to provide +consistent routing; the Source Connection ID is used to set the Destination +Connection ID used by the peer.¶
+During the handshake, packets with the long header (Section 17.2) are used to +establish the connection ID that each endpoint uses. Each endpoint uses the +Source Connection ID field to specify the connection ID that is used in the +Destination Connection ID field of packets being sent to them. Upon receiving a +packet, each endpoint sets the Destination Connection ID it sends to match the +value of the Source Connection ID that they receive.¶
+When an Initial packet is sent by a client that has not previously received an +Initial or Retry packet from the server, it populates the Destination Connection +ID field with an unpredictable value. This MUST be at least 8 bytes in length. +Until a packet is received from the server, the client MUST use the same value +unless it abandons the connection attempt and starts a new one. The initial +Destination Connection ID is used to determine packet protection keys for +Initial packets.¶
+The client populates the Source Connection ID field with a value of its choosing +and sets the SCID Len field to indicate the length.¶
+The first flight of 0-RTT packets use the same Destination and Source Connection +ID values as the client's first Initial.¶
+Upon first receiving an Initial or Retry packet from the server, the client uses +the Source Connection ID supplied by the server as the Destination Connection ID +for subsequent packets, including any subsequent 0-RTT packets. That means that +a client might change the Destination Connection ID twice during connection +establishment, once in response to a Retry and once in response to the first +Initial packet from the server. Once a client has received an Initial packet +from the server, it MUST discard any packet it receives with a different Source +Connection ID.¶
+A client MUST only change the value it sends in the Destination Connection ID in +response to the first packet of each type it receives from the server (Retry or +Initial); a server MUST set its value based on the Initial packet. Any +additional changes are not permitted; if subsequent packets of those types +include a different Source Connection ID, they MUST be discarded. This avoids +problems that might arise from stateless processing of multiple Initial packets +producing different connection IDs.¶
+The connection ID can change over the lifetime of a connection, especially in +response to connection migration (Section 9); see Section 5.1.1 for details.¶
+During connection establishment, both endpoints make authenticated declarations +of their transport parameters. These declarations are made unilaterally by each +endpoint. Endpoints are required to comply with the restrictions implied by +these parameters; the description of each parameter includes rules for its +handling.¶
+The encoding of the transport parameters is detailed in +Section 18.¶
+QUIC includes the encoded transport parameters in the cryptographic handshake. +Once the handshake completes, the transport parameters declared by the peer are +available. Each endpoint validates the value provided by its peer.¶
+Definitions for each of the defined transport parameters are included in +Section 18.2.¶
+An endpoint MUST treat receipt of a transport parameter with an invalid value as +a connection error of type TRANSPORT_PARAMETER_ERROR.¶
+An endpoint MUST NOT send a parameter more than once in a given transport +parameters extension. An endpoint SHOULD treat receipt of duplicate transport +parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.¶
+A server MUST include the original_connection_id transport parameter +(Section 18.2) if it sent a Retry packet to enable +validation of the Retry, as described in Section 17.2.5.¶
+Both endpoints store the value of the server transport parameters from +a connection and apply them to any 0-RTT packets that are sent in +subsequent connections to that peer, except for transport parameters that +are explicitly excluded. Remembered transport parameters apply to the new +connection until the handshake completes and the client starts sending +1-RTT packets. Once the handshake completes, the client uses the transport +parameters established in the handshake.¶
+The definition of new transport parameters (Section 7.3.2) MUST +specify whether they MUST, MAY, or MUST NOT be stored for 0-RTT. A client need +not store a transport parameter it cannot process.¶
+A client MUST NOT use remembered values for the following parameters: +original_connection_id, preferred_address, stateless_reset_token, +ack_delay_exponent and active_connection_id_limit. The client MUST use the +server's new values in the handshake instead, and absent new values from the +server, the default value.¶
+A client that attempts to send 0-RTT data MUST remember all other transport +parameters used by the server. The server can remember these transport +parameters, or store an integrity-protected copy of the values in the ticket +and recover the information when accepting 0-RTT data. A server uses the +transport parameters in determining whether to accept 0-RTT data.¶
+If 0-RTT data is accepted by the server, the server MUST NOT reduce any +limits or alter any values that might be violated by the client with its +0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT set +values for the following parameters (Section 18.2) +that are smaller than the remembered value of the parameters.¶
Omitting or setting a zero value for certain transport parameters can result in 0-RTT data being enabled, but not usable. The applicable subset of transport parameters that permit sending of application data SHOULD be set to non-zero values for 0-RTT. This includes initial_max_data and either initial_max_streams_bidi and initial_max_stream_data_bidi_remote, or initial_max_streams_uni and initial_max_stream_data_uni.
-A server MUST either reject 0-RTT data or abort a handshake if the implied values for transport parameters cannot be supported.
-When sending frames in 0-RTT packets, a client MUST only use remembered transport parameters; importantly, it MUST NOT use updated values that it learns from the server’s updated transport parameters or from frames received in 1-RTT packets. Updated values of transport parameters from the handshake apply only to 1-RTT packets. For instance, flow control limits from remembered transport parameters apply to all 0-RTT packets even if those values are increased by the handshake or by frames sent in 1-RTT packets. A server MAY treat use of updated transport parameters in 0-RTT as a connection error of type PROTOCOL_VIOLATION.
-New transport parameters can be used to negotiate new protocol behavior. An endpoint MUST ignore transport parameters that it does not support. Absence of a transport parameter therefore disables any optional protocol feature that is negotiated using the parameter. As described in Section 18.1, some identifiers are reserved in order to exercise this requirement.
-New transport parameters can be registered according to the rules in Section 22.1.
-Implementations need to maintain a buffer of CRYPTO data received out of order. Because there is no flow control of CRYPTO frames, an endpoint could potentially force its peer to buffer an unbounded amount of data.
-Implementations MUST support buffering at least 4096 bytes of data received in CRYPTO frames out of order. Endpoints MAY choose to allow more data to be buffered during the handshake. A larger limit during the handshake could allow for larger keys or credentials to be exchanged. An endpoint’s buffer size does not need to remain constant during the life of the connection.
-Being unable to buffer CRYPTO frames during the handshake can lead to a connection failure. If an endpoint’s buffer is exceeded during the handshake, it can expand its buffer temporarily to complete the handshake. If an endpoint does not expand its buffer, it MUST close the connection with a CRYPTO_BUFFER_EXCEEDED error code.
-Once the handshake completes, if an endpoint is unable to buffer all data in a CRYPTO frame, it MAY discard that CRYPTO frame and all CRYPTO frames received in the future, or it MAY close the connection with a CRYPTO_BUFFER_EXCEEDED error code. Packets containing discarded CRYPTO frames MUST be acknowledged because the packet has been received and processed by the transport even though the CRYPTO frame was discarded.
-Address validation is used by QUIC to avoid being used for a traffic amplification attack. In such an attack, a packet is sent to a server with spoofed source address information that identifies a victim. If a server generates more or larger packets in response to that packet, the attacker can use the server to send more data toward the victim than it would be able to send on its own.
-The primary defense against amplification attack is verifying that an endpoint is able to receive packets at the transport address that it claims. Address validation is performed both during connection establishment (see Section 8.1) and during connection migration (see Section 8.2).
-Connection establishment implicitly provides address validation for both endpoints. In particular, receipt of a packet protected with Handshake keys confirms that the client received the Initial packet from the server. Once the server has successfully processed a Handshake packet from the client, it can consider the client address to have been validated.
-Prior to validating the client address, servers MUST NOT send more than three times as many bytes as the number of bytes they have received. This limits the magnitude of any amplification attack that can be mounted using spoofed source addresses. In determining this limit, servers only count the size of successfully processed packets.
-Clients MUST ensure that UDP datagrams containing Initial packets have UDP payloads of at least 1200 bytes, adding padding to packets in the datagram as necessary. Sending padded datagrams ensures that the server is not overly constrained by the amplification restriction.
-Packet loss, in particular loss of a Handshake packet from the server, can cause a situation in which the server cannot send when the client has no data to send and the anti-amplification limit is reached. In order to avoid this causing a handshake deadlock, clients SHOULD send a packet upon a probe timeout, as described in [QUIC-RECOVERY]. If the client has no data to retransmit and does not have Handshake keys, it SHOULD send an Initial packet in a UDP datagram of at least 1200 bytes. If the client has Handshake keys, it SHOULD send a Handshake packet.
-A server might wish to validate the client address before starting the cryptographic handshake. QUIC uses a token in the Initial packet to provide address validation prior to completing the handshake. This token is delivered to the client during connection establishment with a Retry packet (see Section 8.1.1) or in a previous connection using the NEW_TOKEN frame (see Section 8.1.2).
-In addition to sending limits imposed prior to address validation, servers are also constrained in what they can send by the limits set by the congestion controller. Clients are only constrained by the congestion controller.
-Upon receiving the client’s Initial packet, the server can request address validation by sending a Retry packet (Section 17.2.5) containing a token. This token MUST be repeated by the client in all Initial packets it sends for that connection after it receives the Retry packet. In response to processing an Initial containing a token, a server can either abort the connection or permit it to proceed.
-As long as it is not possible for an attacker to generate a valid token for its own address (see Section 8.1.3) and the client is able to return that token, it proves to the server that it received the token.
-A server can also use a Retry packet to defer the state and processing costs of connection establishment. Requiring the server to provide a different connection ID, along with the original_connection_id transport parameter defined in Section 18.2, forces the server to demonstrate that it, or an entity it cooperates with, received the original Initial packet from the client. Providing a different connection ID also grants a server some control over how subsequent packets are routed. This can be used to direct connections to a different server instance.
-A flow showing the use of a Retry packet is shown in Figure 5.
- - +Omitting or setting a zero value for certain transport parameters can result in +0-RTT data being enabled, but not usable. The applicable subset of transport +parameters that permit sending of application data SHOULD be set to non-zero +values for 0-RTT. This includes initial_max_data and either +initial_max_streams_bidi and initial_max_stream_data_bidi_remote, or +initial_max_streams_uni and initial_max_stream_data_uni.¶
+A server MUST either reject 0-RTT data or abort a handshake if the implied +values for transport parameters cannot be supported.¶
+When sending frames in 0-RTT packets, a client MUST only use remembered +transport parameters; importantly, it MUST NOT use updated values that it learns +from the server's updated transport parameters or from frames received in 1-RTT +packets. Updated values of transport parameters from the handshake apply only +to 1-RTT packets. For instance, flow control limits from remembered transport +parameters apply to all 0-RTT packets even if those values are increased by the +handshake or by frames sent in 1-RTT packets. A server MAY treat use of updated +transport parameters in 0-RTT as a connection error of type PROTOCOL_VIOLATION.¶
+New transport parameters can be used to negotiate new protocol behavior. An +endpoint MUST ignore transport parameters that it does not support. Absence of +a transport parameter therefore disables any optional protocol feature that is +negotiated using the parameter. As described in Section 18.1, +some identifiers are reserved in order to exercise this requirement.¶
+New transport parameters can be registered according to the rules in +Section 22.2.¶
+Implementations need to maintain a buffer of CRYPTO data received out of order. +Because there is no flow control of CRYPTO frames, an endpoint could +potentially force its peer to buffer an unbounded amount of data.¶
+Implementations MUST support buffering at least 4096 bytes of data received in +CRYPTO frames out of order. Endpoints MAY choose to allow more data to be +buffered during the handshake. A larger limit during the handshake could allow +for larger keys or credentials to be exchanged. An endpoint's buffer size does +not need to remain constant during the life of the connection.¶
+Being unable to buffer CRYPTO frames during the handshake can lead to a +connection failure. If an endpoint's buffer is exceeded during the handshake, it +can expand its buffer temporarily to complete the handshake. If an endpoint +does not expand its buffer, it MUST close the connection with a +CRYPTO_BUFFER_EXCEEDED error code.¶
+Once the handshake completes, if an endpoint is unable to buffer all data in a +CRYPTO frame, it MAY discard that CRYPTO frame and all CRYPTO frames received in +the future, or it MAY close the connection with a CRYPTO_BUFFER_EXCEEDED error +code. Packets containing discarded CRYPTO frames MUST be acknowledged because +the packet has been received and processed by the transport even though the +CRYPTO frame was discarded.¶
+Address validation is used by QUIC to avoid being used for a traffic +amplification attack. In such an attack, a packet is sent to a server with +spoofed source address information that identifies a victim. If a server +generates more or larger packets in response to that packet, the attacker can +use the server to send more data toward the victim than it would be able to send +on its own.¶
+The primary defense against amplification attack is verifying that an endpoint +is able to receive packets at the transport address that it claims. Address +validation is performed both during connection establishment (see +Section 8.1) and during connection migration (see +Section 8.2).¶
+Connection establishment implicitly provides address validation for both +endpoints. In particular, receipt of a packet protected with Handshake keys +confirms that the client received the Initial packet from the server. Once the +server has successfully processed a Handshake packet from the client, it can +consider the client address to have been validated.¶
+Prior to validating the client address, servers MUST NOT send more than three +times as many bytes as the number of bytes they have received. This limits the +magnitude of any amplification attack that can be mounted using spoofed source +addresses. In determining this limit, servers only count the size of +successfully processed packets.¶
+Clients MUST ensure that UDP datagrams containing Initial packets have UDP +payloads of at least 1200 bytes, adding padding to packets in the datagram as +necessary. Sending padded datagrams ensures that the server is not overly +constrained by the amplification restriction.¶
+Packet loss, in particular loss of a Handshake packet from the server, can cause +a situation in which the server cannot send when the client has no data to send +and the anti-amplification limit is reached. In order to avoid this causing a +handshake deadlock, clients SHOULD send a packet upon a probe timeout, as +described in [QUIC-RECOVERY]. If the client has no data to retransmit and does +not have Handshake keys, it SHOULD send an Initial packet in a UDP datagram of +at least 1200 bytes. If the client has Handshake keys, it SHOULD send a +Handshake packet.¶
+A server might wish to validate the client address before starting the +cryptographic handshake. QUIC uses a token in the Initial packet to provide +address validation prior to completing the handshake. This token is delivered to +the client during connection establishment with a Retry packet (see +Section 8.1.2) or in a previous connection using the NEW_TOKEN frame (see +Section 8.1.3).¶
+In addition to sending limits imposed prior to address validation, servers are +also constrained in what they can send by the limits set by the congestion +controller. Clients are only constrained by the congestion controller.¶
+A token sent in a NEW_TOKEN frames or a Retry packet MUST be constructed in a +way that allows the server to identity how it was provided to a client. These +tokens are carried in the same field, but require different handling from +servers.¶
+Upon receiving the client's Initial packet, the server can request address +validation by sending a Retry packet (Section 17.2.5) containing a token. This +token MUST be repeated by the client in all Initial packets it sends for that +connection after it receives the Retry packet. In response to processing an +Initial containing a token, a server can either abort the connection or permit +it to proceed.¶
+As long as it is not possible for an attacker to generate a valid token for +its own address (see Section 8.1.4) and the client is able to return +that token, it proves to the server that it received the token.¶
+A server can also use a Retry packet to defer the state and processing costs of +connection establishment. Requiring the server to provide a different +connection ID, along with the original_connection_id transport parameter defined +in Section 18.2, forces the server to demonstrate that +it, or an entity it cooperates with, received the original Initial packet from +the client. Providing a different connection ID also grants a server some +control over how subsequent packets are routed. This can be used to direct +connections to a different server instance.¶
+If a server receives a client Initial that can be unprotected but contains an +invalid Retry token, it knows the client will not accept another Retry token. +The server can discard such a packet and allow the client to time out to +detect handshake failure, but that could impose a significant latency penalty on +the client. A server MAY proceed with the connection without verifying the +token, though the server MUST NOT consider the client address validated. If a +server chooses not to proceed with the handshake, it SHOULD immediately close +(Section 10.3) the connection with an INVALID_TOKEN error. Note that a +server has not established any state for the connection at this point and so +does not enter the closing period.¶
+A flow showing the use of a Retry packet is shown in Figure 5.¶
+Client Server @@ -1658,235 +3347,824 @@-Handshake[0]: CRYPTO[EE, CERT, CV, FIN] <- 1-RTT[0]: STREAM[1, "..."]
Figure 5: Example Handshake with Retry
-A server MAY provide clients with an address validation token during one connection that can be used on a subsequent connection. Address validation is especially important with 0-RTT because a server potentially sends a significant amount of data to a client in response to 0-RTT data.
-The server uses the NEW_TOKEN frame Section 19.7 to provide the client with an address validation token that can be used to validate future connections. The client includes this token in Initial packets to provide address validation in a future connection. The client MUST include the token in all Initial packets it sends, unless a Retry replaces the token with a newer one. The client MUST NOT use the token provided in a Retry for future connections. Servers MAY discard any Initial packet that does not carry the expected token.
-A token SHOULD be constructed in a way that allows the server to distinguish it from tokens that are sent in Retry packets as they are carried in the same field.
-The token MUST NOT include information that would allow it to be linked by an on-path observer to the connection on which it was issued. For example, it cannot include the connection ID or addressing information unless the values are encrypted.
-Unlike the token that is created for a Retry packet, there might be some time between when the token is created and when the token is subsequently used. Thus, a token SHOULD have an expiration time, which could be either an explicit expiration time or an issued timestamp that can be used to dynamically calculate the expiration time. A server can store the expiration time or include it in an encrypted form in the token.
-It is unlikely that the client port number is the same on two different connections; validating the port is therefore unlikely to be successful.
-If the client has a token received in a NEW_TOKEN frame on a previous connection to what it believes to be the same server, it SHOULD include that value in the Token field of its Initial packet. Including a token might allow the server to validate the client address without an additional round trip.
-A token allows a server to correlate activity between the connection where the token was issued and any connection where it is used. Clients that want to break continuity of identity with a server MAY discard tokens provided using the NEW_TOKEN frame. A token obtained in a Retry packet MUST be used immediately during the connection attempt and cannot be used in subsequent connection attempts.
-A client SHOULD NOT reuse a token in different connections. Reusing a token allows connections to be linked by entities on the network path; see Section 9.5. A client MUST NOT reuse a token if it believes that its point of network attachment has changed since the token was last used; that is, if there is a change in its local IP address or network interface. A client needs to start the connection process over if there is any change in its local address prior to completing the handshake.
-Clients might receive multiple tokens on a single connection. Aside from preventing linkability, any token can be used in any connection attempt. Servers can send additional tokens to either enable address validation for multiple connection attempts or to replace older tokens that might become invalid. For a client, this ambiguity means that sending the most recent unused token is most likely to be effective. Though saving and using older tokens has no negative consequences, clients can regard older tokens as being less likely be useful to the server for address validation.
-When a server receives an Initial packet with an address validation token, it MUST attempt to validate the token, unless it has already completed address validation. If the token is invalid then the server SHOULD proceed as if the client did not have a validated address, including potentially sending a Retry. If the validation succeeds, the server SHOULD then allow the handshake to proceed.
- - -In a stateless design, a server can use encrypted and authenticated tokens to pass information to clients that the server can later recover and use to validate a client address. Tokens are not integrated into the cryptographic handshake and so they are not authenticated. For instance, a client might be able to reuse a token. To avoid attacks that exploit this property, a server can limit its use of tokens to only the information needed to validate client addresses.
-Attackers could replay tokens to use servers as amplifiers in DDoS attacks. To protect against such attacks, servers SHOULD ensure that tokens sent in Retry packets are only accepted for a short time. Tokens that are provided in NEW_TOKEN frames (see Section 19.7) need to be valid for longer, but SHOULD NOT be accepted multiple times in a short period. Servers are encouraged to allow tokens to be used only once, if possible.
-An address validation token MUST be difficult to guess. Including a large enough random value in the token would be sufficient, but this depends on the server remembering the value it sends to clients.
-A token-based scheme allows the server to offload any state associated with validation to the client. For this design to work, the token MUST be covered by integrity protection against modification or falsification by clients. Without integrity protection, malicious clients could generate or guess values for tokens that would be accepted by the server. Only the server requires access to the integrity protection key for tokens.
-There is no need for a single well-defined format for the token because the server that generates the token also consumes it. A token could include information about the claimed client address (IP and port), a timestamp, and any other supplementary information the server will need to validate the token in the future.
-Path validation is used during connection migration (see Section 9 and Section 9.6) by the migrating endpoint to verify reachability of a peer from a new local address. In path validation, endpoints test reachability between a specific local address and a specific peer address, where an address is the two-tuple of IP address and port.
-Path validation tests that packets (PATH_CHALLENGE) can be both sent to and received (PATH_RESPONSE) from a peer on the path. Importantly, it validates that the packets received from the migrating endpoint do not carry a spoofed source address.
-Path validation can be used at any time by either endpoint. For instance, an endpoint might check that a peer is still in possession of its address after a period of quiescence.
-Path validation is not designed as a NAT traversal mechanism. Though the mechanism described here might be effective for the creation of NAT bindings that support NAT traversal, the expectation is that one or other peer is able to receive packets without first having sent a packet on that path. Effective NAT traversal needs additional synchronization mechanisms that are not provided here.
-An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that are used for path validation with other frames. In particular, an endpoint may pad a packet carrying a PATH_CHALLENGE for PMTU discovery, or an endpoint may bundle a PATH_RESPONSE with its own PATH_CHALLENGE.
-When probing a new path, an endpoint might want to ensure that its peer has an unused connection ID available for responses. The endpoint can send NEW_CONNECTION_ID and PATH_CHALLENGE frames in the same packet. This ensures that an unused connection ID will be available to the peer when sending a response.
-To initiate path validation, an endpoint sends a PATH_CHALLENGE frame containing a random payload on the path to be validated.
-An endpoint MAY send multiple PATH_CHALLENGE frames to guard against packet loss. However, an endpoint SHOULD NOT send multiple PATH_CHALLENGE frames in a single packet. An endpoint SHOULD NOT send a PATH_CHALLENGE more frequently than it would an Initial packet, ensuring that connection migration is no more load on a new path than establishing a new connection.
-The endpoint MUST use unpredictable data in every PATH_CHALLENGE frame so that it can associate the peer’s response with the corresponding PATH_CHALLENGE.
-On receiving a PATH_CHALLENGE frame, an endpoint MUST respond immediately by echoing the data contained in the PATH_CHALLENGE frame in a PATH_RESPONSE frame.
-An endpoint MUST NOT send more than one PATH_RESPONSE frame in response to one PATH_CHALLENGE frame (see Section 13.3). The peer is expected to send more PATH_CHALLENGE frames as necessary to evoke additional PATH_RESPONSE frames.
-A new address is considered valid when a PATH_RESPONSE frame is received that contains the data that was sent in a previous PATH_CHALLENGE. Receipt of an acknowledgment for a packet containing a PATH_CHALLENGE frame is not adequate validation, since the acknowledgment can be spoofed by a malicious peer.
-Note that receipt on a different local address does not result in path validation failure, as it might be a result of a forwarded packet (see Section 9.3.3) or misrouting. It is possible that a valid PATH_RESPONSE might be received in the future.
-Path validation only fails when the endpoint attempting to validate the path abandons its attempt to validate the path.
-Endpoints SHOULD abandon path validation based on a timer. When setting this timer, implementations are cautioned that the new path could have a longer round-trip time than the original. A value of three times the larger of the current Probe Timeout (PTO) or the initial timeout (that is, 2*kInitialRtt) as defined in [QUIC-RECOVERY] is RECOMMENDED. That is:
-+
A server MAY provide clients with an address validation token during one +connection that can be used on a subsequent connection. Address validation is +especially important with 0-RTT because a server potentially sends a significant +amount of data to a client in response to 0-RTT data.¶
+The server uses the NEW_TOKEN frame Section 19.7 to provide the client +with an address validation token that can be used to validate future +connections. The client includes this token in Initial packets to provide +address validation in a future connection. The client MUST include the token in +all Initial packets it sends, unless a Retry replaces the token with a newer +one. The client MUST NOT use the token provided in a Retry for future +connections. Servers MAY discard any Initial packet that does not carry the +expected token.¶
+Unlike the token that is created for a Retry packet, there might be some time +between when the token is created and when the token is subsequently used. +Thus, a token SHOULD have an expiration time, which could be either an explicit +expiration time or an issued timestamp that can be used to dynamically calculate +the expiration time. A server can store the expiration time or include it in an +encrypted form in the token.¶
+A token issued with NEW_TOKEN MUST NOT include information that would allow +values to be linked by an on-path observer to the connection on which it was +issued, unless the values are encrypted. For example, it cannot include the +previous connection ID or addressing information. A server MUST ensure that +every NEW_TOKEN frame it sends is unique across all clients, with the exception +of those sent to repair losses of previously sent NEW_TOKEN frames. Information +that allows the server to distinguish between tokens from Retry and NEW_TOKEN +MAY be accessible to entities other than the server.¶
+It is unlikely that the client port number is the same on two different +connections; validating the port is therefore unlikely to be successful.¶
+A token received in a NEW_TOKEN frame is applicable to any server that the +connection is considered authoritative for (e.g., server names included in the +certificate). When connecting to a server for which the client retains an +applicable and unused token, it SHOULD include that token in the Token field of +its Initial packet. Including a token might allow the server to validate the +client address without an additional round trip. A client MUST NOT include a +token that is not applicable to the server that it is connecting to, unless the +client has the knowledge that the server that issued the token and the server +the client is connecting to are jointly managing the tokens.¶
+A token allows a server to correlate activity between the connection where the +token was issued and any connection where it is used. Clients that want to +break continuity of identity with a server MAY discard tokens provided using the +NEW_TOKEN frame. In comparison, a token obtained in a Retry packet MUST be used +immediately during the connection attempt and cannot be used in subsequent +connection attempts.¶
+A client SHOULD NOT reuse a NEW_TOKEN token for different connection attempts. +Reusing a token allows connections to be linked by entities on the network path; +see Section 9.5. A client MUST NOT reuse a token if it believes +that its point of network attachment has changed since the token was last used; +that is, if there is a change in its local IP address or network interface. A +client needs to start the connection process over if there is any change in its +local address prior to completing the handshake.¶
+Clients might receive multiple tokens on a single connection. Aside from +preventing linkability, any token can be used in any connection attempt. +Servers can send additional tokens to either enable address validation for +multiple connection attempts or to replace older tokens that might become +invalid. For a client, this ambiguity means that sending the most recent unused +token is most likely to be effective. Though saving and using older tokens has +no negative consequences, clients can regard older tokens as being less likely +be useful to the server for address validation.¶
+When a server receives an Initial packet with an address validation token, it +MUST attempt to validate the token, unless it has already completed address +validation. If the token is invalid then the server SHOULD proceed as if +the client did not have a validated address, including potentially sending +a Retry. If the validation succeeds, the server SHOULD then allow the +handshake to proceed.¶
+In a stateless design, a server can use encrypted and authenticated tokens to +pass information to clients that the server can later recover and use to +validate a client address. Tokens are not integrated into the cryptographic +handshake and so they are not authenticated. For instance, a client might be +able to reuse a token. To avoid attacks that exploit this property, a server +can limit its use of tokens to only the information needed to validate client +addresses.¶
+Attackers could replay tokens to use servers as amplifiers in DDoS attacks. To +protect against such attacks, servers SHOULD ensure that tokens sent in Retry +packets are only accepted for a short time. Tokens that are provided in +NEW_TOKEN frames (see Section 19.7) need to be valid for longer, but +SHOULD NOT be accepted multiple times in a short period. Servers are encouraged +to allow tokens to be used only once, if possible.¶
+An address validation token MUST be difficult to guess. Including a large +enough random value in the token would be sufficient, but this depends on the +server remembering the value it sends to clients.¶
+A token-based scheme allows the server to offload any state associated with +validation to the client. For this design to work, the token MUST be covered by +integrity protection against modification or falsification by clients. Without +integrity protection, malicious clients could generate or guess values for +tokens that would be accepted by the server. Only the server requires access to +the integrity protection key for tokens.¶
+There is no need for a single well-defined format for the token because the +server that generates the token also consumes it. A token could include +information about the claimed client address (IP and port), a timestamp, and any +other supplementary information the server will need to validate the token in +the future.¶
+Path validation is used during connection migration (see Section 9 and +Section 9.6) by the migrating endpoint to verify reachability of a +peer from a new local address. In path validation, endpoints test reachability +between a specific local address and a specific peer address, where an address +is the two-tuple of IP address and port.¶
+Path validation tests that packets (PATH_CHALLENGE) can be both sent to and +received (PATH_RESPONSE) from a peer on the path. Importantly, it validates +that the packets received from the migrating endpoint do not carry a spoofed +source address.¶
+Path validation can be used at any time by either endpoint. For instance, an +endpoint might check that a peer is still in possession of its address after a +period of quiescence.¶
+Path validation is not designed as a NAT traversal mechanism. Though the +mechanism described here might be effective for the creation of NAT bindings +that support NAT traversal, the expectation is that one or other peer is able to +receive packets without first having sent a packet on that path. Effective NAT +traversal needs additional synchronization mechanisms that are not provided +here.¶
+An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that are used for +path validation with other frames. In particular, an endpoint may pad a packet +carrying a PATH_CHALLENGE for PMTU discovery, or an endpoint may bundle a +PATH_RESPONSE with its own PATH_CHALLENGE.¶
+When probing a new path, an endpoint might want to ensure that its peer has an +unused connection ID available for responses. The endpoint can send +NEW_CONNECTION_ID and PATH_CHALLENGE frames in the same packet. This ensures +that an unused connection ID will be available to the peer when sending a +response.¶
+To initiate path validation, an endpoint sends a PATH_CHALLENGE frame containing +a random payload on the path to be validated.¶
+An endpoint MAY send multiple PATH_CHALLENGE frames to guard against packet +loss. However, an endpoint SHOULD NOT send multiple PATH_CHALLENGE frames in a +single packet. An endpoint SHOULD NOT send a PATH_CHALLENGE more frequently +than it would an Initial packet, ensuring that connection migration is no more +load on a new path than establishing a new connection.¶
+The endpoint MUST use unpredictable data in every PATH_CHALLENGE frame so that +it can associate the peer's response with the corresponding PATH_CHALLENGE.¶
+On receiving a PATH_CHALLENGE frame, an endpoint MUST respond immediately by +echoing the data contained in the PATH_CHALLENGE frame in a PATH_RESPONSE frame.¶
+An endpoint MUST NOT send more than one PATH_RESPONSE frame in response to one +PATH_CHALLENGE frame (see Section 13.3). The peer is +expected to send more PATH_CHALLENGE frames as necessary to evoke additional +PATH_RESPONSE frames.¶
+A new address is considered valid when a PATH_RESPONSE frame is received that +contains the data that was sent in a previous PATH_CHALLENGE. Receipt of an +acknowledgment for a packet containing a PATH_CHALLENGE frame is not adequate +validation, since the acknowledgment can be spoofed by a malicious peer.¶
+Note that receipt on a different local address does not result in path +validation failure, as it might be a result of a forwarded packet (see +Section 9.3.3) or misrouting. It is possible that a valid PATH_RESPONSE +might be received in the future.¶
+Path validation only fails when the endpoint attempting to validate the path +abandons its attempt to validate the path.¶
+Endpoints SHOULD abandon path validation based on a timer. When setting this +timer, implementations are cautioned that the new path could have a longer +round-trip time than the original. A value of three times the larger of the +current Probe Timeout (PTO) or the initial timeout (that is, 2*kInitialRtt) as +defined in [QUIC-RECOVERY] is RECOMMENDED. That is:¶
+
validation_timeout = max(3*PTO, 6*kInitialRtt)
-
-Note that the endpoint might receive packets containing other frames on the new path, but a PATH_RESPONSE frame with appropriate data is required for path validation to succeed.
-When an endpoint abandons path validation, it determines that the path is unusable. This does not necessarily imply a failure of the connection - endpoints can continue sending packets over other paths as appropriate. If no paths are available, an endpoint can wait for a new path to become available or close the connection.
-A path validation might be abandoned for other reasons besides failure. Primarily, this happens if a connection migration to a new path is initiated while a path validation on the old path is in progress.
-The use of a connection ID allows connections to survive changes to endpoint addresses (IP address and port), such as those caused by an endpoint migrating to a new network. This section describes the process by which an endpoint migrates to a new address.
-The design of QUIC relies on endpoints retaining a stable address for the duration of the handshake. An endpoint MUST NOT initiate connection migration before the handshake is confirmed, as defined in section 4.1.2 of [QUIC-TLS].
-An endpoint also MUST NOT send packets from a different local address, actively initiating migration, if the peer sent the disable_active_migration transport parameter during the handshake. An endpoint which has sent this transport parameter, but detects that a peer has nonetheless migrated to a different network MUST either drop the incoming packets on that path without generating a stateless reset or proceed with path validation and allow the peer to migrate. Generating a stateless reset or closing the connection would allow third parties in the network to cause connections to close by spoofing or otherwise manipulating observed traffic.
-Not all changes of peer address are intentional, or active, migrations. The peer could experience NAT rebinding: a change of address due to a middlebox, usually a NAT, allocating a new outgoing port or even a new outgoing IP address for a flow. An endpoint MUST perform path validation (Section 8.2) if it detects any change to a peer’s address, unless it has previously validated that address.
-When an endpoint has no validated path on which to send packets, it MAY discard connection state. An endpoint capable of connection migration MAY wait for a new path to become available before discarding connection state.
-This document limits migration of connections to new client addresses, except as described in Section 9.6. Clients are responsible for initiating all migrations. Servers do not send non-probing packets (see Section 9.1) toward a client address until they see a non-probing packet from that address. If a client receives packets from an unknown server address, the client MUST discard these packets.
-An endpoint MAY probe for peer reachability from a new local address using path validation Section 8.2 prior to migrating the connection to the new local address. Failure of path validation simply means that the new path is not usable for this connection. Failure to validate a path does not cause the connection to end unless there are no valid alternative paths available.
-An endpoint uses a new connection ID for probes sent from a new local address, see Section 9.5 for further discussion. An endpoint that uses a new local address needs to ensure that at least one new connection ID is available at the peer. That can be achieved by including a NEW_CONNECTION_ID frame in the probe.
-Receiving a PATH_CHALLENGE frame from a peer indicates that the peer is probing for reachability on a path. An endpoint sends a PATH_RESPONSE in response as per Section 8.2.
-PATH_CHALLENGE, PATH_RESPONSE, NEW_CONNECTION_ID, and PADDING frames are “probing frames”, and all other frames are “non-probing frames”. A packet containing only probing frames is a “probing packet”, and a packet containing any other frame is a “non-probing packet”.
-An endpoint can migrate a connection to a new local address by sending packets containing non-probing frames from that address.
-Each endpoint validates its peer’s address during connection establishment. Therefore, a migrating endpoint can send to its peer knowing that the peer is willing to receive at the peer’s current address. Thus an endpoint can migrate to a new local address without first validating the peer’s address.
-When migrating, the new path might not support the endpoint’s current sending rate. Therefore, the endpoint resets its congestion controller, as described in Section 9.4.
-The new path might not have the same ECN capability. Therefore, the endpoint verifies ECN capability as described in Section 13.4.
-Receiving acknowledgments for data sent on the new path serves as proof of the peer’s reachability from the new address. Note that since acknowledgments may be received on any path, return reachability on the new path is not established. To establish return reachability on the new path, an endpoint MAY concurrently initiate path validation Section 8.2 on the new path.
-Receiving a packet from a new peer address containing a non-probing frame indicates that the peer has migrated to that address.
-In response to such a packet, an endpoint MUST start sending subsequent packets to the new peer address and MUST initiate path validation (Section 8.2) to verify the peer’s ownership of the unvalidated address.
-An endpoint MAY send data to an unvalidated peer address, but it MUST protect against potential attacks as described in Section 9.3.1 and Section 9.3.2. An endpoint MAY skip validation of a peer address if that address has been seen recently.
-An endpoint only changes the address that it sends packets to in response to the highest-numbered non-probing packet. This ensures that an endpoint does not send packets to an old peer address in the case that it receives reordered packets.
-After changing the address to which it sends non-probing packets, an endpoint could abandon any path validation for other addresses.
-Receiving a packet from a new peer address might be the result of a NAT rebinding at the peer.
-After verifying a new client address, the server SHOULD send new address validation tokens (Section 8) to the client.
-It is possible that a peer is spoofing its source address to cause an endpoint to send excessive amounts of data to an unwilling host. If the endpoint sends significantly more data than the spoofing peer, connection migration might be used to amplify the volume of data that an attacker can generate toward a victim.
-As described in Section 9.3, an endpoint is required to validate a peer’s new address to confirm the peer’s possession of the new address. Until a peer’s address is deemed valid, an endpoint MUST limit the rate at which it sends data to this address. The endpoint MUST NOT send more than a minimum congestion window’s worth of data per estimated round-trip time (kMinimumWindow, as defined in [QUIC-RECOVERY]). In the absence of this limit, an endpoint risks being used for a denial of service attack against an unsuspecting victim. Note that since the endpoint will not have any round-trip time measurements to this address, the estimate SHOULD be the default initial value (see [QUIC-RECOVERY]).
-If an endpoint skips validation of a peer address as described in Section 9.3, it does not need to limit its sending rate.
-An on-path attacker could cause a spurious connection migration by copying and forwarding a packet with a spoofed address such that it arrives before the original packet. The packet with the spoofed address will be seen to come from a migrating connection, and the original packet will be seen as a duplicate and dropped. After a spurious migration, validation of the source address will fail because the entity at the source address does not have the necessary cryptographic keys to read or respond to the PATH_CHALLENGE frame that is sent to it even if it wanted to.
-To protect the connection from failing due to such a spurious migration, an endpoint MUST revert to using the last validated peer address when validation of a new peer address fails.
-If an endpoint has no state about the last validated peer address, it MUST close the connection silently by discarding all connection state. This results in new packets on the connection being handled generically. For instance, an endpoint MAY send a stateless reset in response to any further incoming packets.
-Note that receipt of packets with higher packet numbers from the legitimate peer address will trigger another connection migration. This will cause the validation of the address of the spurious migration to be abandoned.
-An off-path attacker that can observe packets might forward copies of genuine packets to endpoints. If the copied packet arrives before the genuine packet, this will appear as a NAT rebinding. Any genuine packet will be discarded as a duplicate. If the attacker is able to continue forwarding packets, it might be able to cause migration to a path via the attacker. This places the attacker on path, giving it the ability to observe or drop all subsequent packets.
-Unlike the attack described in Section 9.3.2, the attacker can ensure that the new path is successfully validated.
-This style of attack relies on the attacker using a path that is approximately as fast as the direct path between endpoints. The attack is more reliable if relatively few packets are sent or if packet loss coincides with the attempted attack.
-A non-probing packet received on the original path that increases the maximum received packet number will cause the endpoint to move back to that path. Eliciting packets on this path increases the likelihood that the attack is unsuccessful. Therefore, mitigation of this attack relies on triggering the exchange of packets.
-In response to an apparent migration, endpoints MUST validate the previously active path using a PATH_CHALLENGE frame. This induces the sending of new packets on that path. If the path is no longer viable, the validation attempt will time out and fail; if the path is viable, but no longer desired, the validation will succeed, but only results in probing packets being sent on the path.
-An endpoint that receives a PATH_CHALLENGE on an active path SHOULD send a non-probing packet in response. If the non-probing packet arrives before any copy made by an attacker, this results in the connection being migrated back to the original path. Any subsequent migration to another path restarts this entire process.
-This defense is imperfect, but this is not considered a serious problem. If the path via the attack is reliably faster than the original path despite multiple attempts to use that original path, it is not possible to distinguish between attack and an improvement in routing.
-An endpoint could also use heuristics to improve detection of this style of attack. For instance, NAT rebinding is improbable if packets were recently received on the old path, similarly rebinding is rare on IPv6 paths. Endpoints can also look for duplicated packets. Conversely, a change in connection ID is more likely to indicate an intentional migration rather than an attack.
-The capacity available on the new path might not be the same as the old path. Packets sent on the old path SHOULD NOT contribute to congestion control or RTT estimation for the new path.
-On confirming a peer’s ownership of its new address, an endpoint MUST immediately reset the congestion controller and round-trip time estimator for the new path to initial values (see Sections A.3 and B.3 in [QUIC-RECOVERY]) unless it has knowledge that a previous send rate or round-trip time estimate is valid for the new path. For instance, an endpoint might infer that a change in only the client’s port number is indicative of a NAT rebinding, meaning that the new path is likely to have similar bandwidth and round-trip time. However, this determination will be imperfect. If the determination is incorrect, the congestion controller and the RTT estimator are expected to adapt to the new path. Generally, implementations are advised to be cautious when using previous values on a new path.
-There may be apparent reordering at the receiver when an endpoint sends data and probes from/to multiple addresses during the migration period, since the two resulting paths may have different round-trip times. A receiver of packets on multiple paths will still send ACK frames covering all received packets.
-While multiple paths might be used during connection migration, a single congestion control context and a single loss recovery context (as described in [QUIC-RECOVERY]) may be adequate. For instance, an endpoint might delay switching to a new congestion control context until it is confirmed that an old path is no longer needed (such as the case in Section 9.3.3).
-A sender can make exceptions for probe packets so that their loss detection is independent and does not unduly cause the congestion controller to reduce its sending rate. An endpoint might set a separate timer when a PATH_CHALLENGE is sent, which is cancelled when the corresponding PATH_RESPONSE is received. If the timer fires before the PATH_RESPONSE is received, the endpoint might send a new PATH_CHALLENGE, and restart the timer for a longer period of time.
-Using a stable connection ID on multiple network paths allows a passive observer to correlate activity between those paths. An endpoint that moves between networks might not wish to have their activity correlated by any entity other than their peer, so different connection IDs are used when sending from different local addresses, as discussed in Section 5.1. For this to be effective endpoints need to ensure that connection IDs they provide cannot be linked by any other entity.
-At any time, endpoints MAY change the Destination Connection ID they send to a value that has not been used on another path.
-An endpoint MUST use a new connection ID if it initiates connection migration as described in Section 9.2 or probes a new network path as described in Section 9.1. An endpoint MUST use a new connection ID in response to a change in the address of a peer if the packet with the new peer address uses an active connection ID that has not been previously used by the peer.
-Using different connection IDs for packets sent in both directions on each new network path eliminates the use of the connection ID for linking packets from the same connection across different network paths. Header protection ensures that packet numbers cannot be used to correlate activity. This does not prevent other properties of packets, such as timing and size, from being used to correlate activity.
-Unintentional changes in path without a change in connection ID are possible. For example, after a period of network inactivity, NAT rebinding might cause packets to be sent on a new path when the client resumes sending.
-A client might wish to reduce linkability by employing a new connection ID and source UDP port when sending traffic after a period of inactivity. Changing the UDP port from which it sends packets at the same time might cause the packet to appear as a connection migration. This ensures that the mechanisms that support migration are exercised even for clients that don’t experience NAT rebindings or genuine migrations. Changing port number can cause a peer to reset its congestion state (see Section 9.4), so the port SHOULD only be changed infrequently.
-An endpoint that exhausts available connection IDs cannot probe new paths or initiate migration, nor can it respond to probes or attempts by its peer to migrate. To ensure that migration is possible and packets sent on different paths cannot be correlated, endpoints SHOULD provide new connection IDs before peers migrate; see Section 5.1.1. If a peer might have exhausted available connection IDs, a migrating endpoint could include a NEW_CONNECTION_ID frame in all packets sent on a new network path.
-QUIC allows servers to accept connections on one IP address and attempt to transfer these connections to a more preferred address shortly after the handshake. This is particularly useful when clients initially connect to an address shared by multiple servers but would prefer to use a unicast address to ensure connection stability. This section describes the protocol for migrating a connection to a preferred server address.
-Migrating a connection to a new server address mid-connection is left for future work. If a client receives packets from a new server address not indicated by the preferred_address transport parameter, the client SHOULD discard these packets.
-A server conveys a preferred address by including the preferred_address transport parameter in the TLS handshake.
-Servers MAY communicate a preferred address of each address family (IPv4 and IPv6) to allow clients to pick the one most suited to their network attachment.
-Once the handshake is finished, the client SHOULD select one of the two server’s preferred addresses and initiate path validation (see Section 8.2) of that address using the connection ID provided in the preferred_address transport parameter.
-If path validation succeeds, the client SHOULD immediately begin sending all future packets to the new server address using the new connection ID and discontinue use of the old server address. If path validation fails, the client MUST continue sending all future packets to the server’s original IP address.
-A server might receive a packet addressed to its preferred IP address at any time after it accepts a connection. If this packet contains a PATH_CHALLENGE frame, the server sends a PATH_RESPONSE frame as per Section 8.2. The server MUST send other non-probing frames from its original address until it receives a non-probing packet from the client at its preferred address and until the server has validated the new path.
-The server MUST probe on the path toward the client from its preferred address. This helps to guard against spurious migration initiated by an attacker.
-Once the server has completed its path validation and has received a non-probing packet with a new largest packet number on its preferred address, the server begins sending non-probing packets to the client exclusively from its preferred IP address. It SHOULD drop packets for this connection received on the old IP address, but MAY continue to process delayed packets.
-A client might need to perform a connection migration before it has migrated to the server’s preferred address. In this case, the client SHOULD perform path validation to both the original and preferred server address from the client’s new address concurrently.
-If path validation of the server’s preferred address succeeds, the client MUST abandon validation of the original address and migrate to using the server’s preferred address. If path validation of the server’s preferred address fails but validation of the server’s original address succeeds, the client MAY migrate to its new address and continue sending to the server’s original address.
-If the connection to the server’s preferred address is not from the same client address, the server MUST protect against potential attacks as described in Section 9.3.1 and Section 9.3.2. In addition to intentional simultaneous migration, this might also occur because the client’s access network used a different NAT binding for the server’s preferred address.
-Servers SHOULD initiate path validation to the client’s new address upon receiving a probe packet from a different address. Servers MUST NOT send more than a minimum congestion window’s worth of non-probing packets to the new address before path validation is complete.
-A client that migrates to a new address SHOULD use a preferred address from the same address family for the server.
-Endpoints that send data using IPv6 SHOULD apply an IPv6 flow label in compliance with [RFC6437], unless the local API does not allow setting IPv6 flow labels.
-The IPv6 flow label SHOULD be a pseudo-random function of the source and destination addresses, source and destination UDP ports, and the destination CID. The flow label generation MUST be designed to minimize the chances of linkability with a previously used flow label, as this would enable correlating activity on multiple paths (see Section 9.5).
-A possible implementation is to compute the flow label as a cryptographic hash function of the source and destination addresses, source and destination UDP ports, destination CID, and a local secret.
-An established QUIC connection can be terminated in one of three ways:
- - +¶ +Note that the endpoint might receive packets containing other frames on the new +path, but a PATH_RESPONSE frame with appropriate data is required for path +validation to succeed.¶
+When an endpoint abandons path validation, it determines that the path is +unusable. This does not necessarily imply a failure of the connection - +endpoints can continue sending packets over other paths as appropriate. If no +paths are available, an endpoint can wait for a new path to become available or +close the connection.¶
+A path validation might be abandoned for other reasons besides +failure. Primarily, this happens if a connection migration to a new path is +initiated while a path validation on the old path is in progress.¶
+The use of a connection ID allows connections to survive changes to endpoint +addresses (IP address and port), such as those caused by an +endpoint migrating to a new network. This section describes the process by +which an endpoint migrates to a new address.¶
+The design of QUIC relies on endpoints retaining a stable address for the +duration of the handshake. An endpoint MUST NOT initiate connection migration +before the handshake is confirmed, as defined in section 4.1.2 of [QUIC-TLS].¶
+An endpoint also MUST NOT send packets from a different local address, actively
+initiating migration, if the peer sent the disable_active_migration transport
+parameter during the handshake. An endpoint which has sent this transport
+parameter, but detects that a peer has nonetheless migrated to a different
+network MUST either drop the incoming packets on that path without generating a
+stateless reset or proceed with path validation and allow the peer to migrate.
+Generating a stateless reset or closing the connection would allow third parties
+in the network to cause connections to close by spoofing or otherwise
+manipulating observed traffic.¶
Not all changes of peer address are intentional, or active, migrations. The peer +could experience NAT rebinding: a change of address due to a middlebox, usually +a NAT, allocating a new outgoing port or even a new outgoing IP address for a +flow. An endpoint MUST perform path validation (Section 8.2) if it +detects any change to a peer's address, unless it has previously validated that +address.¶
+When an endpoint has no validated path on which to send packets, it MAY discard +connection state. An endpoint capable of connection migration MAY wait for a +new path to become available before discarding connection state.¶
+This document limits migration of connections to new client addresses, except as +described in Section 9.6. Clients are responsible for initiating all +migrations. Servers do not send non-probing packets (see Section 9.1) toward a +client address until they see a non-probing packet from that address. If a +client receives packets from an unknown server address, the client MUST discard +these packets.¶
+An endpoint MAY probe for peer reachability from a new local address using path +validation Section 8.2 prior to migrating the connection to the new +local address. Failure of path validation simply means that the new path is not +usable for this connection. Failure to validate a path does not cause the +connection to end unless there are no valid alternative paths available.¶
+An endpoint uses a new connection ID for probes sent from a new local address; +see Section 9.5 for further discussion. An endpoint that uses +a new local address needs to ensure that at least one new connection ID is +available at the peer. That can be achieved by including a NEW_CONNECTION_ID +frame in the probe.¶
+Receiving a PATH_CHALLENGE frame from a peer indicates that the peer is probing +for reachability on a path. An endpoint sends a PATH_RESPONSE in response as per +Section 8.2.¶
+PATH_CHALLENGE, PATH_RESPONSE, NEW_CONNECTION_ID, and PADDING frames are +"probing frames", and all other frames are "non-probing frames". A packet +containing only probing frames is a "probing packet", and a packet containing +any other frame is a "non-probing packet".¶
+An endpoint can migrate a connection to a new local address by sending packets +containing non-probing frames from that address.¶
+Each endpoint validates its peer's address during connection establishment. +Therefore, a migrating endpoint can send to its peer knowing that the peer is +willing to receive at the peer's current address. Thus an endpoint can migrate +to a new local address without first validating the peer's address.¶
+When migrating, the new path might not support the endpoint's current sending +rate. Therefore, the endpoint resets its congestion controller, as described in +Section 9.4.¶
+The new path might not have the same ECN capability. Therefore, the endpoint +verifies ECN capability as described in Section 13.4.¶
+Receiving acknowledgments for data sent on the new path serves as proof of the +peer's reachability from the new address. Note that since acknowledgments may +be received on any path, return reachability on the new path is not +established. To establish return reachability on the new path, an endpoint MAY +concurrently initiate path validation Section 8.2 on the new path.¶
+Receiving a packet from a new peer address containing a non-probing frame +indicates that the peer has migrated to that address.¶
+In response to such a packet, an endpoint MUST start sending subsequent packets +to the new peer address and MUST initiate path validation (Section 8.2) +to verify the peer's ownership of the unvalidated address.¶
+An endpoint MAY send data to an unvalidated peer address, but it MUST protect +against potential attacks as described in Section 9.3.1 and +Section 9.3.2. An endpoint MAY skip validation of a peer address if that +address has been seen recently. In particular, if an endpoint returns to a +previously-validated path after detecting some form of spurious migration, +skipping address validation and restoring loss detection and congestion state +can reduce the performance impact of the attack.¶
+An endpoint only changes the address that it sends packets to in response to the +highest-numbered non-probing packet. This ensures that an endpoint does not send +packets to an old peer address in the case that it receives reordered packets.¶
+After changing the address to which it sends non-probing packets, an endpoint +could abandon any path validation for other addresses.¶
+Receiving a packet from a new peer address might be the result of a NAT +rebinding at the peer.¶
+After verifying a new client address, the server SHOULD send new address +validation tokens (Section 8) to the client.¶
+It is possible that a peer is spoofing its source address to cause an endpoint +to send excessive amounts of data to an unwilling host. If the endpoint sends +significantly more data than the spoofing peer, connection migration might be +used to amplify the volume of data that an attacker can generate toward a +victim.¶
+As described in Section 9.3, an endpoint is required to validate a +peer's new address to confirm the peer's possession of the new address. Until a +peer's address is deemed valid, an endpoint MUST limit the rate at which it +sends data to this address. The endpoint MUST NOT send more than a minimum +congestion window's worth of data per estimated round-trip time (kMinimumWindow, +as defined in [QUIC-RECOVERY]). In the absence of this limit, an endpoint +risks being used for a denial of service attack against an unsuspecting victim. +Note that since the endpoint will not have any round-trip time measurements to +this address, the estimate SHOULD be the default initial value (see +[QUIC-RECOVERY]).¶
+If an endpoint skips validation of a peer address as described in +Section 9.3, it does not need to limit its sending rate.¶
+An on-path attacker could cause a spurious connection migration by copying and +forwarding a packet with a spoofed address such that it arrives before the +original packet. The packet with the spoofed address will be seen to come from +a migrating connection, and the original packet will be seen as a duplicate and +dropped. After a spurious migration, validation of the source address will fail +because the entity at the source address does not have the necessary +cryptographic keys to read or respond to the PATH_CHALLENGE frame that is sent +to it even if it wanted to.¶
+To protect the connection from failing due to such a spurious migration, an +endpoint MUST revert to using the last validated peer address when validation of +a new peer address fails.¶
+If an endpoint has no state about the last validated peer address, it MUST close +the connection silently by discarding all connection state. This results in new +packets on the connection being handled generically. For instance, an endpoint +MAY send a stateless reset in response to any further incoming packets.¶
+Note that receipt of packets with higher packet numbers from the legitimate peer +address will trigger another connection migration. This will cause the +validation of the address of the spurious migration to be abandoned.¶
+An off-path attacker that can observe packets might forward copies of genuine +packets to endpoints. If the copied packet arrives before the genuine packet, +this will appear as a NAT rebinding. Any genuine packet will be discarded as a +duplicate. If the attacker is able to continue forwarding packets, it might be +able to cause migration to a path via the attacker. This places the attacker on +path, giving it the ability to observe or drop all subsequent packets.¶
+Unlike the attack described in Section 9.3.2, the attacker can ensure +that the new path is successfully validated.¶
+This style of attack relies on the attacker using a path that is approximately +as fast as the direct path between endpoints. The attack is more reliable if +relatively few packets are sent or if packet loss coincides with the attempted +attack.¶
+A non-probing packet received on the original path that increases the maximum +received packet number will cause the endpoint to move back to that path. +Eliciting packets on this path increases the likelihood that the attack is +unsuccessful. Therefore, mitigation of this attack relies on triggering the +exchange of packets.¶
+In response to an apparent migration, endpoints MUST validate the previously +active path using a PATH_CHALLENGE frame. This induces the sending of new +packets on that path. If the path is no longer viable, the validation attempt +will time out and fail; if the path is viable, but no longer desired, the +validation will succeed, but only results in probing packets being sent on the +path.¶
+An endpoint that receives a PATH_CHALLENGE on an active path SHOULD send a +non-probing packet in response. If the non-probing packet arrives before any +copy made by an attacker, this results in the connection being migrated back to +the original path. Any subsequent migration to another path restarts this +entire process.¶
+This defense is imperfect, but this is not considered a serious problem. If the +path via the attack is reliably faster than the original path despite multiple +attempts to use that original path, it is not possible to distinguish between +attack and an improvement in routing.¶
+An endpoint could also use heuristics to improve detection of this style of +attack. For instance, NAT rebinding is improbable if packets were recently +received on the old path, similarly rebinding is rare on IPv6 paths. Endpoints +can also look for duplicated packets. Conversely, a change in connection ID is +more likely to indicate an intentional migration rather than an attack.¶
+The capacity available on the new path might not be the same as the old path. +Packets sent on the old path MUST NOT contribute to congestion control or RTT +estimation for the new path.¶
+On confirming a peer's ownership of its new address, an endpoint MUST +immediately reset the congestion controller and round-trip time estimator for +the new path to initial values (see Sections A.3 and B.3 in [QUIC-RECOVERY]) +unless it has knowledge that a previous send rate or round-trip time estimate is +valid for the new path. For instance, an endpoint might infer that a change in +only the client's port number is indicative of a NAT rebinding, meaning that the +new path is likely to have similar bandwidth and round-trip time. However, this +determination will be imperfect. If the determination is incorrect, the +congestion controller and the RTT estimator are expected to adapt to the new +path. Generally, implementations are advised to be cautious when using previous +values on a new path.¶
+There may be apparent reordering at the receiver when an endpoint sends data and +probes from/to multiple addresses during the migration period, since the two +resulting paths may have different round-trip times. A receiver of packets on +multiple paths will still send ACK frames covering all received packets.¶
+While multiple paths might be used during connection migration, a single +congestion control context and a single loss recovery context (as described in +[QUIC-RECOVERY]) may be adequate. For instance, an endpoint might delay +switching to a new congestion control context until it is confirmed that an old +path is no longer needed (such as the case in Section 9.3.3).¶
+A sender can make exceptions for probe packets so that their loss detection is +independent and does not unduly cause the congestion controller to reduce its +sending rate. An endpoint might set a separate timer when a PATH_CHALLENGE is +sent, which is cancelled when the corresponding PATH_RESPONSE is received. If +the timer fires before the PATH_RESPONSE is received, the endpoint might send a +new PATH_CHALLENGE, and restart the timer for a longer period of time.¶
+Using a stable connection ID on multiple network paths allows a passive observer +to correlate activity between those paths. An endpoint that moves between +networks might not wish to have their activity correlated by any entity other +than their peer, so different connection IDs are used when sending from +different local addresses, as discussed in Section 5.1. For this to be +effective endpoints need to ensure that connection IDs they provide cannot be +linked by any other entity.¶
+At any time, endpoints MAY change the Destination Connection ID they send to a +value that has not been used on another path.¶
+An endpoint MUST use a new connection ID if it initiates connection migration as +described in Section 9.2 or probes a new network path as described +in Section 9.1. An endpoint MUST use a new connection ID in response to a +change in the address of a peer if the packet with the new peer address uses an +active connection ID that has not been previously used by the peer.¶
+Using different connection IDs for packets sent in both directions on each new +network path eliminates the use of the connection ID for linking packets from +the same connection across different network paths. Header protection ensures +that packet numbers cannot be used to correlate activity. This does not prevent +other properties of packets, such as timing and size, from being used to +correlate activity.¶
+Unintentional changes in path without a change in connection ID are possible. +For example, after a period of network inactivity, NAT rebinding might cause +packets to be sent on a new path when the client resumes sending.¶
+A client might wish to reduce linkability by employing a new connection ID and +source UDP port when sending traffic after a period of inactivity. Changing the +UDP port from which it sends packets at the same time might cause the packet to +appear as a connection migration. This ensures that the mechanisms that support +migration are exercised even for clients that don't experience NAT rebindings or +genuine migrations. Changing port number can cause a peer to reset its +congestion state (see Section 9.4), so the port SHOULD only be changed +infrequently.¶
+An endpoint that exhausts available connection IDs cannot probe new paths or +initiate migration, nor can it respond to probes or attempts by its peer to +migrate. To ensure that migration is possible and packets sent on different +paths cannot be correlated, endpoints SHOULD provide new connection IDs before +peers migrate; see Section 5.1.1. If a peer might have exhausted available +connection IDs, a migrating endpoint could include a NEW_CONNECTION_ID frame in +all packets sent on a new network path.¶
+QUIC allows servers to accept connections on one IP address and attempt to +transfer these connections to a more preferred address shortly after the +handshake. This is particularly useful when clients initially connect to an +address shared by multiple servers but would prefer to use a unicast address to +ensure connection stability. This section describes the protocol for migrating a +connection to a preferred server address.¶
+Migrating a connection to a new server address mid-connection is left for future +work. If a client receives packets from a new server address not indicated by +the preferred_address transport parameter, the client SHOULD discard these +packets.¶
+A server conveys a preferred address by including the preferred_address +transport parameter in the TLS handshake.¶
+Servers MAY communicate a preferred address of each address family (IPv4 and +IPv6) to allow clients to pick the one most suited to their network attachment.¶
+Once the handshake is finished, the client SHOULD select one of the two +server's preferred addresses and initiate path validation (see +Section 8.2) of that address using the connection ID provided in the +preferred_address transport parameter.¶
+If path validation succeeds, the client SHOULD immediately begin sending all +future packets to the new server address using the new connection ID and +discontinue use of the old server address. If path validation fails, the client +MUST continue sending all future packets to the server's original IP address.¶
+A server might receive a packet addressed to its preferred IP address at any +time after it accepts a connection. If this packet contains a PATH_CHALLENGE +frame, the server sends a PATH_RESPONSE frame as per Section 8.2. The +server MUST send other non-probing frames from its original address until it +receives a non-probing packet from the client at its preferred address and until +the server has validated the new path.¶
+The server MUST probe on the path toward the client from its preferred address. +This helps to guard against spurious migration initiated by an attacker.¶
+Once the server has completed its path validation and has received a non-probing +packet with a new largest packet number on its preferred address, the server +begins sending non-probing packets to the client exclusively from its preferred +IP address. It SHOULD drop packets for this connection received on the old IP +address, but MAY continue to process delayed packets.¶
+A client might need to perform a connection migration before it has migrated to +the server's preferred address. In this case, the client SHOULD perform path +validation to both the original and preferred server address from the client's +new address concurrently.¶
+If path validation of the server's preferred address succeeds, the client MUST +abandon validation of the original address and migrate to using the server's +preferred address. If path validation of the server's preferred address fails +but validation of the server's original address succeeds, the client MAY migrate +to its new address and continue sending to the server's original address.¶
+If the connection to the server's preferred address is not from the same client +address, the server MUST protect against potential attacks as described in +Section 9.3.1 and Section 9.3.2. In addition to intentional +simultaneous migration, this might also occur because the client's access +network used a different NAT binding for the server's preferred address.¶
+Servers SHOULD initiate path validation to the client's new address upon +receiving a probe packet from a different address. Servers MUST NOT send more +than a minimum congestion window's worth of non-probing packets to the new +address before path validation is complete.¶
+A client that migrates to a new address SHOULD use a preferred address from the +same address family for the server.¶
+Endpoints that send data using IPv6 SHOULD apply an IPv6 flow label +in compliance with [RFC6437], unless the local API does not allow +setting IPv6 flow labels.¶
+The IPv6 flow label SHOULD be a pseudo-random function of the source +and destination addresses, source and destination UDP ports, and the destination +CID. The flow label generation MUST be designed to minimize the chances of +linkability with a previously used flow label, as this would enable correlating +activity on multiple paths (see Section 9.5).¶
+A possible implementation is to compute the flow label as a cryptographic hash +function of the source and destination addresses, source and destination +UDP ports, destination CID, and a local secret.¶
+An established QUIC connection can be terminated in one of three ways:¶
An endpoint MAY discard connection state if it does not have a validated path on which it can send packets (see Section 8.2).
-The closing and draining connection states exist to ensure that connections close cleanly and that delayed or reordered packets are properly discarded. These states SHOULD persist for at least three times the current Probe Timeout (PTO) interval as defined in [QUIC-RECOVERY].
-An endpoint enters a closing period after initiating an immediate close (Section 10.3). While closing, an endpoint MUST NOT send packets unless they contain a CONNECTION_CLOSE frame (see Section 10.3 for details). An endpoint retains only enough information to generate a packet containing a CONNECTION_CLOSE frame and to identify packets as belonging to the connection. The endpoint’s selected connection ID and the QUIC version are sufficient information to identify packets for a closing connection; an endpoint can discard all other connection state. An endpoint MAY retain packet protection keys for incoming packets to allow it to read and process a CONNECTION_CLOSE frame.
-The draining state is entered once an endpoint receives a signal that its peer is closing or draining. While otherwise identical to the closing state, an endpoint in the draining state MUST NOT send any packets. Retaining packet protection keys is unnecessary once a connection is in the draining state.
-An endpoint MAY transition from the closing period to the draining period if it receives a CONNECTION_CLOSE frame or stateless reset, both of which indicate that the peer is also closing or draining. The draining period SHOULD end when the closing period would have ended. In other words, the endpoint can use the same end time, but cease retransmission of the closing packet.
-Disposing of connection state prior to the end of the closing or draining period could cause delayed or reordered packets to generate an unnecessary stateless reset. Endpoints that have some alternative means to ensure that late-arriving packets on the connection do not induce a response, such as those that are able to close the UDP socket, MAY use an abbreviated draining period which can allow for faster resource recovery. Servers that retain an open socket for accepting new connections SHOULD NOT exit the closing or draining period early.
-Once the closing or draining period has ended, an endpoint SHOULD discard all connection state. This results in new packets on the connection being handled generically. For instance, an endpoint MAY send a stateless reset in response to any further incoming packets.
-The draining and closing periods do not apply when a stateless reset (Section 10.4) is sent.
-An endpoint is not expected to handle key updates when it is closing or draining. A key update might prevent the endpoint from moving from the closing state to draining, but it otherwise has no impact.
-While in the closing period, an endpoint could receive packets from a new source address, indicating a connection migration (Section 9). An endpoint in the closing state MUST strictly limit the number of packets it sends to this new address until the address is validated (see Section 8.2). A server in the closing state MAY instead choose to discard packets received from a new source address.
-If the idle timeout is enabled, a connection is silently closed and the state is discarded when it remains idle for longer than both the advertised idle timeout (see Section 18.2) and three times the current Probe Timeout (PTO).
-Each endpoint advertises its own idle timeout to its peer. An endpoint restarts any timer it maintains when a packet from its peer is received and processed successfully. The timer is also restarted when sending an ack-eliciting packet (see [QUIC-RECOVERY]), but only if no other ack-eliciting packets have been sent since last receiving a packet. Restarting when sending packets ensures that connections do not prematurely time out when initiating new activity.
-The value for an idle timeout can be asymmetric. The value advertised by an endpoint is only used to determine whether the connection is live at that endpoint. An endpoint that sends packets near the end of the idle timeout period of a peer risks having those packets discarded if its peer enters the draining state before the packets arrive. If a peer could timeout within a Probe Timeout (PTO; see Section 6.3 of [QUIC-RECOVERY]), it is advisable to test for liveness before sending any data that cannot be retried safely. Note that it is likely that only applications or application protocols will know what information can be retried.
-An endpoint sends a CONNECTION_CLOSE frame (Section 19.19) to terminate the connection immediately. A CONNECTION_CLOSE frame causes all streams to immediately become closed; open streams can be assumed to be implicitly reset.
-After sending a CONNECTION_CLOSE frame, endpoints immediately enter the closing state. During the closing period, an endpoint that sends a CONNECTION_CLOSE frame SHOULD respond to any packet that it receives with another packet containing a CONNECTION_CLOSE frame. To minimize the state that an endpoint maintains for a closing connection, endpoints MAY send the exact same packet. However, endpoints SHOULD limit the number of packets they generate containing a CONNECTION_CLOSE frame. For instance, an endpoint could progressively increase the number of packets that it receives before sending additional packets or increase the time between packets.
- - -An endpoint MAY discard connection state if it does not have a validated path on +which it can send packets (see Section 8.2).¶
+The closing and draining connection states exist to ensure that connections +close cleanly and that delayed or reordered packets are properly discarded. +These states SHOULD persist for at least three times the current Probe Timeout +(PTO) interval as defined in [QUIC-RECOVERY].¶
+An endpoint enters a closing period after initiating an immediate close +(Section 10.3). While closing, an endpoint MUST NOT send packets unless +they contain a CONNECTION_CLOSE frame (see Section 10.3 for details). An +endpoint retains only enough information to generate a packet containing a +CONNECTION_CLOSE frame and to identify packets as belonging to the connection. +The endpoint's selected connection ID and the QUIC version are sufficient +information to identify packets for a closing connection; an endpoint can +discard all other connection state. An endpoint MAY retain packet protection +keys for incoming packets to allow it to read and process a CONNECTION_CLOSE +frame.¶
+The draining state is entered once an endpoint receives a signal that its peer +is closing or draining. While otherwise identical to the closing state, an +endpoint in the draining state MUST NOT send any packets. Retaining packet +protection keys is unnecessary once a connection is in the draining state.¶
+An endpoint MAY transition from the closing period to the draining period if it +receives a CONNECTION_CLOSE frame or stateless reset, both of which indicate +that the peer is also closing or draining. The draining period SHOULD end when +the closing period would have ended. In other words, the endpoint can use the +same end time, but cease retransmission of the closing packet.¶
+Disposing of connection state prior to the end of the closing or draining period +could cause delayed or reordered packets to generate an unnecessary stateless +reset. Endpoints that have some alternative means to ensure that late-arriving +packets on the connection do not induce a response, such as those that are able +to close the UDP socket, MAY use an abbreviated draining period which can allow +for faster resource recovery. Servers that retain an open socket for accepting +new connections SHOULD NOT exit the closing or draining period early.¶
+Once the closing or draining period has ended, an endpoint SHOULD discard all +connection state. This results in new packets on the connection being handled +generically. For instance, an endpoint MAY send a stateless reset in response +to any further incoming packets.¶
+The draining and closing periods do not apply when a stateless reset +(Section 10.4) is sent.¶
+An endpoint is not expected to handle key updates when it is closing or +draining. A key update might prevent the endpoint from moving from the closing +state to draining, but it otherwise has no impact.¶
+While in the closing period, an endpoint could receive packets from a new source +address, indicating a connection migration (Section 9). An endpoint in the +closing state MUST strictly limit the number of packets it sends to this new +address until the address is validated (see Section 8.2). A server in +the closing state MAY instead choose to discard packets received from a new +source address.¶
+If the idle timeout is enabled, a connection is silently closed and the state is +discarded when it remains idle for longer than both the advertised +idle timeout (see Section 18.2) and three times the +current Probe Timeout (PTO).¶
+Each endpoint advertises its own idle timeout to its peer. An endpoint +restarts any timer it maintains when a packet from its peer is received and +processed successfully. The timer is also restarted when sending an +ack-eliciting packet (see [QUIC-RECOVERY]), but only if no other ack-eliciting +packets have been sent since last receiving a packet. Restarting when sending +packets ensures that connections do not prematurely time out when initiating new +activity.¶
+The value for an idle timeout can be asymmetric. The value advertised by an +endpoint is only used to determine whether the connection is live at that +endpoint. An endpoint that sends packets near the end of the idle timeout +period of a peer risks having those packets discarded if its peer enters the +draining state before the packets arrive. If a peer could timeout within a +Probe Timeout (PTO; see Section 6.3 of [QUIC-RECOVERY]), it is advisable to +test for liveness before sending any data that cannot be retried safely. Note +that it is likely that only applications or application protocols will +know what information can be retried.¶
+An endpoint sends a CONNECTION_CLOSE frame (Section 19.19) to +terminate the connection immediately. A CONNECTION_CLOSE frame causes all +streams to immediately become closed; open streams can be assumed to be +implicitly reset.¶
+After sending a CONNECTION_CLOSE frame, an endpoint immediately enters the +closing state.¶
+During the closing period, an endpoint that sends a CONNECTION_CLOSE frame +SHOULD respond to any incoming packet that can be decrypted with another packet +containing a CONNECTION_CLOSE frame. Such an endpoint SHOULD limit the number +of packets it generates containing a CONNECTION_CLOSE frame. For instance, an +endpoint could wait for a progressively increasing number of received packets or +amount of time before responding to a received packet.¶
+An endpoint is allowed to drop the packet protection keys when entering the +closing period (Section 10.1) and send a packet containing a CONNECTION_CLOSE in +response to any UDP datagram that is received. However, an endpoint without the +packet protection keys cannot identify and discard invalid packets. To avoid +creating an unwitting amplification attack, such endpoints MUST reduce the +frequency with which it sends packets containing a CONNECTION_CLOSE frame. To +minimize the state that an endpoint maintains for a closing connection, +endpoints MAY send the exact same packet.¶
+New packets from unverified addresses could be used to create an amplification attack (see Section 8). To avoid this, endpoints MUST either limit transmission of CONNECTION_CLOSE frames to validated addresses or drop packets without response if the response would be more than three times larger than the received packet.
-After receiving a CONNECTION_CLOSE frame, endpoints enter the draining state. An endpoint that receives a CONNECTION_CLOSE frame MAY send a single packet containing a CONNECTION_CLOSE frame before entering the draining state, using a CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An endpoint MUST NOT send further packets, which could result in a constant exchange of CONNECTION_CLOSE frames until the closing period on either peer ended.
-An immediate close can be used after an application protocol has arranged to close a connection. This might be after the application protocols negotiates a graceful shutdown. The application protocol exchanges whatever messages that are needed to cause both endpoints to agree to close the connection, after which the application requests that the connection be closed. The application protocol can use a CONNECTION_CLOSE frame with an appropriate error code to signal closure.
-When sending CONNECTION_CLOSE, the goal is to ensure that the peer will process the frame. Generally, this means sending the frame in a packet with the highest level of packet protection to avoid the packet being discarded. However, during the handshake, it is possible that more advanced packet protection keys are not available to the peer, so the frame MAY be replicated in a packet that uses a lower packet protection level.
-After the handshake is confirmed, an endpoint MUST send any CONNECTION_CLOSE frames in a 1-RTT packet. Prior to handshake confirmation, the peer might not have 1-RTT keys, so the endpoint SHOULD send CONNECTION_CLOSE frames in a Handshake packet. If the endpoint does not have Handshake keys, it SHOULD send CONNECTION_CLOSE frames in an Initial packet.
-A client will always know whether the server has Handshake keys (see Section 17.2.2.1), but it is possible that a server does not know whether the client has Handshake keys. Under these circumstances, a server SHOULD send a CONNECTION_CLOSE frame in both Handshake and Initial packets to ensure that at least one of them is processable by the client. These packets can be coalesced into a single UDP datagram (see Section 12.2).
-A stateless reset is provided as an option of last resort for an endpoint that does not have access to the state of a connection. A crash or outage might result in peers continuing to send data to an endpoint that is unable to properly continue the connection. An endpoint MAY send a stateless reset in response to receiving a packet that it cannot associate with an active connection.
-A stateless reset is not appropriate for signaling error conditions. An endpoint that wishes to communicate a fatal connection error MUST use a CONNECTION_CLOSE frame if it has sufficient state to do so.
-To support this process, a token is sent by endpoints. The token is carried in the Stateless Reset Token field of a NEW_CONNECTION_ID frame. Servers can also specify a stateless_reset_token transport parameter during the handshake that applies to the connection ID that it selected during the handshake; clients cannot use this transport parameter because their transport parameters don’t have confidentiality protection. These tokens are protected by encryption, so only client and server know their value. Tokens are invalidated when their associated connection ID is retired via a RETIRE_CONNECTION_ID frame (Section 19.16).
-An endpoint that receives packets that it cannot process sends a packet in the following layout:
- - +New packets from unverified addresses could be used to create an amplification +attack (see Section 8). To avoid this, endpoints MUST either limit +transmission of CONNECTION_CLOSE frames to validated addresses or drop packets +without response if the response would be more than three times larger than the +received packet.¶
+After receiving a CONNECTION_CLOSE frame, endpoints enter the draining state. +An endpoint that receives a CONNECTION_CLOSE frame MAY send a single packet +containing a CONNECTION_CLOSE frame before entering the draining state, using a +CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An endpoint MUST NOT +send further packets, which could result in a constant exchange of +CONNECTION_CLOSE frames until the closing period on either peer ended.¶
+An immediate close can be used after an application protocol has arranged to +close a connection. This might be after the application protocols negotiates a +graceful shutdown. The application protocol exchanges whatever messages that +are needed to cause both endpoints to agree to close the connection, after which +the application requests that the connection be closed. The application +protocol can use a CONNECTION_CLOSE frame with an appropriate error code to +signal closure.¶
+When sending CONNECTION_CLOSE, the goal is to ensure that the peer will process +the frame. Generally, this means sending the frame in a packet with the highest +level of packet protection to avoid the packet being discarded. After the +handshake is confirmed (see Section 4.1.2 of [QUIC-TLS]), an endpoint MUST +send any CONNECTION_CLOSE frames in a 1-RTT packet. However, prior to +confirming the handshake, it is possible that more advanced packet protection +keys are not available to the peer, so the frame MAY be replicated in a packet +that uses a lower packet protection level.¶
+A client will always know whether the server has Handshake keys (see +Section 17.2.2.1), but it is possible that a server does not know whether the +client has Handshake keys. Under these circumstances, a server SHOULD send a +CONNECTION_CLOSE frame in both Handshake and Initial packets to ensure that at +least one of them is processable by the client. Similarly, a peer might be +unable to read 1-RTT packets, so an endpoint SHOULD send CONNECTION_CLOSE in +Handshake and 1-RTT packets prior to confirming the handshake. These packets +can be coalesced into a single UDP datagram; see Section 12.2.¶
+A stateless reset is provided as an option of last resort for an endpoint that +does not have access to the state of a connection. A crash or outage might +result in peers continuing to send data to an endpoint that is unable to +properly continue the connection. An endpoint MAY send a stateless reset in +response to receiving a packet that it cannot associate with an active +connection.¶
+A stateless reset is not appropriate for signaling error conditions. An +endpoint that wishes to communicate a fatal connection error MUST use a +CONNECTION_CLOSE frame if it has sufficient state to do so.¶
+To support this process, a token is sent by endpoints. The token is carried in +the Stateless Reset Token field of a NEW_CONNECTION_ID frame. Servers can also +specify a stateless_reset_token transport parameter during the handshake that +applies to the connection ID that it selected during the handshake; clients +cannot use this transport parameter because their transport parameters don't +have confidentiality protection. These tokens are protected by encryption, so +only client and server know their value. Tokens are invalidated when their +associated connection ID is retired via a RETIRE_CONNECTION_ID frame +(Section 19.16).¶
+An endpoint that receives packets that it cannot process sends a packet in the +following layout:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -1902,119 +4180,402 @@-| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: Stateless Reset Packet
-This design ensures that a stateless reset packet is - to the extent possible - indistinguishable from a regular packet with a short header.
-A stateless reset uses an entire UDP datagram, starting with the first two bits of the packet header. The remainder of the first byte and an arbitrary number of bytes following it that are set to unpredictable values. The last 16 bytes of the datagram contain a Stateless Reset Token.
-To entities other than its intended recipient, a stateless reset will appear to be a packet with a short header. For the stateless reset to appear as a valid QUIC packet, the Unpredictable Bits field needs to include at least 38 bits of data (or 5 bytes, less the two fixed bits).
-A minimum size of 21 bytes does not guarantee that a stateless reset is difficult to distinguish from other packets if the recipient requires the use of a connection ID. To prevent a resulting stateless reset from being trivially distinguishable from a valid packet, all packets sent by an endpoint SHOULD be padded to at least 22 bytes longer than the minimum connection ID that the endpoint might use. An endpoint that sends a stateless reset in response to packet that is 43 bytes or less in length SHOULD send a stateless reset that is one byte shorter than the packet it responds to.
-These values assume that the Stateless Reset Token is the same as the minimum expansion of the packet protection AEAD. Additional unpredictable bytes are necessary if the endpoint could have negotiated a packet protection scheme with a larger minimum expansion.
-An endpoint MUST NOT send a stateless reset that is three times or more larger than the packet it receives to avoid being used for amplification. Section 10.4.3 describes additional limits on stateless reset size.
-Endpoints MUST discard packets that are too small to be valid QUIC packets. With the set of AEAD functions defined in [QUIC-TLS], packets that are smaller than 21 bytes are never valid.
-Endpoints MUST send stateless reset packets formatted as a packet with a short header. However, endpoints MUST treat any packet ending in a valid stateless reset token as a stateless reset, as other QUIC versions might allow the use of a long header.
-An endpoint MAY send a stateless reset in response to a packet with a long header. Sending a stateless reset is not effective prior to the stateless reset token being available to a peer. In this QUIC version, packets with a long header are only used during connection establishment. Because the stateless reset token is not available until connection establishment is complete or near completion, ignoring an unknown packet with a long header might be as effective as sending a stateless reset.
-An endpoint cannot determine the Source Connection ID from a packet with a short header, therefore it cannot set the Destination Connection ID in the stateless reset packet. The Destination Connection ID will therefore differ from the value used in previous packets. A random Destination Connection ID makes the connection ID appear to be the result of moving to a new connection ID that was provided using a NEW_CONNECTION_ID frame (Section 19.15).
-Using a randomized connection ID results in two problems:
- - +This design ensures that a stateless reset packet is - to the extent possible - +indistinguishable from a regular packet with a short header.¶
+A stateless reset uses an entire UDP datagram, starting with the first two bits +of the packet header. The remainder of the first byte and an arbitrary number +of bytes following it that are set to unpredictable values. The last 16 bytes +of the datagram contain a Stateless Reset Token.¶
+To entities other than its intended recipient, a stateless reset will appear to +be a packet with a short header. For the stateless reset to appear as a valid +QUIC packet, the Unpredictable Bits field needs to include at least 38 bits of +data (or 5 bytes, less the two fixed bits).¶
+A minimum size of 21 bytes does not guarantee that a stateless reset is +difficult to distinguish from other packets if the recipient requires the use of +a connection ID. To prevent a resulting stateless reset from being trivially +distinguishable from a valid packet, all packets sent by an endpoint SHOULD be +padded to at least 22 bytes longer than the minimum connection ID that the +endpoint might use. An endpoint that sends a stateless reset in response to +packet that is 43 bytes or less in length SHOULD send a stateless reset that is +one byte shorter than the packet it responds to.¶
+These values assume that the Stateless Reset Token is the same as the minimum +expansion of the packet protection AEAD. Additional unpredictable bytes are +necessary if the endpoint could have negotiated a packet protection scheme with +a larger minimum expansion.¶
+An endpoint MUST NOT send a stateless reset that is three times or more larger +than the packet it receives to avoid being used for amplification. +Section 10.4.3 describes additional limits on stateless reset size.¶
+Endpoints MUST discard packets that are too small to be valid QUIC packets. +With the set of AEAD functions defined in [QUIC-TLS], packets that are smaller +than 21 bytes are never valid.¶
+Endpoints MUST send stateless reset packets formatted as a packet with a short +header. However, endpoints MUST treat any packet ending in a valid stateless +reset token as a stateless reset, as other QUIC versions might allow the use of +a long header.¶
+An endpoint MAY send a stateless reset in response to a packet with a long +header. Sending a stateless reset is not effective prior to the stateless reset +token being available to a peer. In this QUIC version, packets with a long +header are only used during connection establishment. Because the stateless +reset token is not available until connection establishment is complete or near +completion, ignoring an unknown packet with a long header might be as effective +as sending a stateless reset.¶
+An endpoint cannot determine the Source Connection ID from a packet with a short +header, therefore it cannot set the Destination Connection ID in the stateless +reset packet. The Destination Connection ID will therefore differ from the +value used in previous packets. A random Destination Connection ID makes the +connection ID appear to be the result of moving to a new connection ID that was +provided using a NEW_CONNECTION_ID frame (Section 19.15).¶
+Using a randomized connection ID results in two problems:¶
This stateless reset design is specific to QUIC version 1. An endpoint that supports multiple versions of QUIC needs to generate a stateless reset that will be accepted by peers that support any version that the endpoint might support (or might have supported prior to losing state). Designers of new versions of QUIC need to be aware of this and either reuse this design, or use a portion of the packet other than the last 16 bytes for carrying data.
-An endpoint detects a potential stateless reset using the trailing 16 bytes of the UDP datagram. An endpoint remembers all Stateless Reset Tokens associated with the connection IDs and remote addresses for datagrams it has recently sent. This includes Stateless Reset Tokens from NEW_CONNECTION_ID frames and the server’s transport parameters but excludes Stateless Reset Tokens associated with connection IDs that are either unused or retired. The endpoint identifies a received datagram as a stateless reset by comparing the last 16 bytes of the datagram with all Stateless Reset Tokens associated with the remote address on which the datagram was received.
-This comparison can be performed for every inbound datagram. Endpoints MAY skip this check if any packet from a datagram is successfully processed. However, the comparison MUST be performed when the first packet in an incoming datagram either cannot be associated with a connection, or cannot be decrypted.
-An endpoint MUST NOT check for any Stateless Reset Tokens associated with connection IDs it has not used or for connection IDs that have been retired.
-When comparing a datagram to Stateless Reset Token values, endpoints MUST perform the comparison without leaking information about the value of the token. For example, performing this comparison in constant time protects the value of individual Stateless Reset Tokens from information leakage through timing side channels. Another approach would be to store and compare the transformed values of Stateless Reset Tokens instead of the raw token values, where the transformation is defined as a cryptographically-secure pseudo-random function using a secret key (e.g., block cipher, HMAC [RFC2104]). An endpoint is not expected to protect information about whether a packet was successfully decrypted, or the number of valid Stateless Reset Tokens.
-If the last 16 bytes of the datagram are identical in value to a Stateless Reset Token, the endpoint MUST enter the draining period and not send any further packets on this connection.
-The stateless reset token MUST be difficult to guess. In order to create a Stateless Reset Token, an endpoint could randomly generate [RFC4086] a secret for every connection that it creates. However, this presents a coordination problem when there are multiple instances in a cluster or a storage problem for an endpoint that might lose state. Stateless reset specifically exists to handle the case where state is lost, so this approach is suboptimal.
-A single static key can be used across all connections to the same endpoint by generating the proof using a second iteration of a preimage-resistant function that takes a static key and the connection ID chosen by the endpoint (see Section 5.1) as input. An endpoint could use HMAC [RFC2104] (for example, HMAC(static_key, connection_id)) or HKDF [RFC5869] (for example, using the static key as input keying material, with the connection ID as salt). The output of this function is truncated to 16 bytes to produce the Stateless Reset Token for that connection.
-An endpoint that loses state can use the same method to generate a valid Stateless Reset Token. The connection ID comes from the packet that the endpoint receives.
-This design relies on the peer always sending a connection ID in its packets so that the endpoint can use the connection ID from a packet to reset the connection. An endpoint that uses this design MUST either use the same connection ID length for all connections or encode the length of the connection ID such that it can be recovered without state. In addition, it cannot provide a zero-length connection ID.
-Revealing the Stateless Reset Token allows any entity to terminate the connection, so a value can only be used once. This method for choosing the Stateless Reset Token means that the combination of connection ID and static key MUST NOT be used for another connection. A denial of service attack is possible if the same connection ID is used by instances that share a static key, or if an attacker can cause a packet to be routed to an instance that has no state but the same static key; see Section 21.9. A connection ID from a connection that is reset by revealing the Stateless Reset Token MUST NOT be reused for new connections at nodes that share a static key.
-The same Stateless Reset Token MUST NOT be used for multiple connection IDs. Endpoints are not required to compare new values against all previous values, but a duplicate value MAY be treated as a connection error of type PROTOCOL_VIOLATION.
-Note that Stateless Reset packets do not have any cryptographic protection.
-The design of a Stateless Reset is such that without knowing the stateless reset token it is indistinguishable from a valid packet. For instance, if a server sends a Stateless Reset to another server it might receive another Stateless Reset in response, which could lead to an infinite exchange.
-An endpoint MUST ensure that every Stateless Reset that it sends is smaller than the packet which triggered it, unless it maintains state sufficient to prevent looping. In the event of a loop, this results in packets eventually being too small to trigger a response.
-An endpoint can remember the number of Stateless Reset packets that it has sent and stop generating new Stateless Reset packets once a limit is reached. Using separate limits for different remote addresses will ensure that Stateless Reset packets can be used to close connections when other peers or connections have exhausted limits.
-Reducing the size of a Stateless Reset below 41 bytes means that the packet could reveal to an observer that it is a Stateless Reset, depending upon the length of the peer’s connection IDs. Conversely, refusing to send a Stateless Reset in response to a small packet might result in Stateless Reset not being useful in detecting cases of broken connections where only very small packets are sent; such failures might only be detected by other means, such as timers.
-An endpoint that detects an error SHOULD signal the existence of that error to its peer. Both transport-level and application-level errors can affect an entire connection (see Section 11.1), while only application-level errors can be isolated to a single stream (see Section 11.2).
-The most appropriate error code (Section 20) SHOULD be included in the frame that signals the error. Where this specification identifies error conditions, it also identifies the error code that is used; though these are worded as requirements, different implementation strategies might lead to different errors being reported. In particular, an endpoint MAY use any applicable error code when it detects an error condition; a generic error code (such as PROTOCOL_VIOLATION or INTERNAL_ERROR) can always be used in place of specific error codes.
-A stateless reset (Section 10.4) is not suitable for any error that can be signaled with a CONNECTION_CLOSE or RESET_STREAM frame. A stateless reset MUST NOT be used by an endpoint that has the state necessary to send a frame on the connection.
-Errors that result in the connection being unusable, such as an obvious violation of protocol semantics or corruption of state that affects an entire connection, MUST be signaled using a CONNECTION_CLOSE frame (Section 19.19). An endpoint MAY close the connection in this manner even if the error only affects a single stream.
-Application protocols can signal application-specific protocol errors using the application-specific variant of the CONNECTION_CLOSE frame. Errors that are specific to the transport, including all those described in this document, are carried in the QUIC-specific variant of the CONNECTION_CLOSE frame.
-A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if it receives more packets on a terminated connection. Limiting the number of retransmissions and the time over which this final packet is sent limits the effort expended on terminated connections.
-An endpoint that chooses not to retransmit packets containing a CONNECTION_CLOSE frame risks a peer missing the first such packet. The only mechanism available to an endpoint that continues to receive data for a terminated connection is to use the stateless reset process (Section 10.4).
-An endpoint that receives an invalid CONNECTION_CLOSE frame MUST NOT signal the existence of the error to its peer.
-If an application-level error affects a single stream, but otherwise leaves the connection in a recoverable state, the endpoint can send a RESET_STREAM frame (Section 19.4) with an appropriate error code to terminate just the affected stream.
-RESET_STREAM MUST be instigated by the protocol using QUIC. RESET_STREAM carries an application error code. Only the application protocol is able to cause a stream to be terminated. A local instance of the application protocol uses a direct API call and a remote instance uses the STOP_SENDING frame, which triggers an automatic RESET_STREAM.
-Resetting a stream without knowledge of the application protocol could cause the protocol to enter an unrecoverable state. Application protocols might require certain streams to be reliably delivered in order to guarantee consistent state between endpoints. Application protocols SHOULD define rules for handling streams that are prematurely cancelled by either endpoint.
-QUIC endpoints communicate by exchanging packets. Packets have confidentiality and integrity protection (see Section 12.1) and are carried in UDP datagrams (see Section 12.2).
-This version of QUIC uses the long packet header (see Section 17.2) during connection establishment. Packets with the long header are Initial (Section 17.2.2), 0-RTT (Section 17.2.3), Handshake (Section 17.2.4), and Retry (Section 17.2.5). Version negotiation uses a version-independent packet with a long header (see Section 17.2.1).
-Packets with the short header (Section 17.3) are designed for minimal overhead and are used after a connection is established and 1-RTT keys are available.
-All QUIC packets except Version Negotiation and Retry packets use authenticated encryption with additional data (AEAD) [RFC5116] to provide confidentiality and integrity protection. Details of packet protection are found in [QUIC-TLS]; this section includes an overview of the process.
-Initial packets are protected using keys that are statically derived. This packet protection is not effective confidentiality protection. Initial protection only exists to ensure that the sender of the packet is on the network path. Any entity that receives the Initial packet from a client can recover the keys necessary to remove packet protection or to generate packets that will be successfully authenticated.
-All other packets are protected with keys derived from the cryptographic handshake. The type of the packet from the long header or key phase from the short header are used to identify which encryption level - and therefore the keys - that are used. Packets protected with 0-RTT and 1-RTT keys are expected to have confidentiality and data origin authentication; the cryptographic handshake ensures that only the communicating endpoints receive the corresponding keys.
-The packet number field contains a packet number, which has additional confidentiality protection that is applied after packet protection is applied (see [QUIC-TLS] for details). The underlying packet number increases with each packet sent in a given packet number space; see Section 12.3 for details.
-Initial (Section 17.2.2), 0-RTT (Section 17.2.3), and Handshake (Section 17.2.4) packets contain a Length field, which determines the end of the packet. The length includes both the Packet Number and Payload fields, both of which are confidentiality protected and initially of unknown length. The length of the Payload field is learned once header protection is removed.
-Using the Length field, a sender can coalesce multiple QUIC packets into one UDP datagram. This can reduce the number of UDP datagrams needed to complete the cryptographic handshake and start sending data. This can also be used to construct PMTU probes (see Section 14.3.1). Receivers MUST be able to process coalesced packets.
-Coalescing packets in order of increasing encryption levels (Initial, 0-RTT, Handshake, 1-RTT) makes it more likely the receiver will be able to process all the packets in a single pass. A packet with a short header does not include a length, so it can only be the last packet included in a UDP datagram. An endpoint SHOULD NOT coalesce multiple packets at the same encryption level.
-Senders MUST NOT coalesce QUIC packets for different connections into a single UDP datagram. Receivers SHOULD ignore any subsequent packets with a different Destination Connection ID than the first packet in the datagram.
-Every QUIC packet that is coalesced into a single UDP datagram is separate and complete. The receiver of coalesced QUIC packets MUST individually process each QUIC packet and separately acknowledge them, as if they were received as the payload of different UDP datagrams. For example, if decryption fails (because the keys are not available or any other reason), the receiver MAY either discard or buffer the packet for later processing and MUST attempt to process the remaining packets.
-Retry packets (Section 17.2.5), Version Negotiation packets (Section 17.2.1), and packets with a short header (Section 17.3) do not contain a Length field and so cannot be followed by other packets in the same UDP datagram. Note also that there is no situation where a Retry or Version Negotiation packet is coalesced with another packet.
-The packet number is an integer in the range 0 to 2^62-1. This number is used in determining the cryptographic nonce for packet protection. Each endpoint maintains a separate packet number for sending and receiving.
-Packet numbers are limited to this range because they need to be representable in whole in the Largest Acknowledged field of an ACK frame (Section 19.3). When present in a long or short header however, packet numbers are reduced and encoded in 1 to 4 bytes (see Section 17.1).
-Version Negotiation (Section 17.2.1) and Retry (Section 17.2.5) packets do not include a packet number.
-Packet numbers are divided into 3 spaces in QUIC:
- - +This stateless reset design is specific to QUIC version 1. An endpoint that +supports multiple versions of QUIC needs to generate a stateless reset that will +be accepted by peers that support any version that the endpoint might support +(or might have supported prior to losing state). Designers of new versions of +QUIC need to be aware of this and either reuse this design, or use a portion of +the packet other than the last 16 bytes for carrying data.¶
+An endpoint detects a potential stateless reset using the trailing 16 bytes of +the UDP datagram. An endpoint remembers all Stateless Reset Tokens associated +with the connection IDs and remote addresses for datagrams it has recently sent. +This includes Stateless Reset Tokens from NEW_CONNECTION_ID frames and the +server's transport parameters but excludes Stateless Reset Tokens associated +with connection IDs that are either unused or retired. The endpoint identifies +a received datagram as a stateless reset by comparing the last 16 bytes of the +datagram with all Stateless Reset Tokens associated with the remote address on +which the datagram was received.¶
+This comparison can be performed for every inbound datagram. Endpoints MAY skip +this check if any packet from a datagram is successfully processed. However, +the comparison MUST be performed when the first packet in an incoming datagram +either cannot be associated with a connection, or cannot be decrypted.¶
+An endpoint MUST NOT check for any Stateless Reset Tokens associated with +connection IDs it has not used or for connection IDs that have been retired.¶
+When comparing a datagram to Stateless Reset Token values, endpoints MUST +perform the comparison without leaking information about the value of the token. +For example, performing this comparison in constant time protects the value of +individual Stateless Reset Tokens from information leakage through timing side +channels. Another approach would be to store and compare the transformed values +of Stateless Reset Tokens instead of the raw token values, where the +transformation is defined as a cryptographically-secure pseudo-random function +using a secret key (e.g., block cipher, HMAC [RFC2104]). An endpoint is not +expected to protect information about whether a packet was successfully +decrypted, or the number of valid Stateless Reset Tokens.¶
+If the last 16 bytes of the datagram are identical in value to a Stateless Reset +Token, the endpoint MUST enter the draining period and not send any further +packets on this connection.¶
+The stateless reset token MUST be difficult to guess. In order to create a +Stateless Reset Token, an endpoint could randomly generate [RFC4086] a secret +for every connection that it creates. However, this presents a coordination +problem when there are multiple instances in a cluster or a storage problem for +an endpoint that might lose state. Stateless reset specifically exists to +handle the case where state is lost, so this approach is suboptimal.¶
+A single static key can be used across all connections to the same endpoint by +generating the proof using a second iteration of a preimage-resistant function +that takes a static key and the connection ID chosen by the endpoint (see +Section 5.1) as input. An endpoint could use HMAC [RFC2104] (for +example, HMAC(static_key, connection_id)) or HKDF [RFC5869] (for example, +using the static key as input keying material, with the connection ID as salt). +The output of this function is truncated to 16 bytes to produce the Stateless +Reset Token for that connection.¶
+An endpoint that loses state can use the same method to generate a valid +Stateless Reset Token. The connection ID comes from the packet that the +endpoint receives.¶
+This design relies on the peer always sending a connection ID in its packets so +that the endpoint can use the connection ID from a packet to reset the +connection. An endpoint that uses this design MUST either use the same +connection ID length for all connections or encode the length of the connection +ID such that it can be recovered without state. In addition, it cannot provide +a zero-length connection ID.¶
+Revealing the Stateless Reset Token allows any entity to terminate the +connection, so a value can only be used once. This method for choosing the +Stateless Reset Token means that the combination of connection ID and static key +MUST NOT be used for another connection. A denial of service attack is possible +if the same connection ID is used by instances that share a static key, or if an +attacker can cause a packet to be routed to an instance that has no state but +the same static key; see Section 21.9. A connection ID from a connection +that is reset by revealing the Stateless Reset Token MUST NOT be reused for new +connections at nodes that share a static key.¶
+The same Stateless Reset Token MUST NOT be used for multiple connection IDs. +Endpoints are not required to compare new values against all previous values, +but a duplicate value MAY be treated as a connection error of type +PROTOCOL_VIOLATION.¶
+Note that Stateless Reset packets do not have any cryptographic protection.¶
+The design of a Stateless Reset is such that without knowing the stateless reset +token it is indistinguishable from a valid packet. For instance, if a server +sends a Stateless Reset to another server it might receive another Stateless +Reset in response, which could lead to an infinite exchange.¶
+An endpoint MUST ensure that every Stateless Reset that it sends is smaller than +the packet which triggered it, unless it maintains state sufficient to prevent +looping. In the event of a loop, this results in packets eventually being too +small to trigger a response.¶
+An endpoint can remember the number of Stateless Reset packets that it has sent +and stop generating new Stateless Reset packets once a limit is reached. Using +separate limits for different remote addresses will ensure that Stateless Reset +packets can be used to close connections when other peers or connections have +exhausted limits.¶
+Reducing the size of a Stateless Reset below 41 bytes means that the packet +could reveal to an observer that it is a Stateless Reset, depending upon the +length of the peer's connection IDs. Conversely, refusing to send a Stateless +Reset in response to a small packet might result in Stateless Reset not being +useful in detecting cases of broken connections where only very small packets +are sent; such failures might only be detected by other means, such as timers.¶
+An endpoint that detects an error SHOULD signal the existence of that error to +its peer. Both transport-level and application-level errors can affect an +entire connection (see Section 11.1), while only application-level +errors can be isolated to a single stream (see Section 11.2).¶
+The most appropriate error code (Section 20) SHOULD be included in the +frame that signals the error. Where this specification identifies error +conditions, it also identifies the error code that is used; though these are +worded as requirements, different implementation strategies might lead to +different errors being reported. In particular, an endpoint MAY use any +applicable error code when it detects an error condition; a generic error code +(such as PROTOCOL_VIOLATION or INTERNAL_ERROR) can always be used in place of +specific error codes.¶
+A stateless reset (Section 10.4) is not suitable for any error that can +be signaled with a CONNECTION_CLOSE or RESET_STREAM frame. A stateless reset +MUST NOT be used by an endpoint that has the state necessary to send a frame on +the connection.¶
+Errors that result in the connection being unusable, such as an obvious +violation of protocol semantics or corruption of state that affects an entire +connection, MUST be signaled using a CONNECTION_CLOSE frame +(Section 19.19). An endpoint MAY close the connection in this +manner even if the error only affects a single stream.¶
+Application protocols can signal application-specific protocol errors using the +application-specific variant of the CONNECTION_CLOSE frame. Errors that are +specific to the transport, including all those described in this document, are +carried in the QUIC-specific variant of the CONNECTION_CLOSE frame.¶
+A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint +SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if +it receives more packets on a terminated connection. Limiting the number of +retransmissions and the time over which this final packet is sent limits the +effort expended on terminated connections.¶
+An endpoint that chooses not to retransmit packets containing a CONNECTION_CLOSE +frame risks a peer missing the first such packet. The only mechanism available +to an endpoint that continues to receive data for a terminated connection is to +use the stateless reset process (Section 10.4).¶
+If an application-level error affects a single stream, but otherwise leaves the +connection in a recoverable state, the endpoint can send a RESET_STREAM frame +(Section 19.4) with an appropriate error code to terminate just the +affected stream.¶
+Resetting a stream without the involvement of the application protocol could +cause the application protocol to enter an unrecoverable state. RESET_STREAM +MUST only be instigated by the application protocol that uses QUIC.¶
+RESET_STREAM carries an application error code, for which the semantics are +defined by the application protocol. Only the application protocol is able to +cause a stream to be terminated. A local instance of the application protocol +uses a direct API call and a remote instance uses the STOP_SENDING frame, which +triggers an automatic RESET_STREAM.¶
+Application protocols SHOULD define rules for handling streams that are +prematurely cancelled by either endpoint.¶
+QUIC endpoints communicate by exchanging packets. Packets have confidentiality +and integrity protection (see Section 12.1) and are carried in UDP +datagrams (see Section 12.2).¶
+This version of QUIC uses the long packet header (see Section 17.2) during +connection establishment. Packets with the long header are Initial +(Section 17.2.2), 0-RTT (Section 17.2.3), Handshake (Section 17.2.4), +and Retry (Section 17.2.5). Version negotiation uses a version-independent +packet with a long header (see Section 17.2.1).¶
+Packets with the short header (Section 17.3) are designed for minimal +overhead and are used after a connection is established and 1-RTT keys are +available.¶
+All QUIC packets except Version Negotiation and Retry packets use authenticated +encryption with additional data (AEAD) [RFC5116] to provide confidentiality +and integrity protection. Details of packet protection are found in +[QUIC-TLS]; this section includes an overview of the process.¶
+Initial packets are protected using keys that are statically derived. This +packet protection is not effective confidentiality protection. Initial +protection only exists to ensure that the sender of the packet is on the network +path. Any entity that receives the Initial packet from a client can recover the +keys necessary to remove packet protection or to generate packets that will be +successfully authenticated.¶
+All other packets are protected with keys derived from the cryptographic +handshake. The type of the packet from the long header or key phase from the +short header are used to identify which encryption level - and therefore the +keys - that are used. Packets protected with 0-RTT and 1-RTT keys are expected +to have confidentiality and data origin authentication; the cryptographic +handshake ensures that only the communicating endpoints receive the +corresponding keys.¶
+The packet number field contains a packet number, which has additional +confidentiality protection that is applied after packet protection is applied +(see [QUIC-TLS] for details). The underlying packet number increases with +each packet sent in a given packet number space; see Section 12.3 for +details.¶
+Initial (Section 17.2.2), 0-RTT (Section 17.2.3), and Handshake +(Section 17.2.4) packets contain a Length field, which determines the end +of the packet. The length includes both the Packet Number and Payload +fields, both of which are confidentiality protected and initially of unknown +length. The length of the Payload field is learned once header protection is +removed.¶
+Using the Length field, a sender can coalesce multiple QUIC packets into one UDP +datagram. This can reduce the number of UDP datagrams needed to complete the +cryptographic handshake and start sending data. This can also be used to +construct PMTU probes (see Section 14.3.1). Receivers MUST be able to +process coalesced packets.¶
+Coalescing packets in order of increasing encryption levels (Initial, 0-RTT, +Handshake, 1-RTT) makes it more likely the receiver will be able to process all +the packets in a single pass. A packet with a short header does not include a +length, so it can only be the last packet included in a UDP datagram. An +endpoint SHOULD NOT coalesce multiple packets at the same encryption level.¶
+Senders MUST NOT coalesce QUIC packets for different connections into a single +UDP datagram. Receivers SHOULD ignore any subsequent packets with a different +Destination Connection ID than the first packet in the datagram.¶
+Every QUIC packet that is coalesced into a single UDP datagram is separate and +complete. The receiver of coalesced QUIC packets MUST individually process each +QUIC packet and separately acknowledge them, as if they were received as the +payload of different UDP datagrams. For example, if decryption fails (because +the keys are not available or any other reason), the receiver MAY either discard +or buffer the packet for later processing and MUST attempt to process the +remaining packets.¶
+Retry packets (Section 17.2.5), Version Negotiation packets +(Section 17.2.1), and packets with a short header (Section 17.3) do not +contain a Length field and so cannot be followed by other packets in the same +UDP datagram. Note also that there is no situation where a Retry or Version +Negotiation packet is coalesced with another packet.¶
+The packet number is an integer in the range 0 to 2^62-1. This number is used +in determining the cryptographic nonce for packet protection. Each endpoint +maintains a separate packet number for sending and receiving.¶
+Packet numbers are limited to this range because they need to be representable +in whole in the Largest Acknowledged field of an ACK frame (Section 19.3). +When present in a long or short header however, packet numbers are reduced and +encoded in 1 to 4 bytes (see Section 17.1).¶
+Version Negotiation (Section 17.2.1) and Retry (Section 17.2.5) packets +do not include a packet number.¶
+Packet numbers are divided into 3 spaces in QUIC:¶
As described in [QUIC-TLS], each packet type uses different protection keys.
-Conceptually, a packet number space is the context in which a packet can be processed and acknowledged. Initial packets can only be sent with Initial packet protection keys and acknowledged in packets which are also Initial packets. Similarly, Handshake packets are sent at the Handshake encryption level and can only be acknowledged in Handshake packets.
-This enforces cryptographic separation between the data sent in the different packet sequence number spaces. Packet numbers in each space start at packet number 0. Subsequent packets sent in the same packet number space MUST increase the packet number by at least one.
-0-RTT and 1-RTT data exist in the same packet number space to make loss recovery algorithms easier to implement between the two packet types.
-A QUIC endpoint MUST NOT reuse a packet number within the same packet number space in one connection. If the packet number for sending reaches 2^62 - 1, the sender MUST close the connection without sending a CONNECTION_CLOSE frame or any further packets; an endpoint MAY send a Stateless Reset (Section 10.4) in response to further packets that it receives.
-A receiver MUST discard a newly unprotected packet unless it is certain that it has not processed another packet with the same packet number from the same packet number space. Duplicate suppression MUST happen after removing packet protection for the reasons described in Section 9.3 of [QUIC-TLS]. An efficient algorithm for duplicate suppression can be found in Section 3.4.3 of [RFC4303].
-Packet number encoding at a sender and decoding at a receiver are described in Section 17.1.
-The payload of QUIC packets, after removing packet protection, consists of a sequence of complete frames, as shown in Figure 7. Version Negotiation, Stateless Reset, and Retry packets do not contain frames.
- - +As described in [QUIC-TLS], each packet type uses different protection keys.¶
+Conceptually, a packet number space is the context in which a packet can be +processed and acknowledged. Initial packets can only be sent with Initial +packet protection keys and acknowledged in packets which are also Initial +packets. Similarly, Handshake packets are sent at the Handshake encryption +level and can only be acknowledged in Handshake packets.¶
+This enforces cryptographic separation between the data sent in the different +packet sequence number spaces. Packet numbers in each space start at packet +number 0. Subsequent packets sent in the same packet number space MUST increase +the packet number by at least one.¶
+0-RTT and 1-RTT data exist in the same packet number space to make loss recovery +algorithms easier to implement between the two packet types.¶
+A QUIC endpoint MUST NOT reuse a packet number within the same packet number +space in one connection. If the packet number for sending reaches 2^62 - 1, the +sender MUST close the connection without sending a CONNECTION_CLOSE frame or any +further packets; an endpoint MAY send a Stateless Reset (Section 10.4) in +response to further packets that it receives.¶
+A receiver MUST discard a newly unprotected packet unless it is certain that it +has not processed another packet with the same packet number from the same +packet number space. Duplicate suppression MUST happen after removing packet +protection for the reasons described in Section 9.3 of [QUIC-TLS]. An +efficient algorithm for duplicate suppression can be found in Section 3.4.3 of +[RFC4303].¶
+Packet number encoding at a sender and decoding at a receiver are described in +Section 17.1.¶
+The payload of QUIC packets, after removing packet protection, consists of a +sequence of complete frames, as shown in Figure 7. Version +Negotiation, Stateless Reset, and Retry packets do not contain frames.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2028,11 +4589,19 @@-| Frame N (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: QUIC Payload
-The payload of a packet that contains frames MUST contain at least one frame, and MAY contain multiple frames and multiple frame types. Frames always fit within a single QUIC packet and cannot span multiple packets.
-Each frame begins with a Frame Type, indicating its type, followed by additional type-dependent fields:
- - +The payload of a packet that contains frames MUST contain at least one frame, +and MAY contain multiple frames and multiple frame types. Frames always fit +within a single QUIC packet and cannot span multiple packets.¶
+Each frame begins with a Frame Type, indicating its type, followed by +additional type-dependent fields:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2042,376 +4611,996 @@-| Type-Dependent Fields (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: Generic Frame Layout
-The frame types defined in this specification are listed in Table 3. The Frame Type in ACK, STREAM, MAX_STREAMS, STREAMS_BLOCKED, and CONNECTION_CLOSE frames is used to carry other frame-specific flags. For all other frames, the Frame Type field simply identifies the frame. These frames are explained in more detail in Section 19.
- - -| Type Value | -Frame Type Name | -Definition | -
|---|---|---|
| 0x00 | -PADDING | -Section 19.1 | -
| 0x01 | -PING | -Section 19.2 | -
| 0x02 - 0x03 | -ACK | -Section 19.3 | -
| 0x04 | -RESET_STREAM | -Section 19.4 | -
| 0x05 | -STOP_SENDING | -Section 19.5 | -
| 0x06 | -CRYPTO | -Section 19.6 | -
| 0x07 | -NEW_TOKEN | -Section 19.7 | -
| 0x08 - 0x0f | -STREAM | -Section 19.8 | -
| 0x10 | -MAX_DATA | -Section 19.9 | -
| 0x11 | -MAX_STREAM_DATA | -Section 19.10 | -
| 0x12 - 0x13 | -MAX_STREAMS | -Section 19.11 | -
| 0x14 | -DATA_BLOCKED | -Section 19.12 | -
| 0x15 | -STREAM_DATA_BLOCKED | -Section 19.13 | -
| 0x16 - 0x17 | -STREAMS_BLOCKED | -Section 19.14 | -
| 0x18 | -NEW_CONNECTION_ID | -Section 19.15 | -
| 0x19 | -RETIRE_CONNECTION_ID | -Section 19.16 | -
| 0x1a | -PATH_CHALLENGE | -Section 19.17 | -
| 0x1b | -PATH_RESPONSE | -Section 19.18 | -
| 0x1c - 0x1d | -CONNECTION_CLOSE | -Section 19.19 | -
An endpoint MUST treat the receipt of a frame of unknown type as a connection error of type FRAME_ENCODING_ERROR.
-All QUIC frames are idempotent in this version of QUIC. That is, a valid frame does not cause undesirable side effects or errors when received more than once.
-The Frame Type field uses a variable length integer encoding (see Section 16) with one exception. To ensure simple and efficient implementations of frame parsing, a frame type MUST use the shortest possible encoding. Though a two-, four- or eight-byte encoding of the frame types defined in this document is possible, the Frame Type field for these frames is encoded on a single byte. For instance, though 0x4001 is a legitimate two-byte encoding for a variable-length integer with a value of 1, PING frames are always encoded as a single byte with the value 0x01. An endpoint MAY treat the receipt of a frame type that uses a longer encoding than necessary as a connection error of type PROTOCOL_VIOLATION.
-A sender bundles one or more frames in a QUIC packet (see Section 12.4).
-A sender can minimize per-packet bandwidth and computational costs by bundling as many frames as possible within a QUIC packet. A sender MAY wait for a short period of time to bundle multiple frames before sending a packet that is not maximally packed, to avoid sending out large numbers of small packets. An implementation MAY use knowledge about application sending behavior or heuristics to determine whether and for how long to wait. This waiting period is an implementation decision, and an implementation should be careful to delay conservatively, since any delay is likely to increase application-visible latency.
-Stream multiplexing is achieved by interleaving STREAM frames from multiple streams into one or more QUIC packets. A single QUIC packet can include multiple STREAM frames from one or more streams.
-One of the benefits of QUIC is avoidance of head-of-line blocking across multiple streams. When a packet loss occurs, only streams with data in that packet are blocked waiting for a retransmission to be received, while other streams can continue making progress. Note that when data from multiple streams is bundled into a single QUIC packet, loss of that packet blocks all those streams from making progress. Implementations are advised to bundle as few streams as necessary in outgoing packets without losing transmission efficiency to underfilled packets.
-A packet MUST NOT be acknowledged until packet protection has been successfully removed and all frames contained in the packet have been processed. For STREAM frames, this means the data has been enqueued in preparation to be received by the application protocol, but it does not require that data is delivered and consumed.
-Once the packet has been fully processed, a receiver acknowledges receipt by sending one or more ACK frames containing the packet number of the received packet.
-Endpoints acknowledge all packets they receive and process. However, only ack-eliciting packets cause an ACK frame to be sent within the maximum ack delay. Packets that are not ack-eliciting are only acknowledged when an ACK frame is sent for other reasons.
-When sending a packet for any reason, an endpoint should attempt to bundle an ACK frame if one has not been sent recently. Doing so helps with timely loss detection at the peer.
-In general, frequent feedback from a receiver improves loss and congestion response, but this has to be balanced against excessive load generated by a receiver that sends an ACK frame in response to every ack-eliciting packet. The guidance offered below seeks to strike this balance.
-Every packet SHOULD be acknowledged at least once, and ack-eliciting packets MUST be acknowledged at least once within the maximum ack delay. An endpoint communicates its maximum delay using the max_ack_delay transport parameter; see Section 18.2. max_ack_delay declares an explicit contract: an endpoint promises to never intentionally delay acknowledgments of an ack-eliciting packet by more than the indicated value. If it does, any excess accrues to the RTT estimate and could result in spurious or delayed retransmissions from the peer. For Initial and Handshake packets, a max_ack_delay of 0 is used. The sender uses the receiver’s max_ack_delay value in determining timeouts for timer-based retransmission, as detailed in Section 5.2.1 of [QUIC-RECOVERY].
-An ACK frame SHOULD be generated for at least every second ack-eliciting packet. This recommendation is in keeping with standard practice for TCP [RFC5681].
-In order to assist loss detection at the sender, an endpoint SHOULD send an ACK frame immediately on receiving an ack-eliciting packet that is out of order. The endpoint MAY continue sending ACK frames immediately on each subsequently received packet, but the endpoint SHOULD return to acknowledging every other packet within a period of 1/8 x RTT, unless more ack-eliciting packets are received out of order. If every subsequent ack-eliciting packet arrives out of order, then an ACK frame SHOULD be sent immediately for every received ack-eliciting packet.
-Similarly, packets marked with the ECN Congestion Experienced (CE) codepoint in the IP header SHOULD be acknowledged immediately, to reduce the peer’s response time to congestion events.
-As an optimization, a receiver MAY process multiple packets before sending any ACK frames in response. In this case the receiver can determine whether an immediate or delayed acknowledgement should be generated after processing incoming packets.
-Packets containing PADDING frames are considered to be in flight for congestion control purposes [QUIC-RECOVERY]. Sending only PADDING frames might cause the sender to become limited by the congestion controller with no acknowledgments forthcoming from the receiver. Therefore, a sender SHOULD ensure that other frames are sent in addition to PADDING frames to elicit acknowledgments from the receiver.
-An endpoint that is only sending ACK frames will not receive acknowledgments from its peer unless those acknowledgements are included in packets with ack-eliciting frames. An endpoint SHOULD bundle ACK frames with other frames when there are new ack-eliciting packets to acknowledge. When only non-ack-eliciting packets need to be acknowledged, an endpoint MAY wait until an ack-eliciting packet has been received to bundle an ACK frame with outgoing frames.
-The algorithms in [QUIC-RECOVERY] are resilient to receivers that do not follow guidance offered above. However, an implementor should only deviate from these requirements after careful consideration of the performance implications of doing so.
-Packets containing only ACK frames are not congestion controlled, so there are limits on how frequently they can be sent. An endpoint MUST NOT send more than one ACK-frame-only packet in response to receiving an ack-eliciting packet. An endpoint MUST NOT send a non-ack-eliciting packet in response to a non-ack-eliciting packet, even if there are packet gaps which precede the received packet. Limiting ACK frames avoids an infinite feedback loop of acknowledgements, which could prevent the connection from ever becoming idle. However, the endpoint acknowledges non-ACK-eliciting packets when it sends an ACK frame.
-An endpoint SHOULD treat receipt of an acknowledgment for a packet it did not send as a connection error of type PROTOCOL_VIOLATION, if it is able to detect the condition.
-When an ACK frame is sent, one or more ranges of acknowledged packets are included. Including older packets reduces the chance of spurious retransmits caused by losing previously sent ACK frames, at the cost of larger ACK frames.
-ACK frames SHOULD always acknowledge the most recently received packets, and the more out-of-order the packets are, the more important it is to send an updated ACK frame quickly, to prevent the peer from declaring a packet as lost and spuriously retransmitting the frames it contains.
-Section 13.2.3 and Section 13.2.4 describe an exemplary approach for determining what packets to acknowledge in each ACK frame.
-When a packet containing an ACK frame is sent, the largest acknowledged in that frame may be saved. When a packet containing an ACK frame is acknowledged, the receiver can stop acknowledging packets less than or equal to the largest acknowledged in the sent ACK frame.
-In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT of reordering. In cases with ACK frame loss and reordering, this approach does not guarantee that every acknowledgement is seen by the sender before it is no longer included in the ACK frame. Packets could be received out of order and all subsequent ACK frames containing them could be lost. In this case, the loss recovery algorithm could cause spurious retransmits, but the sender will continue making forward progress.
-To limit ACK Ranges (see Section 19.3.1) to those that have not yet been received by the sender, the receiver SHOULD track which ACK frames have been acknowledged by its peer. The receiver SHOULD exclude already acknowledged packets from future ACK frames whenever these packets would unnecessarily contribute to the ACK frame size. When the receiver is only sending non-ack-eliciting packets, it can bundle a PING or other small ack-eliciting frame with a fraction of them, such as once per round trip, to enable dropping unnecessary ACK ranges and any state for previously sent packets. The receiver MUST NOT bundle an ack-eliciting frame, such as a PING, with all packets that would otherwise be non-ack-eliciting, in order to avoid an infinite feedback loop of acknowledgements.
-To limit receiver state or the size of ACK frames, a receiver MAY limit the number of ACK Ranges it sends. A receiver can do this even without receiving acknowledgment of its ACK frames, with the knowledge this could cause the sender to unnecessarily retransmit some data. Standard QUIC algorithms ([QUIC-RECOVERY]) declare packets lost after sufficiently newer packets are acknowledged. Therefore, the receiver SHOULD repeatedly acknowledge newly received packets in preference to packets received in the past.
-An endpoint measures the delays intentionally introduced between when an ack-eliciting packet is received and the corresponding acknowledgment is sent. The endpoint encodes this delay for the largest acknowledged packet in the Ack Delay field of an ACK frame (see Section 19.3). This allows the receiver of the ACK to adjust for any intentional delays, which is important for getting a better estimate of the path RTT when acknowledgments are delayed. A packet might be held in the OS kernel or elsewhere on the host before being processed. An endpoint MUST NOT include delays that is does not control when populating the Ack Delay field in an ACK frame.
-ACK frames MUST only be carried in a packet that has the same packet number space as the packet being ACKed (see Section 12.1). For instance, packets that are protected with 1-RTT keys MUST be acknowledged in packets that are also protected with 1-RTT keys.
-Packets that a client sends with 0-RTT packet protection MUST be acknowledged by the server in packets protected by 1-RTT keys. This can mean that the client is unable to use these acknowledgments if the server cryptographic handshake messages are delayed or lost. Note that the same limitation applies to other data sent by the server protected by the 1-RTT keys.
-QUIC packets that are determined to be lost are not retransmitted whole. The same applies to the frames that are contained within lost packets. Instead, the information that might be carried in frames is sent again in new frames as needed.
-New frames and packets are used to carry information that is determined to have been lost. In general, information is sent again when a packet containing that information is determined to be lost and sending ceases when a packet containing that information is acknowledged.
- - +The frame types defined in this specification are listed in Table 3. +The Frame Type in ACK, STREAM, MAX_STREAMS, STREAMS_BLOCKED, and +CONNECTION_CLOSE frames is used to carry other frame-specific flags. For all +other frames, the Frame Type field simply identifies the frame. These +frames are explained in more detail in Section 19.¶
+| Type Value | +Frame Type Name | +Definition | +Packets | +
|---|---|---|---|
| 0x00 | +PADDING | ++ Section 19.1 + | +IH01 | +
| 0x01 | +PING | ++ Section 19.2 + | +IH01 | +
| 0x02 - 0x03 | +ACK | ++ Section 19.3 + | +IH_1 | +
| 0x04 | +RESET_STREAM | ++ Section 19.4 + | +__01 | +
| 0x05 | +STOP_SENDING | ++ Section 19.5 + | +__01 | +
| 0x06 | +CRYPTO | ++ Section 19.6 + | +IH_1 | +
| 0x07 | +NEW_TOKEN | ++ Section 19.7 + | +___1 | +
| 0x08 - 0x0f | +STREAM | ++ Section 19.8 + | +__01 | +
| 0x10 | +MAX_DATA | ++ Section 19.9 + | +__01 | +
| 0x11 | +MAX_STREAM_DATA | ++ Section 19.10 + | +__01 | +
| 0x12 - 0x13 | +MAX_STREAMS | ++ Section 19.11 + | +__01 | +
| 0x14 | +DATA_BLOCKED | ++ Section 19.12 + | +__01 | +
| 0x15 | +STREAM_DATA_BLOCKED | ++ Section 19.13 + | +__01 | +
| 0x16 - 0x17 | +STREAMS_BLOCKED | ++ Section 19.14 + | +__01 | +
| 0x18 | +NEW_CONNECTION_ID | ++ Section 19.15 + | +__01 | +
| 0x19 | +RETIRE_CONNECTION_ID | ++ Section 19.16 + | +__01 | +
| 0x1a | +PATH_CHALLENGE | ++ Section 19.17 + | +__01 | +
| 0x1b | +PATH_RESPONSE | ++ Section 19.18 + | +__01 | +
| 0x1c - 0x1d | +CONNECTION_CLOSE | ++ Section 19.19 + | +IH_1* | +
The "Packets" column in Table 3 does not form part of the IANA registry +(see Section 22.3). This column summarizes the types of packets that each +frame type can appear in, indicated as up to four characters indicating:¶
+Section 4 of [QUIC-TLS] provides more detail about these restrictions. Note +that all frames can appear in 1-RTT packets.¶
+An endpoint MUST treat the receipt of a frame of unknown type as a connection +error of type FRAME_ENCODING_ERROR.¶
+All QUIC frames are idempotent in this version of QUIC. That is, a valid +frame does not cause undesirable side effects or errors when received more +than once.¶
+The Frame Type field uses a variable length integer encoding (see +Section 16) with one exception. To ensure simple and efficient +implementations of frame parsing, a frame type MUST use the shortest possible +encoding. Though a two-, four- or eight-byte encoding of the frame types +defined in this document is possible, the Frame Type field for these frames is +encoded on a single byte. For instance, though 0x4001 is a legitimate two-byte +encoding for a variable-length integer with a value of 1, PING frames are always +encoded as a single byte with the value 0x01. An endpoint MAY treat the receipt +of a frame type that uses a longer encoding than necessary as a connection error +of type PROTOCOL_VIOLATION.¶
+A sender bundles one or more frames in a QUIC packet (see Section 12.4).¶
+A sender can minimize per-packet bandwidth and computational costs by bundling +as many frames as possible within a QUIC packet. A sender MAY wait for a short +period of time to bundle multiple frames before sending a packet that is not +maximally packed, to avoid sending out large numbers of small packets. An +implementation MAY use knowledge about application sending behavior or +heuristics to determine whether and for how long to wait. This waiting period +is an implementation decision, and an implementation should be careful to delay +conservatively, since any delay is likely to increase application-visible +latency.¶
+Stream multiplexing is achieved by interleaving STREAM frames from multiple +streams into one or more QUIC packets. A single QUIC packet can include +multiple STREAM frames from one or more streams.¶
+One of the benefits of QUIC is avoidance of head-of-line blocking across +multiple streams. When a packet loss occurs, only streams with data in that +packet are blocked waiting for a retransmission to be received, while other +streams can continue making progress. Note that when data from multiple streams +is bundled into a single QUIC packet, loss of that packet blocks all those +streams from making progress. Implementations are advised to bundle as few +streams as necessary in outgoing packets without losing transmission efficiency +to underfilled packets.¶
+A packet MUST NOT be acknowledged until packet protection has been successfully +removed and all frames contained in the packet have been processed. For STREAM +frames, this means the data has been enqueued in preparation to be received by +the application protocol, but it does not require that data is delivered and +consumed.¶
+Once the packet has been fully processed, a receiver acknowledges receipt by +sending one or more ACK frames containing the packet number of the received +packet.¶
+Endpoints acknowledge all packets they receive and process. However, only +ack-eliciting packets cause an ACK frame to be sent within the maximum ack +delay. Packets that are not ack-eliciting are only acknowledged when an ACK +frame is sent for other reasons.¶
+When sending a packet for any reason, an endpoint should attempt to bundle an +ACK frame if one has not been sent recently. Doing so helps with timely loss +detection at the peer.¶
+In general, frequent feedback from a receiver improves loss and congestion +response, but this has to be balanced against excessive load generated by a +receiver that sends an ACK frame in response to every ack-eliciting packet. The +guidance offered below seeks to strike this balance.¶
+Every packet SHOULD be acknowledged at least once, and ack-eliciting packets
+MUST be acknowledged at least once within the maximum ack delay. An endpoint
+communicates its maximum delay using the max_ack_delay transport parameter;
+see Section 18.2. max_ack_delay declares an explicit
+contract: an endpoint promises to never intentionally delay acknowledgments
+of an ack-eliciting packet by more than the indicated value. If it does,
+any excess accrues to the RTT estimate and could result in spurious or
+delayed retransmissions from the peer. For Initial and Handshake packets,
+a max_ack_delay of 0 is used. The sender uses the receiver's max_ack_delay
+value in determining timeouts for timer-based retransmission, as detailed in
+Section 5.2.1 of [QUIC-RECOVERY].¶
An ACK frame SHOULD be generated for at least every second ack-eliciting packet. +This recommendation is in keeping with standard practice for TCP [RFC5681].¶
+In order to assist loss detection at the sender, an endpoint SHOULD send an ACK +frame immediately on receiving an ack-eliciting packet that is out of order. The +endpoint MAY continue sending ACK frames immediately on each subsequently +received packet, but the endpoint SHOULD return to acknowledging every other +packet within a period of 1/8 x RTT, unless more ack-eliciting packets are +received out of order. If every subsequent ack-eliciting packet arrives out of +order, then an ACK frame SHOULD be sent immediately for every received +ack-eliciting packet.¶
+Similarly, packets marked with the ECN Congestion Experienced (CE) codepoint in +the IP header SHOULD be acknowledged immediately, to reduce the peer's response +time to congestion events.¶
+As an optimization, a receiver MAY process multiple packets before sending any +ACK frames in response. In this case the receiver can determine whether an +immediate or delayed acknowledgement should be generated after processing +incoming packets.¶
+Packets containing PADDING frames are considered to be in flight for congestion +control purposes [QUIC-RECOVERY]. Sending only PADDING frames might cause the +sender to become limited by the congestion controller with no acknowledgments +forthcoming from the receiver. Therefore, a sender SHOULD ensure that other +frames are sent in addition to PADDING frames to elicit acknowledgments from +the receiver.¶
+An endpoint that is only sending ACK frames will not receive acknowledgments +from its peer unless those acknowledgements are included in packets with +ack-eliciting frames. An endpoint SHOULD bundle ACK frames with other frames +when there are new ack-eliciting packets to acknowledge. When only +non-ack-eliciting packets need to be acknowledged, an endpoint MAY wait until an +ack-eliciting packet has been received to bundle an ACK frame with outgoing +frames.¶
+The algorithms in [QUIC-RECOVERY] are resilient to receivers that do not +follow guidance offered above. However, an implementor should only deviate from +these requirements after careful consideration of the performance implications +of doing so.¶
+Packets containing only ACK frames are not congestion controlled, so there are +limits on how frequently they can be sent. An endpoint MUST NOT send more than +one ACK-frame-only packet in response to receiving an ack-eliciting packet. An +endpoint MUST NOT send a non-ack-eliciting packet in response to a +non-ack-eliciting packet, even if there are packet gaps which precede the +received packet. Limiting ACK frames avoids an infinite feedback loop of +acknowledgements, which could prevent the connection from ever becoming idle. +However, the endpoint acknowledges non-ACK-eliciting packets when it sends an +ACK frame.¶
+An endpoint SHOULD treat receipt of an acknowledgment for a packet it did not +send as a connection error of type PROTOCOL_VIOLATION, if it is able to detect +the condition.¶
+When an ACK frame is sent, one or more ranges of acknowledged packets are +included. Including older packets reduces the chance of spurious retransmits +caused by losing previously sent ACK frames, at the cost of larger ACK frames.¶
+ACK frames SHOULD always acknowledge the most recently received packets, and the +more out-of-order the packets are, the more important it is to send an updated +ACK frame quickly, to prevent the peer from declaring a packet as lost and +spuriously retransmitting the frames it contains. An ACK frame is expected +to fit within a single QUIC packet. If it does not, then older ranges +(those with the smallest packet numbers) are omitted.¶
+Section 13.2.3 and Section 13.2.4 describe an exemplary approach for +determining what packets to acknowledge in each ACK frame.¶
+When a packet containing an ACK frame is sent, the largest acknowledged in that +frame may be saved. When a packet containing an ACK frame is acknowledged, the +receiver can stop acknowledging packets less than or equal to the largest +acknowledged in the sent ACK frame.¶
+In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT +of reordering. In cases with ACK frame loss and reordering, this approach does +not guarantee that every acknowledgement is seen by the sender before it is no +longer included in the ACK frame. Packets could be received out of order and +all subsequent ACK frames containing them could be lost. In this case, the +loss recovery algorithm could cause spurious retransmits, but the sender will +continue making forward progress.¶
+To limit ACK Ranges (see Section 19.3.1) to those that have not yet been +received by the sender, the receiver SHOULD track which ACK frames have been +acknowledged by its peer. The receiver SHOULD exclude already acknowledged +packets from future ACK frames whenever these packets would unnecessarily +contribute to the ACK frame size. When the receiver is only sending +non-ack-eliciting packets, it can bundle a PING or other small ack-eliciting +frame with a fraction of them, such as once per round trip, to enable +dropping unnecessary ACK ranges and any state for previously sent packets. +The receiver MUST NOT bundle an ack-eliciting frame, such as a PING, with all +packets that would otherwise be non-ack-eliciting, in order to avoid an +infinite feedback loop of acknowledgements.¶
+To limit receiver state or the size of ACK frames, a receiver MAY limit the +number of ACK Ranges it sends. A receiver can do this even without receiving +acknowledgment of its ACK frames, with the knowledge this could cause the sender +to unnecessarily retransmit some data. Standard QUIC algorithms +([QUIC-RECOVERY]) declare packets lost after sufficiently newer packets are +acknowledged. Therefore, the receiver SHOULD repeatedly acknowledge newly +received packets in preference to packets received in the past.¶
+An endpoint measures the delays intentionally introduced between when an +ack-eliciting packet is received and the corresponding acknowledgment is sent. +The endpoint encodes this delay for the largest acknowledged packet in the Ack +Delay field of an ACK frame (see Section 19.3). This allows the receiver of the +ACK to adjust for any intentional delays, which is important for getting a +better estimate of the path RTT when acknowledgments are delayed. A packet might +be held in the OS kernel or elsewhere on the host before being processed. An +endpoint MUST NOT include delays that is does not control when populating the +Ack Delay field in an ACK frame.¶
+ACK frames MUST only be carried in a packet that has the same packet +number space as the packet being ACKed (see Section 12.1). For +instance, packets that are protected with 1-RTT keys MUST be +acknowledged in packets that are also protected with 1-RTT keys.¶
+Packets that a client sends with 0-RTT packet protection MUST be acknowledged by +the server in packets protected by 1-RTT keys. This can mean that the client is +unable to use these acknowledgments if the server cryptographic handshake +messages are delayed or lost. Note that the same limitation applies to other +data sent by the server protected by the 1-RTT keys.¶
+QUIC packets that are determined to be lost are not retransmitted whole. The +same applies to the frames that are contained within lost packets. Instead, the +information that might be carried in frames is sent again in new frames as +needed.¶
+New frames and packets are used to carry information that is determined to have +been lost. In general, information is sent again when a packet containing that +information is determined to be lost and sending ceases when a packet +containing that information is acknowledged.¶
Endpoints SHOULD prioritize retransmission of data over sending new data, unless priorities specified by the application indicate otherwise (see Section 2.3).
-Even though a sender is encouraged to assemble frames containing up-to-date information every time it sends a packet, it is not forbidden to retransmit copies of frames from lost packets. A receiver MUST accept packets containing an outdated frame, such as a MAX_DATA frame carrying a smaller maximum data than one found in an older packet.
-Upon detecting losses, a sender MUST take appropriate congestion control action. The details of loss detection and congestion control are described in [QUIC-RECOVERY].
-QUIC endpoints can use Explicit Congestion Notification (ECN) [RFC3168] to detect and respond to network congestion. ECN allows a network node to indicate congestion in the network by setting a codepoint in the IP header of a packet instead of dropping it. Endpoints react to congestion by reducing their sending rate in response, as described in [QUIC-RECOVERY].
-To use ECN, QUIC endpoints first determine whether a path supports ECN marking and the peer is able to access the ECN codepoint in the IP header. A network path does not support ECN if ECN marked packets get dropped or ECN markings are rewritten on the path. An endpoint validates the use of ECN on the path, both during connection establishment and when migrating to a new path (Section 9).
-On receiving a QUIC packet with an ECT or CE codepoint, an ECN-enabled endpoint that can access the ECN codepoints from the enclosing IP packet increases the corresponding ECT(0), ECT(1), or CE count, and includes these counts in subsequent ACK frames (see Section 13.2 and Section 19.3). Note that this requires being able to read the ECN codepoints from the enclosing IP packet, which is not possible on all platforms.
-A packet detected by a receiver as a duplicate does not affect the receiver’s local ECN codepoint counts; see (Section 21.8) for relevant security concerns.
-If an endpoint receives a QUIC packet without an ECT or CE codepoint in the IP packet header, it responds per Section 13.2 with an ACK frame without increasing any ECN counts. If an endpoint does not implement ECN support or does not have access to received ECN codepoints, it does not increase ECN counts.
-Coalesced packets (see Section 12.2) mean that several packets can share the same IP header. The ECN counter for the ECN codepoint received in the associated IP header are incremented once for each QUIC packet, not per enclosing IP packet or UDP datagram.
-Each packet number space maintains separate acknowledgement state and separate ECN counts. For example, if one each of an Initial, 0-RTT, Handshake, and 1-RTT QUIC packet are coalesced, the corresponding counts for the Initial and Handshake packet number space will be incremented by one and the counts for the 1-RTT packet number space will be increased by two.
-It is possible for faulty network devices to corrupt or erroneously drop packets with ECN markings. To provide robust connectivity in the presence of such devices, each endpoint independently validates ECN counts and disables ECN if errors are detected.
-Endpoints validate ECN for packets sent on each network path independently. An endpoint thus validates ECN on new connection establishment, when switching to a new server preferred address, and on active connection migration to a new path.
-Even if an endpoint does not use ECN markings on packets it transmits, the endpoint MUST provide feedback about ECN markings received from the peer if they are accessible. Failing to report ECN counts will cause the peer to disable ECN marking.
-To start ECN validation, an endpoint SHOULD do the following when sending packets on a new path to a peer:
- - +Endpoints SHOULD prioritize retransmission of data over sending new data, unless +priorities specified by the application indicate otherwise (see +Section 2.3).¶
+Even though a sender is encouraged to assemble frames containing up-to-date +information every time it sends a packet, it is not forbidden to retransmit +copies of frames from lost packets. A receiver MUST accept packets containing +an outdated frame, such as a MAX_DATA frame carrying a smaller maximum data than +one found in an older packet.¶
+Upon detecting losses, a sender MUST take appropriate congestion control action. +The details of loss detection and congestion control are described in +[QUIC-RECOVERY].¶
+QUIC endpoints can use Explicit Congestion Notification (ECN) [RFC3168] to +detect and respond to network congestion. ECN allows a network node to indicate +congestion in the network by setting a codepoint in the IP header of a packet +instead of dropping it. Endpoints react to congestion by reducing their sending +rate in response, as described in [QUIC-RECOVERY].¶
+To use ECN, QUIC endpoints first determine whether a path supports ECN marking +and the peer is able to access the ECN codepoint in the IP header. A network +path does not support ECN if ECN marked packets get dropped or ECN markings are +rewritten on the path. An endpoint validates the use of ECN on the path, both +during connection establishment and when migrating to a new path +(Section 9).¶
+On receiving a QUIC packet with an ECT or CE codepoint, an ECN-enabled endpoint +that can access the ECN codepoints from the enclosing IP packet increases the +corresponding ECT(0), ECT(1), or CE count, and includes these counts in +subsequent ACK frames (see Section 13.2 and Section 19.3). Note +that this requires being able to read the ECN codepoints from the enclosing IP +packet, which is not possible on all platforms.¶
+A packet detected by a receiver as a duplicate does not affect the receiver's +local ECN codepoint counts; see (Section 21.8) for relevant security +concerns.¶
+If an endpoint receives a QUIC packet without an ECT or CE codepoint in the IP +packet header, it responds per Section 13.2 with an ACK frame without +increasing any ECN counts. If an endpoint does not implement ECN +support or does not have access to received ECN codepoints, it does not increase +ECN counts.¶
+Coalesced packets (see Section 12.2) mean that several packets can share +the same IP header. The ECN counter for the ECN codepoint received in the +associated IP header are incremented once for each QUIC packet, not per +enclosing IP packet or UDP datagram.¶
+Each packet number space maintains separate acknowledgement state and separate +ECN counts. For example, if one each of an Initial, 0-RTT, Handshake, and 1-RTT +QUIC packet are coalesced, the corresponding counts for the Initial and +Handshake packet number space will be incremented by one and the counts for the +1-RTT packet number space will be increased by two.¶
+It is possible for faulty network devices to corrupt or erroneously drop packets +with ECN markings. To provide robust connectivity in the presence of such +devices, each endpoint independently validates ECN counts and disables ECN if +errors are detected.¶
+Endpoints validate ECN for packets sent on each network path independently. An +endpoint thus validates ECN on new connection establishment, when switching to a +new server preferred address, and on active connection migration to a new path.¶
+Even if an endpoint does not use ECN markings on packets it transmits, the +endpoint MUST provide feedback about ECN markings received from the peer if they +are accessible. Failing to report ECN counts will cause the peer to disable ECN +marking.¶
+To start ECN validation, an endpoint SHOULD do the following when sending +packets on a new path to a peer:¶
To reduce the chances of misinterpreting congestive loss as packets dropped by a faulty network element, an endpoint could set the ECT(0) codepoint in the first ten outgoing packets on a path, or for a period of three RTTs, whichever occurs first.
-Implementations MAY experiment with and use other strategies for use of ECN. Other methods of probing paths for ECN support are possible, as are different marking strategies. Implementations can also use the ECT(1) codepoint, as specified in [RFC8311].
-An endpoint that sets ECT(0) or ECT(1) codepoints on packets it transmits MUST use the following steps on receiving an ACK frame to validate ECN.
- - +To reduce the chances of misinterpreting congestive loss as packets dropped by a +faulty network element, an endpoint could set the ECT(0) codepoint in the first +ten outgoing packets on a path, or for a period of three RTTs, whichever occurs +first.¶
+Implementations MAY experiment with and use other strategies for use of ECN. +Other methods of probing paths for ECN support are possible, as are different +marking strategies. Implementations can also use the ECT(1) codepoint, as +specified in [RFC8311].¶
+An endpoint that sets ECT(0) or ECT(1) codepoints on packets it transmits MUST +use the following steps on receiving an ACK frame to validate ECN.¶
Processing ECN counts out of order can result in validation failure. An endpoint SHOULD NOT perform this validation if this ACK frame does not advance the largest packet number acknowledged in this connection.
-An endpoint could miss acknowledgements for a packet when ACK frames are lost. It is therefore possible for the total increase in ECT(0), ECT(1), and CE counts to be greater than the number of packets acknowledged in an ACK frame. When this happens, and if validation succeeds, the local reference counts MUST be increased to match the counts in the ACK frame.
-If validation fails, then the endpoint stops sending ECN markings in subsequent IP packets with the expectation that either the network path or the peer does not support ECN.
-Upon successful validation, an endpoint can continue to set ECT codepoints in subsequent packets with the expectation that the path is ECN-capable. Network routing and path elements can change mid-connection however; an endpoint MUST disable ECN if validation fails at any point in the connection.
-Even if validation fails, an endpoint MAY revalidate ECN on the same path at any later time in the connection.
-The QUIC packet size includes the QUIC header and protected payload, but not the UDP or IP header.
-Clients MUST ensure they send the first Initial packet in a single IP packet. Similarly, the first Initial packet sent after receiving a Retry packet MUST be sent in a single IP packet.
-The payload of a UDP datagram carrying the first Initial packet MUST be expanded to at least 1200 bytes, by adding PADDING frames to the Initial packet and/or by coalescing the Initial packet (see Section 12.2). Sending a UDP datagram of this size ensures that the network path supports a reasonable Maximum Transmission Unit (MTU), and helps reduce the amplitude of amplification attacks caused by server responses toward an unverified client address; see Section 8.
-The datagram containing the first Initial packet from a client MAY exceed 1200 bytes if the client believes that the Path Maximum Transmission Unit (PMTU) supports the size that it chooses.
-A server MAY send a CONNECTION_CLOSE frame with error code PROTOCOL_VIOLATION in response to the first Initial packet it receives from a client if the UDP datagram is smaller than 1200 bytes. It MUST NOT send any other frame type in response, or otherwise behave as if any part of the offending packet was processed as valid.
-The server MUST also limit the number of bytes it sends before validating the address of the client; see Section 8.
-The PMTU is the maximum size of the entire IP packet including the IP header, UDP header, and UDP payload. The UDP payload includes the QUIC packet header, protected payload, and any authentication fields. The PMTU can depend upon the current path characteristics. Therefore, the current largest UDP payload an implementation will send is referred to as the QUIC maximum packet size.
-QUIC depends on a PMTU of at least 1280 bytes. This is the IPv6 minimum size [RFC8200] and is also supported by most modern IPv4 networks. All QUIC packets (except for PMTU probe packets) SHOULD be sized to fit within the maximum packet size to avoid the packet being fragmented or dropped [RFC8085].
-An endpoint SHOULD use Datagram Packetization Layer PMTU Discovery ([DPLPMTUD]) or implement Path MTU Discovery (PMTUD) [RFC1191] [RFC8201] to determine whether the path to a destination will support a desired message size without fragmentation.
-In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets larger than 1280 bytes. Assuming the minimum IP header size, this results in a QUIC maximum packet size of 1232 bytes for IPv6 and 1252 bytes for IPv4. A QUIC implementation MAY be more conservative in computing the QUIC maximum packet size to allow for unknown tunnel overheads or IP header options/extensions.
-Each pair of local and remote addresses could have a different PMTU. QUIC implementations that implement any kind of PMTU discovery therefore SHOULD maintain a maximum packet size for each combination of local and remote IP addresses.
-If a QUIC endpoint determines that the PMTU between any pair of local and remote IP addresses has fallen below the size needed to support the smallest allowed maximum packet size, it MUST immediately cease sending QUIC packets, except for PMTU probe packets, on the affected path. An endpoint MAY terminate the connection if an alternative path cannot be found.
-PMTU discovery [RFC1191] [RFC8201] relies on reception of ICMP messages (e.g., IPv6 Packet Too Big messages) that indicate when a packet is dropped because it is larger than the local router MTU. DPLPMTUD can also optionally use these messages. This use of ICMP messages is potentially vulnerable to off-path attacks that successfully guess the addresses used on the path and reduce the PMTU to a bandwidth-inefficient value.
-An endpoint MUST ignore an ICMP message that claims the PMTU has decreased below 1280 bytes.
-The requirements for generating ICMP ([RFC1812], [RFC4443]) state that the quoted packet should contain as much of the original packet as possible without exceeding the minimum MTU for the IP version. The size of the quoted packet can actually be smaller, or the information unintelligible, as described in Section 1.1 of [DPLPMTUD].
-QUIC endpoints SHOULD validate ICMP messages to protect from off-path injection as specified in [RFC8201] and Section 5.2 of [RFC8085]. This validation SHOULD use the quoted packet supplied in the payload of an ICMP message to associate the message with a corresponding transport connection [DPLPMTUD].
-ICMP message validation MUST include matching IP addresses and UDP ports [RFC8085] and, when possible, connection IDs to an active QUIC session.
-Further validation can also be provided:
- - +Processing ECN counts out of order can result in validation failure. An +endpoint SHOULD NOT perform this validation if this ACK frame does not advance +the largest packet number acknowledged in this connection.¶
+An endpoint could miss acknowledgements for a packet when ACK frames are lost. +It is therefore possible for the total increase in ECT(0), ECT(1), and CE counts +to be greater than the number of packets acknowledged in an ACK frame. When +this happens, and if validation succeeds, the local reference counts MUST be +increased to match the counts in the ACK frame.¶
+If validation fails, then the endpoint stops sending ECN markings in subsequent +IP packets with the expectation that either the network path or the peer does +not support ECN.¶
+Upon successful validation, an endpoint can continue to set ECT codepoints in +subsequent packets with the expectation that the path is ECN-capable. Network +routing and path elements can change mid-connection however; an endpoint MUST +disable ECN if validation fails at any point in the connection.¶
+Even if validation fails, an endpoint MAY revalidate ECN on the same path at any +later time in the connection.¶
+The QUIC packet size includes the QUIC header and protected payload, but not the +UDP or IP header.¶
+A client MUST expand the payload of all UDP datagrams carrying Initial packets +to at least 1200 bytes, by adding PADDING frames to the Initial packet or by +coalescing the Initial packet (see Section 12.2). Sending a UDP datagram +of this size ensures that the network path from the client to the server +supports a reasonable Maximum Transmission Unit (MTU). Padding datagrams also +helps reduce the amplitude of amplification attacks caused by server responses +toward an unverified client address; see Section 8.¶
+Datagrams containing Initial packets MAY exceed 1200 bytes if the client +believes that the Path Maximum Transmission Unit (PMTU) supports the size that +it chooses.¶
+A server MAY send a CONNECTION_CLOSE frame with error code PROTOCOL_VIOLATION in +response to an Initial packet it receives from a client if the UDP datagram is +smaller than 1200 bytes. It MUST NOT send any other frame type in response, or +otherwise behave as if any part of the offending packet was processed as valid.¶
+The server MUST also limit the number of bytes it sends before validating the +address of the client; see Section 8.¶
+The PMTU is the maximum size of the entire IP packet including the IP header, +UDP header, and UDP payload. The UDP payload includes the QUIC packet header, +protected payload, and any authentication fields. The PMTU can depend upon the +current path characteristics. Therefore, the current largest UDP payload an +implementation will send is referred to as the QUIC maximum packet size.¶
+QUIC depends on a PMTU of at least 1280 bytes. This is the IPv6 minimum size +[RFC8200] and is also supported by most modern IPv4 networks. All QUIC +packets (except for PMTU probe packets) SHOULD be sized to fit within the +maximum packet size to avoid the packet being fragmented or dropped +[RFC8085].¶
+An endpoint SHOULD use Datagram Packetization Layer PMTU Discovery +([DPLPMTUD]) or implement Path MTU Discovery +(PMTUD) [RFC1191] [RFC8201] to determine whether the path to a destination +will support a desired message size without fragmentation.¶
+In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets +larger than 1280 bytes. Assuming the minimum IP header size, this results in a +QUIC maximum packet size of 1232 bytes for IPv6 and 1252 bytes for IPv4. A QUIC +implementation MAY be more conservative in computing the QUIC maximum packet +size to allow for unknown tunnel overheads or IP header options/extensions.¶
+Each pair of local and remote addresses could have a different PMTU. QUIC +implementations that implement any kind of PMTU discovery therefore SHOULD +maintain a maximum packet size for each combination of local and remote IP +addresses.¶
+If a QUIC endpoint determines that the PMTU between any pair of local and remote +IP addresses has fallen below the size needed to support the smallest allowed +maximum packet size, it MUST immediately cease sending QUIC packets, except for +PMTU probe packets, on the affected path. An endpoint MAY terminate the +connection if an alternative path cannot be found.¶
+PMTU discovery [RFC1191] [RFC8201] relies on reception of ICMP messages +(e.g., IPv6 Packet Too Big messages) that indicate when a packet is dropped +because it is larger than the local router MTU. DPLPMTUD can also optionally use +these messages. This use of ICMP messages is potentially vulnerable to off-path +attacks that successfully guess the addresses used on the path and reduce the +PMTU to a bandwidth-inefficient value.¶
+An endpoint MUST ignore an ICMP message that claims the PMTU has decreased below +1280 bytes.¶
+The requirements for generating ICMP ([RFC1812], [RFC4443]) state that the +quoted packet should contain as much of the original packet as possible without +exceeding the minimum MTU for the IP version. The size of the quoted packet can +actually be smaller, or the information unintelligible, as described in Section +1.1 of [DPLPMTUD].¶
+QUIC endpoints SHOULD validate ICMP messages to protect from off-path injection +as specified in [RFC8201] and Section 5.2 of [RFC8085]. This validation +SHOULD use the quoted packet supplied in the payload of an ICMP message to +associate the message with a corresponding transport connection [DPLPMTUD].¶
+ICMP message validation MUST include matching IP addresses and UDP ports +[RFC8085] and, when possible, connection IDs to an active QUIC session.¶
+Further validation can also be provided:¶
The endpoint SHOULD ignore all ICMP messages that fail validation.
-An endpoint MUST NOT increase PMTU based on ICMP messages. Any reduction in the QUIC maximum packet size MAY be provisional until QUIC’s loss detection algorithm determines that the quoted packet has actually been lost.
-Section 6.4 of [DPLPMTUD] provides considerations for implementing Datagram Packetization Layer PMTUD (DPLPMTUD) with QUIC.
-When implementing the algorithm in Section 5.3 of [DPLPMTUD], the initial value of BASE_PMTU SHOULD be consistent with the minimum QUIC packet size (1232 bytes for IPv6 and 1252 bytes for IPv4).
-PING and PADDING frames can be used to generate PMTU probe packets. These frames might not be retransmitted if a probe packet containing them is lost. However, these frames do consume congestion window, which could delay the transmission of subsequent application data.
-A PING frame can be included in a PMTU probe to ensure that a valid probe is acknowledged.
-The considerations for processing ICMP messages in the previous section also apply if these messages are used by DPLPMTUD.
-Endpoints that rely on the destination connection ID for routing QUIC packets are likely to require that the connection ID be included in PMTU probe packets to route any resulting ICMP messages (Section 14.2) back to the correct endpoint. However, only long header packets (Section 17.2) contain source connection IDs, and long header packets are not decrypted or acknowledged by the peer once the handshake is complete. One way to construct a PMTU probe is to coalesce (see Section 12.2) a Handshake packet (Section 17.2.4) with a short header packet in a single UDP datagram. If the UDP datagram reaches the endpoint, the Handshake packet will be ignored, but the short header packet will be acknowledged. If the UDP datagram elicits an ICMP message, that message will likely contain the source connection ID within the quoted portion of the UDP datagram.
-QUIC versions are identified using a 32-bit unsigned number.
-The version 0x00000000 is reserved to represent version negotiation. This version of the specification is identified by the number 0x00000001.
-Other versions of QUIC might have different properties to this version. The properties of QUIC that are guaranteed to be consistent across all versions of the protocol are described in [QUIC-INVARIANTS].
-Version 0x00000001 of QUIC uses TLS as a cryptographic handshake protocol, as described in [QUIC-TLS].
-Versions with the most significant 16 bits of the version number cleared are reserved for use in future IETF consensus documents.
-Versions that follow the pattern 0x?a?a?a?a are reserved for use in forcing version negotiation to be exercised. That is, any version number where the low four bits of all bytes is 1010 (in binary). A client or server MAY advertise support for any of these reserved versions.
-Reserved version numbers will never represent a real protocol; a client MAY use one of these version numbers with the expectation that the server will initiate version negotiation; a server MAY advertise support for one of these versions and can expect that clients ignore the value.
-[[RFC editor: please remove the remainder of this section before publication.]]
-The version number for the final version of this specification (0x00000001), is reserved for the version of the protocol that is published as an RFC.
-Version numbers used to identify IETF drafts are created by adding the draft number to 0xff000000. For example, draft-ietf-quic-transport-13 would be identified as 0xff00000D.
-Implementors are encouraged to register version numbers of QUIC that they are using for private experimentation on the GitHub wiki at <https://github.com/quicwg/base-drafts/wiki/QUIC-Versions>.
-QUIC packets and frames commonly use a variable-length encoding for non-negative integer values. This encoding ensures that smaller integer values need fewer bytes to encode.
-The QUIC variable-length integer encoding reserves the two most significant bits of the first byte to encode the base 2 logarithm of the integer encoding length in bytes. The integer value is encoded on the remaining bits, in network byte order.
-This means that integers are encoded on 1, 2, 4, or 8 bytes and can encode 6, 14, 30, or 62 bit values respectively. Table 4 summarizes the encoding properties.
- - -| 2Bit | -Length | -Usable Bits | -Range | -
|---|---|---|---|
| 00 | -1 | -6 | -0-63 | -
| 01 | -2 | -14 | -0-16383 | -
| 10 | -4 | -30 | -0-1073741823 | -
| 11 | -8 | -62 | -0-4611686018427387903 | -
For example, the eight byte sequence c2 19 7c 5e ff 14 e8 8c (in hexadecimal) decodes to the decimal value 151288809941952652; the four byte sequence 9d 7f 3e 7d decodes to 494878333; the two byte sequence 7b bd decodes to 15293; and the single byte 25 decodes to 37 (as does the two byte sequence 40 25).
-Error codes (Section 20) and versions (Section 15) are described using integers, but do not use this encoding.
-All numeric values are encoded in network byte order (that is, big-endian) and all field sizes are in bits. Hexadecimal notation is used for describing the value of fields.
-Packet numbers are integers in the range 0 to 2^62-1 (Section 12.3). When present in long or short packet headers, they are encoded in 1 to 4 bytes. The number of bits required to represent the packet number is reduced by including the least significant bits of the packet number.
-The encoded packet number is protected as described in Section 5.4 of [QUIC-TLS].
-The sender MUST use a packet number size able to represent more than twice as large a range than the difference between the largest acknowledged packet and packet number being sent. A peer receiving the packet will then correctly decode the packet number, unless the packet is delayed in transit such that it arrives after many higher-numbered packets have been received. An endpoint SHOULD use a large enough packet number encoding to allow the packet number to be recovered even if the packet arrives after packets that are sent afterwards.
-As a result, the size of the packet number encoding is at least one bit more than the base-2 logarithm of the number of contiguous unacknowledged packet numbers, including the new packet.
-For example, if an endpoint has received an acknowledgment for packet 0xabe8bc, sending a packet with a number of 0xac5c02 requires a packet number encoding with 16 bits or more; whereas the 24-bit packet number encoding is needed to send a packet with a number of 0xace8fe.
-At a receiver, protection of the packet number is removed prior to recovering the full packet number. The full packet number is then reconstructed based on the number of significant bits present, the value of those bits, and the largest packet number received on a successfully authenticated packet. Recovering the full packet number is necessary to successfully remove packet protection.
-Once header protection is removed, the packet number is decoded by finding the packet number value that is closest to the next expected packet. The next expected packet is the highest received packet number plus one. For example, if the highest successfully authenticated packet had a packet number of 0xa82f30ea, then a packet containing a 16-bit value of 0x9b32 will be decoded as 0xa82f9b32. Example pseudo-code for packet number decoding can be found in Appendix A.
-The endpoint SHOULD ignore all ICMP messages that fail validation.¶
+An endpoint MUST NOT increase PMTU based on ICMP messages. Any reduction in the +QUIC maximum packet size MAY be provisional until QUIC's loss detection +algorithm determines that the quoted packet has actually been lost.¶
+Section 6.4 of [DPLPMTUD] provides considerations for implementing Datagram +Packetization Layer PMTUD (DPLPMTUD) with QUIC.¶
+When implementing the algorithm in Section 5.3 of [DPLPMTUD], the initial +value of BASE_PMTU SHOULD be consistent with the minimum QUIC packet size (1232 +bytes for IPv6 and 1252 bytes for IPv4).¶
+PING and PADDING frames can be used to generate PMTU probe packets. These frames +might not be retransmitted if a probe packet containing them is lost. However, +these frames do consume congestion window, which could delay the transmission of +subsequent application data.¶
+A PING frame can be included in a PMTU probe to ensure that a valid probe is +acknowledged.¶
+The considerations for processing ICMP messages in the previous section also +apply if these messages are used by DPLPMTUD.¶
+Endpoints that rely on the destination connection ID for routing QUIC packets +are likely to require that the connection ID be included in PMTU probe packets +to route any resulting ICMP messages (Section 14.2) back to the correct +endpoint. However, only long header packets (Section 17.2) contain source +connection IDs, and long header packets are not decrypted or acknowledged by +the peer once the handshake is complete. One way to construct a PMTU probe is +to coalesce (see Section 12.2) a Handshake packet (Section 17.2.4) +with a short header packet in a single UDP datagram. If the UDP datagram +reaches the endpoint, the Handshake packet will be ignored, but the short header +packet will be acknowledged. If the UDP datagram elicits an ICMP message, that +message will likely contain the source connection ID within the quoted portion +of the UDP datagram.¶
+QUIC versions are identified using a 32-bit unsigned number.¶
+The version 0x00000000 is reserved to represent version negotiation. This +version of the specification is identified by the number 0x00000001.¶
+Other versions of QUIC might have different properties to this version. The +properties of QUIC that are guaranteed to be consistent across all versions of +the protocol are described in [QUIC-INVARIANTS].¶
+Version 0x00000001 of QUIC uses TLS as a cryptographic handshake protocol, as +described in [QUIC-TLS].¶
+Versions with the most significant 16 bits of the version number cleared are +reserved for use in future IETF consensus documents.¶
+Versions that follow the pattern 0x?a?a?a?a are reserved for use in forcing +version negotiation to be exercised. That is, any version number where the low +four bits of all bytes is 1010 (in binary). A client or server MAY advertise +support for any of these reserved versions.¶
+Reserved version numbers will never represent a real protocol; a client MAY use +one of these version numbers with the expectation that the server will initiate +version negotiation; a server MAY advertise support for one of these versions +and can expect that clients ignore the value.¶
+[[RFC editor: please remove the remainder of this section before +publication.]]¶
+The version number for the final version of this specification (0x00000001), is +reserved for the version of the protocol that is published as an RFC.¶
+Version numbers used to identify IETF drafts are created by adding the draft +number to 0xff000000. For example, draft-ietf-quic-transport-13 would be +identified as 0xff00000D.¶
+Implementors are encouraged to register version numbers of QUIC that they are +using for private experimentation on the GitHub wiki at +<https://github.com/quicwg/base-drafts/wiki/QUIC-Versions>.¶
+QUIC packets and frames commonly use a variable-length encoding for non-negative +integer values. This encoding ensures that smaller integer values need fewer +bytes to encode.¶
+The QUIC variable-length integer encoding reserves the two most significant bits +of the first byte to encode the base 2 logarithm of the integer encoding length +in bytes. The integer value is encoded on the remaining bits, in network byte +order.¶
+This means that integers are encoded on 1, 2, 4, or 8 bytes and can encode 6, +14, 30, or 62 bit values respectively. Table 4 summarizes the +encoding properties.¶
+| 2Bit | +Length | +Usable Bits | +Range | +
|---|---|---|---|
| 00 | +1 | +6 | +0-63 | +
| 01 | +2 | +14 | +0-16383 | +
| 10 | +4 | +30 | +0-1073741823 | +
| 11 | +8 | +62 | +0-4611686018427387903 | +
For example, the eight byte sequence c2 19 7c 5e ff 14 e8 8c (in hexadecimal) +decodes to the decimal value 151288809941952652; the four byte sequence 9d 7f 3e +7d decodes to 494878333; the two byte sequence 7b bd decodes to 15293; and the +single byte 25 decodes to 37 (as does the two byte sequence 40 25).¶
+Error codes (Section 20) and versions (Section 15) are described using +integers, but do not use this encoding.¶
+All numeric values are encoded in network byte order (that is, big-endian) and +all field sizes are in bits. Hexadecimal notation is used for describing the +value of fields.¶
+Packet numbers are integers in the range 0 to 2^62-1 (Section 12.3). When +present in long or short packet headers, they are encoded in 1 to 4 bytes. The +number of bits required to represent the packet number is reduced by including +the least significant bits of the packet number.¶
+The encoded packet number is protected as described in Section 5.4 of +[QUIC-TLS].¶
+The sender MUST use a packet number size able to represent more than twice as +large a range than the difference between the largest acknowledged packet and +packet number being sent. A peer receiving the packet will then correctly +decode the packet number, unless the packet is delayed in transit such that it +arrives after many higher-numbered packets have been received. An endpoint +SHOULD use a large enough packet number encoding to allow the packet number to +be recovered even if the packet arrives after packets that are sent afterwards.¶
+As a result, the size of the packet number encoding is at least one bit more +than the base-2 logarithm of the number of contiguous unacknowledged packet +numbers, including the new packet.¶
+For example, if an endpoint has received an acknowledgment for packet 0xabe8bc, +sending a packet with a number of 0xac5c02 requires a packet number encoding +with 16 bits or more; whereas the 24-bit packet number encoding is needed to +send a packet with a number of 0xace8fe.¶
+At a receiver, protection of the packet number is removed prior to recovering +the full packet number. The full packet number is then reconstructed based on +the number of significant bits present, the value of those bits, and the largest +packet number received on a successfully authenticated packet. Recovering the +full packet number is necessary to successfully remove packet protection.¶
+Once header protection is removed, the packet number is decoded by finding the +packet number value that is closest to the next expected packet. The next +expected packet is the highest received packet number plus one. For example, if +the highest successfully authenticated packet had a packet number of 0xa82f30ea, +then a packet containing a 16-bit value of 0x9b32 will be decoded as 0xa82f9b32. +Example pseudo-code for packet number decoding can be found in +Appendix A.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2429,85 +5618,181 @@-| Source Connection ID (0..160) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: Long Header Packet Format
-Long headers are used for packets that are sent prior to the establishment of 1-RTT keys. Once both conditions are met, a sender switches to sending packets using the short header (Section 17.3). The long form allows for special packets - such as the Version Negotiation packet - to be represented in this uniform fixed-length packet format. Packets that use the long header contain the following fields:
- - -Long headers are used for packets that are sent prior to the establishment +of 1-RTT keys. Once both conditions are +met, a sender switches to sending packets using the short header +(Section 17.3). The long form allows for special packets - such as the +Version Negotiation packet - to be represented in this uniform fixed-length +packet format. Packets that use the long header contain the following fields:¶
+In this version of QUIC, the following packet types with the long header are defined:
- - -| Type | -Name | -Section | -
|---|---|---|
| 0x0 | -Initial | -Section 17.2.2 | -
| 0x1 | -0-RTT | -Section 17.2.3 | -
| 0x2 | -Handshake | -Section 17.2.4 | -
| 0x3 | -Retry | -Section 17.2.5 | -
The header form bit, connection ID lengths byte, Destination and Source Connection ID fields, and Version fields of a long header packet are version-independent. The other fields in the first byte are version-specific. See [QUIC-INVARIANTS] for details on how packets from different versions of QUIC are interpreted.
-The interpretation of the fields and the payload are specific to a version and packet type. While type-specific semantics for this version are described in the following sections, several long-header packets in this version of QUIC contain these additional fields:
- - -In this version of QUIC, the following packet types with the long header are +defined:¶
+| Type | +Name | +Section | +
|---|---|---|
| 0x0 | +Initial | ++ Section 17.2.2 + | +
| 0x1 | +0-RTT | ++ Section 17.2.3 + | +
| 0x2 | +Handshake | ++ Section 17.2.4 + | +
| 0x3 | +Retry | ++ Section 17.2.5 + | +
The header form bit, connection ID lengths byte, Destination and Source +Connection ID fields, and Version fields of a long header packet are +version-independent. The other fields in the first byte are version-specific. +See [QUIC-INVARIANTS] for details on how packets from different versions of +QUIC are interpreted.¶
+The interpretation of the fields and the payload are specific to a version and +packet type. While type-specific semantics for this version are described in +the following sections, several long-header packets in this version of QUIC +contain these additional fields:¶
+A Version Negotiation packet is inherently not version-specific. Upon receipt by a client, it will be identified as a Version Negotiation packet based on the Version field having a value of 0.
-The Version Negotiation packet is a response to a client packet that contains a version that is not supported by the server, and is only sent by servers.
-The layout of a Version Negotiation packet is:
- - +A Version Negotiation packet is inherently not version-specific. Upon receipt by +a client, it will be identified as a Version Negotiation packet based on the +Version field having a value of 0.¶
+The Version Negotiation packet is a response to a client packet that contains a +version that is not supported by the server, and is only sent by servers.¶
+The layout of a Version Negotiation packet is:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2533,22 +5818,51 @@-| [Supported Version N (32)] ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 10: Version Negotiation Packet
-The value in the Unused field is selected randomly by the server. Clients MUST ignore the value of this field. Servers SHOULD set the most significant bit of this field (0x40) to 1 so that Version Negotiation packets appear to have the Fixed Bit field.
-The Version field of a Version Negotiation packet MUST be set to 0x00000000.
-The server MUST include the value from the Source Connection ID field of the packet it receives in the Destination Connection ID field. The value for Source Connection ID MUST be copied from the Destination Connection ID of the received packet, which is initially randomly selected by a client. Echoing both connection IDs gives clients some assurance that the server received the packet and that the Version Negotiation packet was not generated by an off-path attacker.
-As future versions of QUIC may support Connection IDs larger than the version 1 limit, Version Negotiation packets could carry Connection IDs that are longer than 20 bytes.
-The remainder of the Version Negotiation packet is a list of 32-bit versions which the server supports.
-A Version Negotiation packet cannot be explicitly acknowledged in an ACK frame by a client. Receiving another Initial packet implicitly acknowledges a Version Negotiation packet.
-The Version Negotiation packet does not include the Packet Number and Length fields present in other packets that use the long header form. Consequently, a Version Negotiation packet consumes an entire UDP datagram.
-A server MUST NOT send more than one Version Negotiation packet in response to a single UDP datagram.
-See Section 6 for a description of the version negotiation process.
-An Initial packet uses long headers with a type value of 0x0. It carries the first CRYPTO frames sent by the client and server to perform key exchange, and carries ACKs in either direction.
- - +The value in the Unused field is selected randomly by the server. Clients MUST +ignore the value of this field. Servers SHOULD set the most significant bit of +this field (0x40) to 1 so that Version Negotiation packets appear to have the +Fixed Bit field.¶
+The Version field of a Version Negotiation packet MUST be set to 0x00000000.¶
+The server MUST include the value from the Source Connection ID field of the +packet it receives in the Destination Connection ID field. The value for Source +Connection ID MUST be copied from the Destination Connection ID of the received +packet, which is initially randomly selected by a client. Echoing both +connection IDs gives clients some assurance that the server received the packet +and that the Version Negotiation packet was not generated by an off-path +attacker.¶
+As future versions of QUIC may support Connection IDs larger than the version 1 +limit, Version Negotiation packets could carry Connection IDs that are longer +than 20 bytes.¶
+The remainder of the Version Negotiation packet is a list of 32-bit versions +which the server supports.¶
+A Version Negotiation packet cannot be explicitly acknowledged in an ACK frame +by a client. Receiving another Initial packet implicitly acknowledges a Version +Negotiation packet.¶
+The Version Negotiation packet does not include the Packet Number and Length +fields present in other packets that use the long header form. Consequently, +a Version Negotiation packet consumes an entire UDP datagram.¶
+A server MUST NOT send more than one Version Negotiation packet in response to a +single UDP datagram.¶
+See Section 6 for a description of the version negotiation +process.¶
+An Initial packet uses long headers with a type value of 0x0. It carries the +first CRYPTO frames sent by the client and server to perform key exchange, and +carries ACKs in either direction.¶
++-+-+-+-+-+-+-+-+ |1|1| 0 |R R|P P| @@ -2574,34 +5888,91 @@-| Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 11: Initial Packet
-The Initial packet contains a long header as well as the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. Between the SCID and Length fields, there are two additional field specific to the Initial packet.
- - -The Initial packet contains a long header as well as the Length and Packet +Number fields. The first byte contains the Reserved and Packet Number Length +bits. Between the SCID and Length fields, there are two additional +field specific to the Initial packet.¶
+In order to prevent tampering by version-unaware middleboxes, Initial packets are protected with connection- and version-specific keys (Initial keys) as described in [QUIC-TLS]. This protection does not provide confidentiality or integrity against on-path attackers, but provides some level of protection against off-path attackers.
-The client and server use the Initial packet type for any packet that contains an initial cryptographic handshake message. This includes all cases where a new packet containing the initial cryptographic message needs to be created, such as the packets sent after receiving a Retry packet (Section 17.2.5).
-A server sends its first Initial packet in response to a client Initial. A server may send multiple Initial packets. The cryptographic key exchange could require multiple round trips or retransmissions of this data.
-The payload of an Initial packet includes a CRYPTO frame (or frames) containing a cryptographic handshake message, ACK frames, or both. PING, PADDING, and CONNECTION_CLOSE frames are also permitted. An endpoint that receives an Initial packet containing other frames can either discard the packet as spurious or treat it as a connection error.
-The first packet sent by a client always includes a CRYPTO frame that contains the start or all of the first cryptographic handshake message. The first CRYPTO frame sent always begins at an offset of 0 (see Section 7).
-Note that if the server sends a HelloRetryRequest, the client will send another series of Initial packets. These Initial packets will continue the cryptographic handshake and will contain CRYPTO frames starting at an offset matching the size of the CRYPTO frames sent in the first flight of Initial packets.
-A client stops both sending and processing Initial packets when it sends its first Handshake packet. A server stops sending and processing Initial packets when it receives its first Handshake packet. Though packets might still be in flight or awaiting acknowledgment, no further Initial packets need to be exchanged beyond this point. Initial packet protection keys are discarded (see Section 4.9.1 of [QUIC-TLS]) along with any loss recovery and congestion control state (see Section 6.5 of [QUIC-RECOVERY]).
-Any data in CRYPTO frames is discarded - and no longer retransmitted - when Initial keys are discarded.
-A 0-RTT packet uses long headers with a type value of 0x1, followed by the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. It is used to carry “early” data from the client to the server as part of the first flight, prior to handshake completion. As part of the TLS handshake, the server can accept or reject this early data.
-See Section 2.3 of [TLS13] for a discussion of 0-RTT data and its limitations.
+In order to prevent tampering by version-unaware middleboxes, Initial packets +are protected with connection- and version-specific keys (Initial keys) as +described in [QUIC-TLS]. This protection does not provide confidentiality or +integrity against on-path attackers, but provides some level of protection +against off-path attackers.¶
+The client and server use the Initial packet type for any packet that contains +an initial cryptographic handshake message. This includes all cases where a new +packet containing the initial cryptographic message needs to be created, such as +the packets sent after receiving a Retry packet (Section 17.2.5).¶
+A server sends its first Initial packet in response to a client Initial. A +server may send multiple Initial packets. The cryptographic key exchange could +require multiple round trips or retransmissions of this data.¶
+The payload of an Initial packet includes a CRYPTO frame (or frames) containing +a cryptographic handshake message, ACK frames, or both. PING, PADDING, and +CONNECTION_CLOSE frames are also permitted. An endpoint that receives an +Initial packet containing other frames can either discard the packet as spurious +or treat it as a connection error.¶
+The first packet sent by a client always includes a CRYPTO frame that contains +the start or all of the first cryptographic handshake message. The first +CRYPTO frame sent always begins at an offset of 0 (see Section 7).¶
+Note that if the server sends a HelloRetryRequest, the client will send another +series of Initial packets. These Initial packets will continue the +cryptographic handshake and will contain CRYPTO frames starting at an offset +matching the size of the CRYPTO frames sent in the first flight of Initial +packets.¶
+A client stops both sending and processing Initial packets when it sends its +first Handshake packet. A server stops sending and processing Initial packets +when it receives its first Handshake packet. Though packets might still be in +flight or awaiting acknowledgment, no further Initial packets need to be +exchanged beyond this point. Initial packet protection keys are discarded (see +Section 4.9.1 of [QUIC-TLS]) along with any loss recovery and congestion +control state (see Section 6.5 of [QUIC-RECOVERY]).¶
+Any data in CRYPTO frames is discarded - and no longer retransmitted - when +Initial keys are discarded.¶
+A 0-RTT packet uses long headers with a type value of 0x1, followed by the +Length and Packet Number fields. The first byte contains the Reserved and Packet +Number Length bits. It is used to carry "early" data from the client to the +server as part of the first flight, prior to handshake completion. As part of +the TLS handshake, the server can accept or reject this early data.¶
+See Section 2.3 of [TLS13] for a discussion of 0-RTT data and its +limitations.¶
++-+-+-+-+-+-+-+-+ |1|1| 1 |R R|P P| @@ -2623,19 +5994,48 @@-| Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
0-RTT Packet
-Packet numbers for 0-RTT protected packets use the same space as 1-RTT protected packets.
-After a client receives a Retry packet, 0-RTT packets are likely to have been lost or discarded by the server. A client SHOULD attempt to resend data in 0-RTT packets after it sends a new Initial packet.
-A client MUST NOT reset the packet number it uses for 0-RTT packets, since the keys used to protect 0-RTT packets will not change as a result of responding to a Retry packet. Sending packets with the same packet number in that case is likely to compromise the packet protection for all 0-RTT packets because the same key and nonce could be used to protect different content.
-A client only receives acknowledgments for its 0-RTT packets once the handshake is complete. Consequently, a server might expect 0-RTT packets to start with a packet number of 0. Therefore, in determining the length of the packet number encoding for 0-RTT packets, a client MUST assume that all packets up to the current packet number are in flight, starting from a packet number of 0. Thus, 0-RTT packets could need to use a longer packet number encoding.
-A client MUST NOT send 0-RTT packets once it starts processing 1-RTT packets from the server. This means that 0-RTT packets cannot contain any response to frames from 1-RTT packets. For instance, a client cannot send an ACK frame in a 0-RTT packet, because that can only acknowledge a 1-RTT packet. An acknowledgment for a 1-RTT packet MUST be carried in a 1-RTT packet.
-A server SHOULD treat a violation of remembered limits as a connection error of an appropriate type (for instance, a FLOW_CONTROL_ERROR for exceeding stream data limits).
-A Handshake packet uses long headers with a type value of 0x2, followed by the Length and Packet Number fields. The first byte contains the Reserved and Packet Number Length bits. It is used to carry acknowledgments and cryptographic handshake messages from the server and client.
- - +Packet numbers for 0-RTT protected packets use the same space as 1-RTT protected +packets.¶
+After a client receives a Retry packet, 0-RTT packets are likely to have been +lost or discarded by the server. A client SHOULD attempt to resend data in +0-RTT packets after it sends a new Initial packet.¶
+A client MUST NOT reset the packet number it uses for 0-RTT packets, since the +keys used to protect 0-RTT packets will not change as a result of responding to +a Retry packet. Sending packets with the same packet number in that case is +likely to compromise the packet protection for all 0-RTT packets because the +same key and nonce could be used to protect different content.¶
+A client only receives acknowledgments for its 0-RTT packets once the handshake +is complete. Consequently, a server might expect 0-RTT packets to start with a +packet number of 0. Therefore, in determining the length of the packet number +encoding for 0-RTT packets, a client MUST assume that all packets up to the +current packet number are in flight, starting from a packet number of 0. Thus, +0-RTT packets could need to use a longer packet number encoding.¶
+A client MUST NOT send 0-RTT packets once it starts processing 1-RTT packets +from the server. This means that 0-RTT packets cannot contain any response to +frames from 1-RTT packets. For instance, a client cannot send an ACK frame in a +0-RTT packet, because that can only acknowledge a 1-RTT packet. An +acknowledgment for a 1-RTT packet MUST be carried in a 1-RTT packet.¶
+A server SHOULD treat a violation of remembered limits as a connection error of +an appropriate type (for instance, a FLOW_CONTROL_ERROR for exceeding stream +data limits).¶
+A Handshake packet uses long headers with a type value of 0x2, followed by the +Length and Packet Number fields. The first byte contains the Reserved and +Packet Number Length bits. It is used to carry acknowledgments and +cryptographic handshake messages from the server and client.¶
++-+-+-+-+-+-+-+-+ |1|1| 2 |R R|P P| @@ -2657,18 +6057,40 @@-| Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 12: Handshake Protected Packet
-Once a client has received a Handshake packet from a server, it uses Handshake packets to send subsequent cryptographic handshake messages and acknowledgments to the server.
-The Destination Connection ID field in a Handshake packet contains a connection ID that is chosen by the recipient of the packet; the Source Connection ID includes the connection ID that the sender of the packet wishes to use (see Section 7.2).
-Handshake packets are their own packet number space, and thus the first Handshake packet sent by a server contains a packet number of 0.
-The payload of this packet contains CRYPTO frames and could contain PING, PADDING, or ACK frames. Handshake packets MAY contain CONNECTION_CLOSE frames. Endpoints MUST treat receipt of Handshake packets with other frames as a connection error.
-Like Initial packets (see Section 17.2.2.1), data in CRYPTO frames at the Handshake encryption level is discarded - and no longer retransmitted - when Handshake protection keys are discarded.
-A Retry packet uses a long packet header with a type value of 0x3. It carries an address validation token created by the server. It is used by a server that wishes to perform a retry (see Section 8.1).
- - +Once a client has received a Handshake packet from a server, it uses Handshake +packets to send subsequent cryptographic handshake messages and acknowledgments +to the server.¶
+The Destination Connection ID field in a Handshake packet contains a connection +ID that is chosen by the recipient of the packet; the Source Connection ID +includes the connection ID that the sender of the packet wishes to use (see +Section 7.2).¶
+Handshake packets are their own packet number space, and thus the first +Handshake packet sent by a server contains a packet number of 0.¶
+The payload of this packet contains CRYPTO frames and could contain PING, +PADDING, or ACK frames. Handshake packets MAY contain CONNECTION_CLOSE frames. +Endpoints MUST treat receipt of Handshake packets with other frames as a +connection error.¶
+Like Initial packets (see Section 17.2.2.1), data in CRYPTO frames at the +Handshake encryption level is discarded - and no longer retransmitted - when +Handshake protection keys are discarded.¶
+A Retry packet uses a long packet header with a type value of 0x3. It carries +an address validation token created by the server. It is used by a server that +wishes to perform a retry (see Section 8.1).¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2692,37 +6114,103 @@-| Retry Token (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 13: Retry Packet
-A Retry packet (shown in Figure 13) does not contain any protected fields. The value in the Unused field is selected randomly by the server. In addition to the long header, it contains these additional fields:
- - -A Retry packet (shown in Figure 14) does not contain any protected +fields. The value in the Unused field is selected randomly by the server. In +addition to the long header, it contains these additional fields:¶
+The server populates the Destination Connection ID with the connection ID that the client included in the Source Connection ID of the Initial packet.
-The server includes a connection ID of its choice in the Source Connection ID field. This value MUST not be equal to the Destination Connection ID field of the packet sent by the client. A client MUST discard a Retry packet that contains a Source Connection ID field that is identical to the Destination Connection ID field of its Initial packet. The client MUST use the value from the Source Connection ID field of the Retry packet in the Destination Connection ID field of subsequent packets that it sends.
-A server MAY send Retry packets in response to Initial and 0-RTT packets. A server can either discard or buffer 0-RTT packets that it receives. A server can send multiple Retry packets as it receives Initial or 0-RTT packets. A server MUST NOT send more than one Retry packet in response to a single UDP datagram.
-A client MUST accept and process at most one Retry packet for each connection attempt. After the client has received and processed an Initial or Retry packet from the server, it MUST discard any subsequent Retry packets that it receives.
-Clients MUST discard Retry packets that contain an Original Destination Connection ID field that does not match the Destination Connection ID from its Initial packet. This prevents an off-path attacker from injecting a Retry packet.
-The client responds to a Retry packet with an Initial packet that includes the provided Retry Token to continue connection establishment.
-A client sets the Destination Connection ID field of this Initial packet to the value from the Source Connection ID in the Retry packet. Changing Destination Connection ID also results in a change to the keys used to protect the Initial packet. It also sets the Token field to the token provided in the Retry. The client MUST NOT change the Source Connection ID because the server could include the connection ID as part of its token validation logic (see Section 8.1.3).
-The next Initial packet from the client uses the connection ID and token values from the Retry packet (see Section 7.2). Aside from this, the Initial packet sent by the client is subject to the same restrictions as the first Initial packet. A client MUST use the same cryptographic handshake message it includes in this packet. A server MAY treat a packet that contains a different cryptographic handshake message as a connection error or discard it.
-A client MAY attempt 0-RTT after receiving a Retry packet by sending 0-RTT packets to the connection ID provided by the server. A client MUST NOT change the cryptographic handshake message it sends in response to receiving a Retry.
-A client MUST NOT reset the packet number for any packet number space after processing a Retry packet; Section 17.2.3 contains more information on this.
-A server acknowledges the use of a Retry packet for a connection using the original_connection_id transport parameter (see Section 18.2). If the server sends a Retry packet, it MUST include the value of the Original Destination Connection ID field of the Retry packet (that is, the Destination Connection ID field from the client’s first Initial packet) in the transport parameter.
-If the client received and processed a Retry packet, it MUST validate that the original_connection_id transport parameter is present and correct; otherwise, it MUST validate that the transport parameter is absent. A client MUST treat a failed validation as a connection error of type TRANSPORT_PARAMETER_ERROR.
-A Retry packet does not include a packet number and cannot be explicitly acknowledged by a client.
-This version of QUIC defines a single packet type which uses the short packet header.
- - +The server populates the Destination Connection ID with the connection ID that +the client included in the Source Connection ID of the Initial packet.¶
+The server includes a connection ID of its choice in the Source Connection ID +field. This value MUST not be equal to the Destination Connection ID field of +the packet sent by the client. A client MUST discard a Retry packet that +contains a Source Connection ID field that is identical to the Destination +Connection ID field of its Initial packet. The client MUST use the value from +the Source Connection ID field of the Retry packet in the Destination Connection +ID field of subsequent packets that it sends.¶
+A server MAY send Retry packets in response to Initial and 0-RTT packets. A +server can either discard or buffer 0-RTT packets that it receives. A server +can send multiple Retry packets as it receives Initial or 0-RTT packets. A +server MUST NOT send more than one Retry packet in response to a single UDP +datagram.¶
+A client MUST accept and process at most one Retry packet for each connection +attempt. After the client has received and processed an Initial or Retry packet +from the server, it MUST discard any subsequent Retry packets that it receives.¶
+Clients MUST discard Retry packets that contain an Original Destination +Connection ID field that does not match the Destination Connection ID from its +Initial packet. This prevents an off-path attacker from injecting a Retry +packet. A client MUST discard a Retry packet with a zero-length Retry Token +field.¶
+The client responds to a Retry packet with an Initial packet that includes the +provided Retry Token to continue connection establishment.¶
+A client sets the Destination Connection ID field of this Initial packet to the +value from the Source Connection ID in the Retry packet. Changing Destination +Connection ID also results in a change to the keys used to protect the Initial +packet. It also sets the Token field to the token provided in the Retry. The +client MUST NOT change the Source Connection ID because the server could include +the connection ID as part of its token validation logic (see +Section 8.1.4).¶
+The next Initial packet from the client uses the connection ID and token values +from the Retry packet (see Section 7.2). Aside from this, +the Initial packet sent by the client is subject to the same restrictions as the +first Initial packet. A client MUST use the same cryptographic handshake +message it includes in this packet. A server MAY treat a packet that +contains a different cryptographic handshake message as a connection error or +discard it.¶
+A client MAY attempt 0-RTT after receiving a Retry packet by sending 0-RTT +packets to the connection ID provided by the server. A client MUST NOT change +the cryptographic handshake message it sends in response to receiving a Retry.¶
+A client MUST NOT reset the packet number for any packet number space after +processing a Retry packet; Section 17.2.3 contains more information on this.¶
+A server acknowledges the use of a Retry packet for a connection using the +original_connection_id transport parameter (see +Section 18.2). If the server sends a Retry packet, it +MUST include the value of the Original Destination Connection ID field of the +Retry packet (that is, the Destination Connection ID field from the client's +first Initial packet) in the transport parameter.¶
+If the client received and processed a Retry packet, it MUST validate that the +original_connection_id transport parameter is present and correct; otherwise, it +MUST validate that the transport parameter is absent. A client MUST treat a +failed validation as a connection error of type TRANSPORT_PARAMETER_ERROR.¶
+A Retry packet does not include a packet number and cannot be explicitly +acknowledged by a client.¶
+This version of QUIC defines a single packet type which uses the +short packet header.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2736,49 +6224,138 @@-| Protected Payload (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 14: Short Header Packet Format
-The short header can be used after the version and 1-RTT keys are negotiated. Packets that use the short header contain the following fields:
- - -The short header can be used after the version and 1-RTT keys are negotiated. +Packets that use the short header contain the following fields:¶
+The header form bit and the connection ID field of a short header packet are version-independent. The remaining fields are specific to the selected QUIC version. See [QUIC-INVARIANTS] for details on how packets from different versions of QUIC are interpreted.
-The latency spin bit enables passive latency monitoring from observation points on the network path throughout the duration of a connection. The spin bit is only present in the short packet header, since it is possible to measure the initial RTT of a connection by observing the handshake. Therefore, the spin bit is available after version negotiation and connection establishment are completed. On-path measurement and use of the latency spin bit is further discussed in [QUIC-MANAGEABILITY].
-The spin bit is an OPTIONAL feature of QUIC. A QUIC stack that chooses to support the spin bit MUST implement it as specified in this section.
-Each endpoint unilaterally decides if the spin bit is enabled or disabled for a connection. Implementations MUST allow administrators of clients and servers to disable the spin bit either globally or on a per-connection basis. Even when the spin bit is not disabled by the administrator, implementations MUST disable the spin bit for a given connection with a certain likelihood. The random selection process SHOULD be designed such that on average the spin bit is disabled for at least one eighth of network paths. The selection process performed at the beginning of the connection SHOULD be applied for all paths used by the connection.
-When the spin bit is disabled, endpoints MAY set the spin bit to any value, and MUST ignore any incoming value. It is RECOMMENDED that endpoints set the spin bit to a random value either chosen independently for each packet or chosen independently for each connection ID.
-If the spin bit is enabled for the connection, the endpoint maintains a spin value and sets the spin bit in the short header to the currently stored value when a packet with a short header is sent out. The spin value is initialized to 0 in the endpoint at connection start. Each endpoint also remembers the highest packet number seen from its peer on the connection.
-When a server receives a short header packet that increments the highest packet number seen by the server from the client, it sets the spin value to be equal to the spin bit in the received packet.
-When a client receives a short header packet that increments the highest packet number seen by the client from the server, it sets the spin value to the inverse of the spin bit in the received packet.
-An endpoint resets its spin value to zero when sending the first packet of a given connection with a new connection ID. This reduces the risk that transient spin bit state can be used to link flows across connection migration or ID change.
-With this mechanism, the server reflects the spin value received, while the client ‘spins’ it after one RTT. On-path observers can measure the time between two spin bit toggle events to estimate the end-to-end RTT of a connection.
-The extension_data field of the quic_transport_parameters extension defined in [QUIC-TLS] contains the QUIC transport parameters. They are encoded as a length-prefixed sequence of transport parameters, as shown in Figure 15:
- - +The header form bit and the connection ID field of a short header packet are +version-independent. The remaining fields are specific to the selected QUIC +version. See [QUIC-INVARIANTS] for details on how packets from different +versions of QUIC are interpreted.¶
+The latency spin bit enables passive latency monitoring from observation points +on the network path throughout the duration of a connection. The spin bit is +only present in the short packet header, since it is possible to measure the +initial RTT of a connection by observing the handshake. Therefore, the spin bit +is available after version negotiation and connection establishment are +completed. On-path measurement and use of the latency spin bit is further +discussed in [QUIC-MANAGEABILITY].¶
+The spin bit is an OPTIONAL feature of QUIC. A QUIC stack that chooses to +support the spin bit MUST implement it as specified in this section.¶
+Each endpoint unilaterally decides if the spin bit is enabled or disabled for a +connection. Implementations MUST allow administrators of clients and servers +to disable the spin bit either globally or on a per-connection basis. Even when +the spin bit is not disabled by the administrator, endpoints MUST disable their +use of the spin bit for a random selection of at least one in every 16 network +paths, or for one in every 16 connection IDs. As each endpoint disables the +spin bit independently, this ensures that the spin bit signal is disabled on +approximately one in eight network paths.¶
+When the spin bit is disabled, endpoints MAY set the spin bit to any value, and +MUST ignore any incoming value. It is RECOMMENDED that endpoints set the spin +bit to a random value either chosen independently for each packet or chosen +independently for each connection ID.¶
+If the spin bit is enabled for the connection, the endpoint maintains a spin +value and sets the spin bit in the short header to the currently stored +value when a packet with a short header is sent out. The spin value is +initialized to 0 in the endpoint at connection start. Each endpoint also +remembers the highest packet number seen from its peer on the connection.¶
+When a server receives a short header packet that increments the highest +packet number seen by the server from the client, it sets the spin value to be +equal to the spin bit in the received packet.¶
+When a client receives a short header packet that increments the highest +packet number seen by the client from the server, it sets the spin value to the +inverse of the spin bit in the received packet.¶
+An endpoint resets its spin value to zero when sending the first packet of a +given connection with a new connection ID. This reduces the risk that transient +spin bit state can be used to link flows across connection migration or ID +change.¶
+With this mechanism, the server reflects the spin value received, while the +client 'spins' it after one RTT. On-path observers can measure the time +between two spin bit toggle events to estimate the end-to-end RTT of a +connection.¶
+The extension_data field of the quic_transport_parameters extension defined in
+[QUIC-TLS] contains the QUIC transport parameters. They are encoded as a
+length-prefixed sequence of transport parameters, as shown in
+Figure 16:¶
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2794,10 +6371,17 @@-| Transport Parameter N (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 15: Sequence of Transport Parameters
-The Sequence Length field contains the length of the sequence of transport parameters, in bytes. Each transport parameter is encoded as an (identifier, length, value) tuple, as shown in Figure 16:
- - +The Sequence Length field contains the length of the sequence of transport +parameters, in bytes. Each transport parameter is encoded as an (identifier, +length, value) tuple, as shown in Figure 17:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2807,53 +6391,165 @@-| Transport Parameter Value (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 16: Transport Parameter Encoding
-The Transport Param Length field contains the length of the Transport Parameter Value field.
-QUIC encodes transport parameters into a sequence of bytes, which are then included in the cryptographic handshake.
-Transport parameters with an identifier of the form 31 * N + 27 for integer values of N are reserved to exercise the requirement that unknown transport parameters be ignored. These transport parameters have no semantics, and may carry arbitrary values.
-This section details the transport parameters defined in this document.
-Many transport parameters listed here have integer values. Those transport parameters that are identified as integers use a variable-length integer encoding (see Section 16) and have a default value of 0 if the transport parameter is absent, unless otherwise stated.
-The following transport parameters are defined:
- - -The Transport Param Length field contains the length of the Transport +Parameter Value field.¶
+QUIC encodes transport parameters into a sequence of bytes, which are then +included in the cryptographic handshake.¶
+Transport parameters with an identifier of the form 31 * N + 27 for integer
+values of N are reserved to exercise the requirement that unknown transport
+parameters be ignored. These transport parameters have no semantics, and may
+carry arbitrary values.¶
This section details the transport parameters defined in this document.¶
+Many transport parameters listed here have integer values. Those transport +parameters that are identified as integers use a variable-length integer +encoding (see Section 16) and have a default value of 0 if the +transport parameter is absent, unless otherwise stated.¶
+The following transport parameters are defined:¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2885,41 +6581,111 @@-| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 17: Preferred Address format
- - -If present, transport parameters that set initial flow control limits (initial_max_stream_data_bidi_local, initial_max_stream_data_bidi_remote, and initial_max_stream_data_uni) are equivalent to sending a MAX_STREAM_DATA frame (Section 19.10) on every stream of the corresponding type immediately after opening. If the transport parameter is absent, streams of that type start with a flow control limit of 0.
-A client MUST NOT include server-only transport parameters (original_connection_id, stateless_reset_token, or preferred_address). A server MUST treat receipt of any of these transport parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.
-As described in Section 12.4, packets contain one or more frames. This section describes the format and semantics of the core QUIC frame types.
-The PADDING frame (type=0x00) has no semantic value. PADDING frames can be used to increase the size of a packet. Padding can be used to increase an initial client packet to the minimum required size, or to provide protection against traffic analysis for protected packets.
-A PADDING frame has no content. That is, a PADDING frame consists of the single byte that identifies the frame as a PADDING frame.
-Endpoints can use PING frames (type=0x01) to verify that their peers are still alive or to check reachability to the peer. The PING frame contains no additional fields.
-The receiver of a PING frame simply needs to acknowledge the packet containing this frame.
-The PING frame can be used to keep a connection alive when an application or application protocol wishes to prevent the connection from timing out. An application protocol SHOULD provide guidance about the conditions under which generating a PING is recommended. This guidance SHOULD indicate whether it is the client or the server that is expected to send the PING. Having both endpoints send PING frames without coordination can produce an excessive number of packets and poor performance.
-A connection will time out if no packets are sent or received for a period longer than the time specified in the idle_timeout transport parameter (see Section 10). However, state in middleboxes might time out earlier than that. Though REQ-5 in [RFC4787] recommends a 2 minute timeout interval, experience shows that sending packets every 15 to 30 seconds is necessary to prevent the majority of middleboxes from losing state for UDP flows.
-Receivers send ACK frames (types 0x02 and 0x03) to inform senders of packets they have received and processed. The ACK frame contains one or more ACK Ranges. ACK Ranges identify acknowledged packets. If the frame type is 0x03, ACK frames also contain the sum of QUIC packets with associated ECN marks received on the connection up until this point. QUIC implementations MUST properly handle both types and, if they have enabled ECN for packets they send, they SHOULD use the information in the ECN section to manage their congestion state.
-QUIC acknowledgements are irrevocable. Once acknowledged, a packet remains acknowledged, even if it does not appear in a future ACK frame. This is unlike TCP SACKs ([RFC2018]).
-It is expected that a sender will reuse the same packet number across different packet number spaces. ACK frames only acknowledge the packet numbers that were transmitted by the sender in the same packet number space of the packet that the ACK was received in.
-Version Negotiation and Retry packets cannot be acknowledged because they do not contain a packet number. Rather than relying on ACK frames, these packets are implicitly acknowledged by the next Initial packet sent by the client.
-An ACK frame is as follows:
- - +If present, transport parameters that set initial flow control limits +(initial_max_stream_data_bidi_local, initial_max_stream_data_bidi_remote, and +initial_max_stream_data_uni) are equivalent to sending a MAX_STREAM_DATA frame +(Section 19.10) on every stream of the corresponding type +immediately after opening. If the transport parameter is absent, streams of +that type start with a flow control limit of 0.¶
+A client MUST NOT include server-only transport parameters +(original_connection_id, stateless_reset_token, or preferred_address). A server +MUST treat receipt of any of these transport parameters as a connection error of +type TRANSPORT_PARAMETER_ERROR.¶
+As described in Section 12.4, packets contain one or more frames. This section +describes the format and semantics of the core QUIC frame types.¶
+The PADDING frame (type=0x00) has no semantic value. PADDING frames can be used +to increase the size of a packet. Padding can be used to increase an initial +client packet to the minimum required size, or to provide protection against +traffic analysis for protected packets.¶
+A PADDING frame has no content. That is, a PADDING frame consists of the single +byte that identifies the frame as a PADDING frame.¶
+Endpoints can use PING frames (type=0x01) to verify that their peers are still +alive or to check reachability to the peer. The PING frame contains no +additional fields.¶
+The receiver of a PING frame simply needs to acknowledge the packet containing +this frame.¶
+The PING frame can be used to keep a connection alive when an application or +application protocol wishes to prevent the connection from timing out. An +application protocol SHOULD provide guidance about the conditions under which +generating a PING is recommended. This guidance SHOULD indicate whether it is +the client or the server that is expected to send the PING. Having both +endpoints send PING frames without coordination can produce an excessive number +of packets and poor performance.¶
+A connection will time out if no packets are sent or received for a period +longer than the time specified in the idle_timeout transport parameter (see +Section 10). However, state in middleboxes might time out earlier than +that. Though REQ-5 in [RFC4787] recommends a 2 minute timeout interval, +experience shows that sending packets every 15 to 30 seconds is necessary to +prevent the majority of middleboxes from losing state for UDP flows.¶
+Receivers send ACK frames (types 0x02 and 0x03) to inform senders of packets +they have received and processed. The ACK frame contains one or more ACK Ranges. +ACK Ranges identify acknowledged packets. If the frame type is 0x03, ACK frames +also contain the sum of QUIC packets with associated ECN marks received on the +connection up until this point. QUIC implementations MUST properly handle both +types and, if they have enabled ECN for packets they send, they SHOULD use the +information in the ECN section to manage their congestion state.¶
+QUIC acknowledgements are irrevocable. Once acknowledged, a packet remains +acknowledged, even if it does not appear in a future ACK frame. This is unlike +TCP SACKs ([RFC2018]).¶
+Packets from different packet number spaces can be identified using the same +numeric value. An acknowledgment for a packet needs to indicate both a packet +number and a packet number space. This is accomplished by having each ACK frame +only acknowledge packet numbers in the same space as the packet in which the +ACK frame is contained.¶
+Version Negotiation and Retry packets cannot be acknowledged because they do not +contain a packet number. Rather than relying on ACK frames, these packets are +implicitly acknowledged by the next Initial packet sent by the client.¶
+An ACK frame is shown in Figure 19.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2937,31 +6703,69 @@-| [ECN Counts] ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 18: ACK Frame Format
-ACK frames contain the following fields:
- - -ACK frames contain the following fields:¶
+ack_delay_exponent transport parameter set by the
+sender of the ACK frame (see Section 18.2). Scaling in
+this fashion allows for a larger range of values with a shorter encoding at
+the cost of lower resolution. Because the receiver doesn't use the ACK Delay
+for Initial and Handshake packets, a sender SHOULD send a value of 0.¶
+The ACK Ranges field consists of alternating Gap and ACK Range values in descending packet number order. The number of Gap and ACK Range values is determined by the ACK Range Count field; one of each value is present for each value in the ACK Range Count field.
-ACK Ranges are structured as follows:
- - +The ACK Ranges field consists of alternating Gap and ACK Range values in +descending packet number order. The number of Gap and ACK Range values is +determined by the ACK Range Count field; one of each value is present for each +value in the ACK Range Count field.¶
+ACK Ranges are structured as shown in Figure 20.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -2981,34 +6785,73 @@-| [ACK Range (i)] ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 19: ACK Ranges
-The fields that form the ACK Ranges are:
- - -The fields that form the ACK Ranges are:¶
+Gap and ACK Range value use a relative integer encoding for efficiency. Though each encoded value is positive, the values are subtracted, so that each ACK Range describes progressively lower-numbered packets.
-Each ACK Range acknowledges a contiguous range of packets by indicating the number of acknowledged packets that precede the largest packet number in that range. A value of zero indicates that only the largest packet number is acknowledged. Larger ACK Range values indicate a larger range, with corresponding lower values for the smallest packet number in the range. Thus, given a largest packet number for the range, the smallest value is determined by the formula:
+Gap and ACK Range value use a relative integer encoding for efficiency. Though +each encoded value is positive, the values are subtracted, so that each ACK +Range describes progressively lower-numbered packets.¶
+Each ACK Range acknowledges a contiguous range of packets by indicating the +number of acknowledged packets that precede the largest packet number in that +range. A value of zero indicates that only the largest packet number is +acknowledged. Larger ACK Range values indicate a larger range, with +corresponding lower values for the smallest packet number in the range. Thus, +given a largest packet number for the range, the smallest value is determined by +the formula:¶
+
smallest = largest - ack_range
-
-An ACK Range acknowledges all packets between the smallest packet number and the largest, inclusive.
-The largest value for an ACK Range is determined by cumulatively subtracting the size of all preceding ACK Ranges and Gaps.
-Each Gap indicates a range of packets that are not being acknowledged. The number of packets in the gap is one higher than the encoded value of the Gap field.
-The value of the Gap field establishes the largest packet number value for the subsequent ACK Range using the following formula:
+¶ +An ACK Range acknowledges all packets between the smallest packet number and the +largest, inclusive.¶
+The largest value for an ACK Range is determined by cumulatively subtracting the +size of all preceding ACK Ranges and Gaps.¶
+Each Gap indicates a range of packets that are not being acknowledged. The +number of packets in the gap is one higher than the encoded value of the Gap +field.¶
+The value of the Gap field establishes the largest packet number value for the +subsequent ACK Range using the following formula:¶
+
largest = previous_smallest - gap - 2
-
-If any computed packet number is negative, an endpoint MUST generate a connection error of type FRAME_ENCODING_ERROR.
-The ACK frame uses the least significant bit (that is, type 0x03) to indicate ECN feedback and report receipt of QUIC packets with associated ECN codepoints of ECT(0), ECT(1), or CE in the packet’s IP header. ECN Counts are only present when the ACK frame type is 0x03.
-ECN Counts are only parsed when the ACK frame type is 0x03. There are 3 ECN counts, as follows:
+¶ +If any computed packet number is negative, an endpoint MUST generate a +connection error of type FRAME_ENCODING_ERROR.¶
+The ACK frame uses the least significant bit (that is, type 0x03) to indicate +ECN feedback and report receipt of QUIC packets with associated ECN codepoints +of ECT(0), ECT(1), or CE in the packet's IP header. ECN Counts are only present +when the ACK frame type is 0x03.¶
+ECN Counts are only parsed when the ACK frame type is 0x03. There are 3 ECN +counts, as shown in Figure 21.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3020,25 +6863,50 @@-| ECN-CE Count (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The three ECN Counts are:
- - -The three ECN Counts are:¶
+ECN counts are maintained separately for each packet number space.
-An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly terminate the sending part of a stream.
-After sending a RESET_STREAM, an endpoint ceases transmission and retransmission of STREAM frames on the identified stream. A receiver of RESET_STREAM can discard any data that it already received on that stream.
-An endpoint that receives a RESET_STREAM frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The RESET_STREAM frame is as follows:
+ECN counts are maintained separately for each packet number space.¶
+An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly terminate the +sending part of a stream.¶
+After sending a RESET_STREAM, an endpoint ceases transmission and retransmission +of STREAM frames on the identified stream. A receiver of RESET_STREAM can +discard any data that it already received on that stream.¶
+An endpoint that receives a RESET_STREAM frame for a send-only stream MUST +terminate the connection with error STREAM_STATE_ERROR.¶
+The RESET_STREAM frame is shown in Figure 22.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3050,23 +6918,50 @@-| Final Size (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
RESET_STREAM frames contain the following fields:
- - -RESET_STREAM frames contain the following fields:¶
+An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that incoming data is being discarded on receipt at application request. STOP_SENDING requests that a peer cease transmission on a stream.
-A STOP_SENDING frame can be sent for streams in the Recv or Size Known states (see Section 3.1). Receiving a STOP_SENDING frame for a locally-initiated stream that has not yet been created MUST be treated as a connection error of type STREAM_STATE_ERROR. An endpoint that receives a STOP_SENDING frame for a receive-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STOP_SENDING frame is as follows:
+An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that incoming +data is being discarded on receipt at application request. STOP_SENDING +requests that a peer cease transmission on a stream.¶
+A STOP_SENDING frame can be sent for streams in the Recv or Size Known states +(see Section 3.1). Receiving a STOP_SENDING frame for a +locally-initiated stream that has not yet been created MUST be treated as a +connection error of type STREAM_STATE_ERROR. An endpoint that receives a +STOP_SENDING frame for a receive-only stream MUST terminate the connection with +error STREAM_STATE_ERROR.¶
+The STOP_SENDING frame is shown in Figure 23.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3076,22 +6971,40 @@-| Application Error Code (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
STOP_SENDING frames contain the following fields:
- - -STOP_SENDING frames contain the following fields:¶
+The CRYPTO frame (type=0x06) is used to transmit cryptographic handshake messages. It can be sent in all packet types except 0-RTT. The CRYPTO frame offers the cryptographic protocol an in-order stream of bytes. CRYPTO frames are functionally identical to STREAM frames, except that they do not bear a stream identifier; they are not flow controlled; and they do not carry markers for optional offset, optional length, and the end of the stream.
-The CRYPTO frame is as follows:
- - +The CRYPTO frame (type=0x06) is used to transmit cryptographic handshake +messages. It can be sent in all packet types except 0-RTT. The CRYPTO frame +offers the cryptographic protocol an in-order stream of bytes. CRYPTO frames +are functionally identical to STREAM frames, except that they do not bear a +stream identifier; they are not flow controlled; and they do not carry markers +for optional offset, optional length, and the end of the stream.¶
+The CRYPTO frame is shown in Figure 24.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3103,26 +7016,52 @@-| Crypto Data (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 20: CRYPTO Frame Format
-CRYPTO frames contain the following fields:
- - -CRYPTO frames contain the following fields:¶
+There is a separate flow of cryptographic handshake data in each encryption level, each of which starts at an offset of 0. This implies that each encryption level is treated as a separate CRYPTO stream of data.
-The largest offset delivered on a stream - the sum of the offset and data length - cannot exceed 2^62-1. Receipt of a frame that exceeds this limit MUST be treated as a connection error of type FRAME_ENCODING_ERROR.
-Unlike STREAM frames, which include a Stream ID indicating to which stream the data belongs, the CRYPTO frame carries data for a single stream per encryption level. The stream does not have an explicit end, so CRYPTO frames do not have a FIN bit.
-A server sends a NEW_TOKEN frame (type=0x07) to provide the client with a token to send in the header of an Initial packet for a future connection.
-The NEW_TOKEN frame is as follows:
+There is a separate flow of cryptographic handshake data in each encryption +level, each of which starts at an offset of 0. This implies that each encryption +level is treated as a separate CRYPTO stream of data.¶
+The largest offset delivered on a stream - the sum of the offset and data +length - cannot exceed 2^62-1. Receipt of a frame that exceeds this limit MUST +be treated as a connection error of type FRAME_ENCODING_ERROR or +CRYPTO_BUFFER_EXCEEDED.¶
+Unlike STREAM frames, which include a Stream ID indicating to which stream the +data belongs, the CRYPTO frame carries data for a single stream per encryption +level. The stream does not have an explicit end, so CRYPTO frames do not have a +FIN bit.¶
+A server sends a NEW_TOKEN frame (type=0x07) to provide the client with a token +to send in the header of an Initial packet for a future connection.¶
+The NEW_TOKEN frame is shown in Figure 25.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3132,32 +7071,64 @@-| Token (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
NEW_TOKEN frames contain the following fields:
- - -NEW_TOKEN frames contain the following fields:¶
+An endpoint might receive multiple NEW_TOKEN frames that contain the same token value. Endpoints are responsible for discarding duplicate values, which might be used to link connection attempts; see Section 8.1.2.
-Clients MUST NOT send NEW_TOKEN frames. Servers MUST treat receipt of a NEW_TOKEN frame as a connection error of type PROTOCOL_VIOLATION.
-STREAM frames implicitly create a stream and carry stream data. The STREAM frame takes the form 0b00001XXX (or the set of values from 0x08 to 0x0f). The value of the three low-order bits of the frame type determines the fields that are present in the frame.
- - +An endpoint might receive multiple NEW_TOKEN frames that contain the same token +value if packets containing the frame are incorrectly determined to be lost. +Endpoints are responsible for discarding duplicate values, which might be used +to link connection attempts; see Section 8.1.3.¶
+Clients MUST NOT send NEW_TOKEN frames. Servers MUST treat receipt of a +NEW_TOKEN frame as a connection error of type PROTOCOL_VIOLATION.¶
+STREAM frames implicitly create a stream and carry stream data. The STREAM +frame takes the form 0b00001XXX (or the set of values from 0x08 to 0x0f). The +value of the three low-order bits of the frame type determines the fields that +are present in the frame.¶
An endpoint that receives a STREAM frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STREAM frames are as follows:
- - +An endpoint that receives a STREAM frame for a send-only stream MUST terminate +the connection with error STREAM_STATE_ERROR.¶
+The STREAM frames are shown in Figure 26.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3171,27 +7142,56 @@-| Stream Data (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 21: STREAM Frame Format
-STREAM frames contain the following fields:
- - -STREAM frames contain the following fields:¶
+When a Stream Data field has a length of 0, the offset in the STREAM frame is the offset of the next byte that would be sent.
-The first byte in the stream has an offset of 0. The largest offset delivered on a stream - the sum of the offset and data length - cannot exceed 2^62-1, as it is not possible to provide flow control credit for that data. Receipt of a frame that exceeds this limit MUST be treated as a connection error of type FRAME_ENCODING_ERROR or FLOW_CONTROL_ERROR.
-The MAX_DATA frame (type=0x10) is used in flow control to inform the peer of the maximum amount of data that can be sent on the connection as a whole.
-The MAX_DATA frame is as follows:
+When a Stream Data field has a length of 0, the offset in the STREAM frame is +the offset of the next byte that would be sent.¶
+The first byte in the stream has an offset of 0. The largest offset delivered +on a stream - the sum of the offset and data length - cannot exceed 2^62-1, as +it is not possible to provide flow control credit for that data. Receipt of a +frame that exceeds this limit MUST be treated as a connection error of type +FRAME_ENCODING_ERROR or FLOW_CONTROL_ERROR.¶
+The MAX_DATA frame (type=0x10) is used in flow control to inform the peer of +the maximum amount of data that can be sent on the connection as a whole.¶
+The MAX_DATA frame is shown in Figure 27.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3199,20 +7199,44 @@-| Maximum Data (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
MAX_DATA frames contain the following fields:
- - -MAX_DATA frames contain the following fields:¶
+All data sent in STREAM frames counts toward this limit. The sum of the largest received offsets on all streams - including streams in terminal states - MUST NOT exceed the value advertised by a receiver. An endpoint MUST terminate a connection with a FLOW_CONTROL_ERROR error if it receives more data than the maximum data value that it has sent, unless this is a result of a change in the initial limits (see Section 7.3.1).
-The MAX_STREAM_DATA frame (type=0x11) is used in flow control to inform a peer of the maximum amount of data that can be sent on a stream.
-A MAX_STREAM_DATA frame can be sent for streams in the Recv state (see Section 3.1). Receiving a MAX_STREAM_DATA frame for a locally-initiated stream that has not yet been created MUST be treated as a connection error of type STREAM_STATE_ERROR. An endpoint that receives a MAX_STREAM_DATA frame for a receive-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The MAX_STREAM_DATA frame is as follows:
+All data sent in STREAM frames counts toward this limit. The sum of the largest +received offsets on all streams - including streams in terminal states - MUST +NOT exceed the value advertised by a receiver. An endpoint MUST terminate a +connection with a FLOW_CONTROL_ERROR error if it receives more data than the +maximum data value that it has sent, unless this is a result of a change in +the initial limits (see Section 7.3.1).¶
+The MAX_STREAM_DATA frame (type=0x11) is used in flow control to inform a peer +of the maximum amount of data that can be sent on a stream.¶
+A MAX_STREAM_DATA frame can be sent for streams in the Recv state (see +Section 3.1). Receiving a MAX_STREAM_DATA frame for a +locally-initiated stream that has not yet been created MUST be treated as a +connection error of type STREAM_STATE_ERROR. An endpoint that receives a +MAX_STREAM_DATA frame for a receive-only stream MUST terminate the connection +with error STREAM_STATE_ERROR.¶
+The MAX_STREAM_DATA frame is shown in Figure 28.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3222,22 +7246,49 @@-| Maximum Stream Data (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
MAX_STREAM_DATA frames contain the following fields:
- - -MAX_STREAM_DATA frames contain the following fields:¶
+When counting data toward this limit, an endpoint accounts for the largest received offset of data that is sent or received on the stream. Loss or reordering can mean that the largest received offset on a stream can be greater than the total size of data received on that stream. Receiving STREAM frames might not increase the largest received offset.
-The data sent on a stream MUST NOT exceed the largest maximum stream data value advertised by the receiver. An endpoint MUST terminate a connection with a FLOW_CONTROL_ERROR error if it receives more data than the largest maximum stream data that it has sent for the affected stream, unless this is a result of a change in the initial limits (see Section 7.3.1).
-The MAX_STREAMS frames (type=0x12 and 0x13) inform the peer of the cumulative number of streams of a given type it is permitted to open. A MAX_STREAMS frame with a type of 0x12 applies to bidirectional streams, and a MAX_STREAMS frame with a type of 0x13 applies to unidirectional streams.
-The MAX_STREAMS frames are as follows:
+When counting data toward this limit, an endpoint accounts for the largest +received offset of data that is sent or received on the stream. Loss or +reordering can mean that the largest received offset on a stream can be greater +than the total size of data received on that stream. Receiving STREAM frames +might not increase the largest received offset.¶
+The data sent on a stream MUST NOT exceed the largest maximum stream data value +advertised by the receiver. An endpoint MUST terminate a connection with a +FLOW_CONTROL_ERROR error if it receives more data than the largest maximum +stream data that it has sent for the affected stream, unless this is a result of +a change in the initial limits (see Section 7.3.1).¶
+The MAX_STREAMS frames (type=0x12 and 0x13) inform the peer of the cumulative +number of streams of a given type it is permitted to open. A MAX_STREAMS frame +with a type of 0x12 applies to bidirectional streams, and a MAX_STREAMS frame +with a type of 0x13 applies to unidirectional streams.¶
+The MAX_STREAMS frames are shown in Figure 29;¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3245,21 +7296,48 @@-| Maximum Streams (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
MAX_STREAMS frames contain the following fields:
- - -MAX_STREAMS frames contain the following fields:¶
+Loss or reordering can cause a MAX_STREAMS frame to be received which states a lower stream limit than an endpoint has previously received. MAX_STREAMS frames which do not increase the stream limit MUST be ignored.
-An endpoint MUST NOT open more streams than permitted by the current stream limit set by its peer. For instance, a server that receives a unidirectional stream limit of 3 is permitted to open stream 3, 7, and 11, but not stream 15. An endpoint MUST terminate a connection with a STREAM_LIMIT_ERROR error if a peer opens more streams than was permitted.
-Note that these frames (and the corresponding transport parameters) do not describe the number of streams that can be opened concurrently. The limit includes streams that have been closed as well as those that are open.
-A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes to send data, but is unable to due to connection-level flow control (see Section 4). DATA_BLOCKED frames can be used as input to tuning of flow control algorithms (see Section 4.2).
-The DATA_BLOCKED frame is as follows:
+Loss or reordering can cause a MAX_STREAMS frame to be received which states a +lower stream limit than an endpoint has previously received. MAX_STREAMS frames +which do not increase the stream limit MUST be ignored.¶
+An endpoint MUST NOT open more streams than permitted by the current stream +limit set by its peer. For instance, a server that receives a unidirectional +stream limit of 3 is permitted to open stream 3, 7, and 11, but not stream 15. +An endpoint MUST terminate a connection with a STREAM_LIMIT_ERROR error if a +peer opens more streams than was permitted.¶
+Note that these frames (and the corresponding transport parameters) do not +describe the number of streams that can be opened concurrently. The limit +includes streams that have been closed as well as those that are open.¶
+A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes to send +data, but is unable to due to connection-level flow control (see +Section 4). DATA_BLOCKED frames can be used as input to tuning of flow +control algorithms (see Section 4.2).¶
+The DATA_BLOCKED frame is shown in Figure 30.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3267,19 +7345,35 @@-| Data Limit (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
DATA_BLOCKED frames contain the following fields:
- - -DATA_BLOCKED frames contain the following fields:¶
+A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it wishes to send data, but is unable to due to stream-level flow control. This frame is analogous to DATA_BLOCKED (Section 19.12).
-An endpoint that receives a STREAM_DATA_BLOCKED frame for a send-only stream MUST terminate the connection with error STREAM_STATE_ERROR.
-The STREAM_DATA_BLOCKED frame is as follows:
+A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it wishes to +send data, but is unable to due to stream-level flow control. This frame is +analogous to DATA_BLOCKED (Section 19.12).¶
+An endpoint that receives a STREAM_DATA_BLOCKED frame for a send-only stream +MUST terminate the connection with error STREAM_STATE_ERROR.¶
+The STREAM_DATA_BLOCKED frame is shown in Figure 31.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3289,21 +7383,41 @@-| Stream Data Limit (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
STREAM_DATA_BLOCKED frames contain the following fields:
- - -STREAM_DATA_BLOCKED frames contain the following fields:¶
+A sender SHOULD send a STREAMS_BLOCKED frame (type=0x16 or 0x17) when it wishes to open a stream, but is unable to due to the maximum stream limit set by its peer (see Section 19.11). A STREAMS_BLOCKED frame of type 0x16 is used to indicate reaching the bidirectional stream limit, and a STREAMS_BLOCKED frame of type 0x17 indicates reaching the unidirectional stream limit.
-A STREAMS_BLOCKED frame does not open the stream, but informs the peer that a new stream was needed and the stream limit prevented the creation of the stream.
-The STREAMS_BLOCKED frames are as follows:
+A sender SHOULD send a STREAMS_BLOCKED frame (type=0x16 or 0x17) when it wishes +to open a stream, but is unable to due to the maximum stream limit set by its +peer (see Section 19.11). A STREAMS_BLOCKED frame of type 0x16 is used +to indicate reaching the bidirectional stream limit, and a STREAMS_BLOCKED frame +of type 0x17 indicates reaching the unidirectional stream limit.¶
+A STREAMS_BLOCKED frame does not open the stream, but informs the peer that a +new stream was needed and the stream limit prevented the creation of the stream.¶
+The STREAMS_BLOCKED frames are shown in Figure 32.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3311,18 +7425,35 @@-| Stream Limit (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
STREAMS_BLOCKED frames contain the following fields:
- - -STREAMS_BLOCKED frames contain the following fields:¶
+An endpoint sends a NEW_CONNECTION_ID frame (type=0x18) to provide its peer with alternative connection IDs that can be used to break linkability when migrating connections (see Section 9.5).
-The NEW_CONNECTION_ID frame is as follows:
+An endpoint sends a NEW_CONNECTION_ID frame (type=0x18) to provide its peer with +alternative connection IDs that can be used to break linkability when migrating +connections (see Section 9.5).¶
+The NEW_CONNECTION_ID frame is shown in Figure 33.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3344,32 +7475,87 @@-| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
NEW_CONNECTION_ID frames contain the following fields:
- - -NEW_CONNECTION_ID frames contain the following fields:¶
+An endpoint MUST NOT send this frame if it currently requires that its peer send packets with a zero-length Destination Connection ID. Changing the length of a connection ID to or from zero-length makes it difficult to identify when the value of the connection ID changed. An endpoint that is sending packets with a zero-length Destination Connection ID MUST treat receipt of a NEW_CONNECTION_ID frame as a connection error of type PROTOCOL_VIOLATION.
-Transmission errors, timeouts and retransmissions might cause the same NEW_CONNECTION_ID frame to be received multiple times. Receipt of the same frame multiple times MUST NOT be treated as a connection error. A receiver can use the sequence number supplied in the NEW_CONNECTION_ID frame to identify new connection IDs from old ones.
-If an endpoint receives a NEW_CONNECTION_ID frame that repeats a previously issued connection ID with a different Stateless Reset Token or a different sequence number, or if a sequence number is used for different connection IDs, the endpoint MAY treat that receipt as a connection error of type PROTOCOL_VIOLATION.
-The Retire Prior To field counts connection IDs established during connection setup and the preferred_address transport parameter (see Section 5.1.2). The Retire Prior To field MUST be less than or equal to the Sequence Number field. Receiving a value greater than the Sequence Number MUST be treated as a connection error of type FRAME_ENCODING_ERROR.
-Once a sender indicates a Retire Prior To value, smaller values sent in subsequent NEW_CONNECTION_ID frames have no effect. A receiver MUST ignore any Retire Prior To fields that do not increase the largest received Retire Prior To value.
-An endpoint sends a RETIRE_CONNECTION_ID frame (type=0x19) to indicate that it will no longer use a connection ID that was issued by its peer. This may include the connection ID provided during the handshake. Sending a RETIRE_CONNECTION_ID frame also serves as a request to the peer to send additional connection IDs for future use (see Section 5.1). New connection IDs can be delivered to a peer using the NEW_CONNECTION_ID frame (Section 19.15).
-Retiring a connection ID invalidates the stateless reset token associated with that connection ID.
-The RETIRE_CONNECTION_ID frame is as follows:
+An endpoint MUST NOT send this frame if it currently requires that its peer send +packets with a zero-length Destination Connection ID. Changing the length of a +connection ID to or from zero-length makes it difficult to identify when the +value of the connection ID changed. An endpoint that is sending packets with a +zero-length Destination Connection ID MUST treat receipt of a NEW_CONNECTION_ID +frame as a connection error of type PROTOCOL_VIOLATION.¶
+Transmission errors, timeouts and retransmissions might cause the same +NEW_CONNECTION_ID frame to be received multiple times. Receipt of the same +frame multiple times MUST NOT be treated as a connection error. A receiver can +use the sequence number supplied in the NEW_CONNECTION_ID frame to identify new +connection IDs from old ones.¶
+If an endpoint receives a NEW_CONNECTION_ID frame that repeats a previously +issued connection ID with a different Stateless Reset Token or a different +sequence number, or if a sequence number is used for different connection +IDs, the endpoint MAY treat that receipt as a connection error of type +PROTOCOL_VIOLATION.¶
+The Retire Prior To field counts connection IDs established during connection +setup and the preferred_address transport parameter (see Section 5.1.2). The +Retire Prior To field MUST be less than or equal to the Sequence Number field. +Receiving a value greater than the Sequence Number MUST be treated as a +connection error of type FRAME_ENCODING_ERROR.¶
+Once a sender indicates a Retire Prior To value, smaller values sent in +subsequent NEW_CONNECTION_ID frames have no effect. A receiver MUST ignore any +Retire Prior To fields that do not increase the largest received Retire Prior To +value.¶
+An endpoint that receives a NEW_CONNECTION_ID frame with a sequence number +smaller than the Retire Prior To field of a previously received +NEW_CONNECTION_ID frame MUST immediately send a corresponding +RETIRE_CONNECTION_ID frame that retires the newly received connection ID.¶
+An endpoint sends a RETIRE_CONNECTION_ID frame (type=0x19) to indicate that it +will no longer use a connection ID that was issued by its peer. This may include +the connection ID provided during the handshake. Sending a RETIRE_CONNECTION_ID +frame also serves as a request to the peer to send additional connection IDs for +future use (see Section 5.1). New connection IDs can be delivered to a +peer using the NEW_CONNECTION_ID frame (Section 19.15).¶
+Retiring a connection ID invalidates the stateless reset token associated with +that connection ID.¶
+The RETIRE_CONNECTION_ID frame is shown in Figure 34.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3377,21 +7563,43 @@-| Sequence Number (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
RETIRE_CONNECTION_ID frames contain the following fields:
- - -RETIRE_CONNECTION_ID frames contain the following fields:¶
+Receipt of a RETIRE_CONNECTION_ID frame containing a sequence number greater than any previously sent to the peer MUST be treated as a connection error of type PROTOCOL_VIOLATION.
-The sequence number specified in a RETIRE_CONNECTION_ID frame MUST NOT refer to the Destination Connection ID field of the packet in which the frame is contained. The peer MAY treat this as a connection error of type FRAME_ENCODING_ERROR.
-An endpoint cannot send this frame if it was provided with a zero-length connection ID by its peer. An endpoint that provides a zero-length connection ID MUST treat receipt of a RETIRE_CONNECTION_ID frame as a connection error of type PROTOCOL_VIOLATION.
-Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check reachability to the peer and for path validation during connection migration.
-The PATH_CHALLENGE frames are as follows:
+Receipt of a RETIRE_CONNECTION_ID frame containing a sequence number greater +than any previously sent to the peer MUST be treated as a connection error of +type PROTOCOL_VIOLATION.¶
+The sequence number specified in a RETIRE_CONNECTION_ID frame MUST NOT refer +to the Destination Connection ID field of the packet in which the frame is +contained. The peer MAY treat this as a connection error of type +FRAME_ENCODING_ERROR.¶
+An endpoint cannot send this frame if it was provided with a zero-length +connection ID by its peer. An endpoint that provides a zero-length connection +ID MUST treat receipt of a RETIRE_CONNECTION_ID frame as a connection error of +type PROTOCOL_VIOLATION.¶
+Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check reachability to the +peer and for path validation during connection migration.¶
+The PATH_CHALLENGE frame is shown in Figure 35.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3401,26 +7609,54 @@-| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
PATH_CHALLENGE frames contain the following fields:
- - -PATH_CHALLENGE frames contain the following fields:¶
+A PATH_CHALLENGE frame containing 8 bytes that are hard to guess is sufficient to ensure that it is easier to receive the packet than it is to guess the value correctly.
-The recipient of this frame MUST generate a PATH_RESPONSE frame (Section 19.18) containing the same Data.
-The PATH_RESPONSE frame (type=0x1b) is sent in response to a PATH_CHALLENGE frame. Its format is identical to the PATH_CHALLENGE frame (Section 19.17).
-If the content of a PATH_RESPONSE frame does not match the content of a PATH_CHALLENGE frame previously sent by the endpoint, the endpoint MAY generate a connection error of type PROTOCOL_VIOLATION.
-An endpoint sends a CONNECTION_CLOSE frame (type=0x1c or 0x1d) to notify its peer that the connection is being closed. The CONNECTION_CLOSE with a frame type of 0x1c is used to signal errors at only the QUIC layer, or the absence of errors (with the NO_ERROR code). The CONNECTION_CLOSE frame with a type of 0x1d is used to signal an error with the application that uses QUIC.
-If there are open streams that haven’t been explicitly closed, they are implicitly closed when the connection is closed.
-The CONNECTION_CLOSE frames are as follows:
+A PATH_CHALLENGE frame containing 8 bytes that are hard to guess is sufficient +to ensure that it is easier to receive the packet than it is to guess the value +correctly.¶
+The recipient of this frame MUST generate a PATH_RESPONSE frame +(Section 19.18) containing the same Data.¶
+The PATH_RESPONSE frame (type=0x1b) is sent in response to a PATH_CHALLENGE +frame. Its format is identical to the PATH_CHALLENGE frame +(Section 19.17).¶
+If the content of a PATH_RESPONSE frame does not match the content of a +PATH_CHALLENGE frame previously sent by the endpoint, the endpoint MAY generate +a connection error of type PROTOCOL_VIOLATION.¶
+An endpoint sends a CONNECTION_CLOSE frame (type=0x1c or 0x1d) to notify its +peer that the connection is being closed. The CONNECTION_CLOSE with a frame +type of 0x1c is used to signal errors at only the QUIC layer, or the absence of +errors (with the NO_ERROR code). The CONNECTION_CLOSE frame with a type of 0x1d +is used to signal an error with the application that uses QUIC.¶
+If there are open streams that haven't been explicitly closed, they are +implicitly closed when the connection is closed.¶
+The CONNECTION_CLOSE frames are shown in Figure 36.¶
+0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 @@ -3434,533 +7670,1012 @@-| Reason Phrase (*) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
CONNECTION_CLOSE frames contain the following fields:
- - -CONNECTION_CLOSE frames contain the following fields:¶
+QUIC frames do not use a self-describing encoding. An endpoint therefore needs to understand the syntax of all frames before it can successfully process a packet. This allows for efficient encoding of frames, but it means that an endpoint cannot send a frame of a type that is unknown to its peer.
-An extension to QUIC that wishes to use a new type of frame MUST first ensure that a peer is able to understand the frame. An endpoint can use a transport parameter to signal its willingness to receive one or more extension frame types with the one transport parameter.
-Extension frames MUST be congestion controlled and MUST cause an ACK frame to be sent. The exception is extension frames that replace or supplement the ACK frame. Extension frames are not included in flow control unless specified in the extension.
-An IANA registry is used to manage the assignment of frame types; see Section 22.2.
-QUIC error codes are 62-bit unsigned integers.
-This section lists the defined QUIC transport error codes that may be used in a CONNECTION_CLOSE frame. These errors apply to the entire connection.
- - -The application-specific variant of CONNECTION_CLOSE (type 0x1d) can only be +sent using an 1-RTT packet ([QUIC-TLS], Section 4). When an application +wishes to abandon a connection during the handshake, an endpoint can send a +CONNECTION_CLOSE frame (type 0x1c) with an error code of 0x15a ("user_canceled" +alert; see [TLS13]) in an Initial or a Handshake packet.¶
+QUIC frames do not use a self-describing encoding. An endpoint therefore needs +to understand the syntax of all frames before it can successfully process a +packet. This allows for efficient encoding of frames, but it means that an +endpoint cannot send a frame of a type that is unknown to its peer.¶
+An extension to QUIC that wishes to use a new type of frame MUST first ensure +that a peer is able to understand the frame. An endpoint can use a transport +parameter to signal its willingness to receive one or more extension frame types +with the one transport parameter.¶
+Extension frames MUST be congestion controlled and MUST cause an ACK frame to +be sent. The exception is extension frames that replace or supplement the ACK +frame. Extension frames are not included in flow control unless specified +in the extension.¶
+An IANA registry is used to manage the assignment of frame types; see +Section 22.3.¶
+QUIC error codes are 62-bit unsigned integers.¶
+This section lists the defined QUIC transport error codes that may be used in a +CONNECTION_CLOSE frame. These errors apply to the entire connection.¶
+See Section 22.3 for details of registering new error codes.
-In defining these error codes, several principles are applied. Error conditions that might require specific action on the part of a recipient are given unique codes. Errors that represent common conditions are given specific codes. Absent either of these conditions, error codes are used to identify a general function of the stack, like flow control or transport parameter handling. Finally, generic errors are provided for conditions where implementations are unable or unwilling to use more specific codes.
-Application protocol error codes are 62-bit unsigned integers, but the management of application error codes is left to application protocols. Application protocol error codes are used for the RESET_STREAM frame (Section 19.4), the STOP_SENDING frame (Section 19.5), and the CONNECTION_CLOSE frame with a type of 0x1d (Section 19.19).
-As an encrypted and authenticated transport QUIC provides a range of protections against denial of service. Once the cryptographic handshake is complete, QUIC endpoints discard most packets that are not authenticated, greatly limiting the ability of an attacker to interfere with existing connections.
-Once a connection is established QUIC endpoints might accept some unauthenticated ICMP packets (see Section 14.2), but the use of these packets is extremely limited. The only other type of packet that an endpoint might accept is a stateless reset (Section 10.4) which relies on the token being kept secret until it is used.
-During the creation of a connection, QUIC only provides protection against attack from off the network path. All QUIC packets contain proof that the recipient saw a preceding packet from its peer.
-The first mechanism used is the source and destination connection IDs, which are required to match those set by a peer. Except for an Initial and stateless reset packets, an endpoint only accepts packets that include a destination connection that matches a connection ID the endpoint previously chose. This is the only protection offered for Version Negotiation packets.
-The destination connection ID in an Initial packet is selected by a client to be unpredictable, which serves an additional purpose. The packets that carry the cryptographic handshake are protected with a key that is derived from this connection ID and salt specific to the QUIC version. This allows endpoints to use the same process for authenticating packets that they receive as they use after the cryptographic handshake completes. Packets that cannot be authenticated are discarded. Protecting packets in this fashion provides a strong assurance that the sender of the packet saw the Initial packet and understood it.
-These protections are not intended to be effective against an attacker that is able to receive QUIC packets prior to the connection being established. Such an attacker can potentially send packets that will be accepted by QUIC endpoints. This version of QUIC attempts to detect this sort of attack, but it expects that endpoints will fail to establish a connection rather than recovering. For the most part, the cryptographic handshake protocol [QUIC-TLS] is responsible for detecting tampering during the handshake.
-Endpoints are permitted to use other methods to detect and attempt to recover from interference with the handshake. Invalid packets may be identified and discarded using other methods, but no specific method is mandated in this document.
-An attacker might be able to receive an address validation token (Section 8) from a server and then release the IP address it used to acquire that token. At a later time, the attacker may initiate a 0-RTT connection with a server by spoofing this same address, which might now address a different (victim) endpoint. The attacker can thus potentially cause the server to send an initial congestion window’s worth of data towards the victim.
-Servers SHOULD provide mitigations for this attack by limiting the usage and lifetime of address validation tokens (see Section 8.1.2).
-An endpoint that acknowledges packets it has not received might cause a congestion controller to permit sending at rates beyond what the network supports. An endpoint MAY skip packet numbers when sending packets to detect this behavior. An endpoint can then immediately close the connection with a connection error of type PROTOCOL_VIOLATION (see Section 10.3).
-The attacks commonly known as Slowloris [SLOWLORIS] try to keep many connections to the target endpoint open and hold them open as long as possible. These attacks can be executed against a QUIC endpoint by generating the minimum amount of activity necessary to avoid being closed for inactivity. This might involve sending small amounts of data, gradually opening flow control windows in order to control the sender rate, or manufacturing ACK frames that simulate a high loss rate.
-QUIC deployments SHOULD provide mitigations for the Slowloris attacks, such as increasing the maximum number of clients the server will allow, limiting the number of connections a single IP address is allowed to make, imposing restrictions on the minimum transfer speed a connection is allowed to have, and restricting the length of time an endpoint is allowed to stay connected.
-An adversarial sender might intentionally send fragments of stream data in order to cause disproportionate receive buffer memory commitment and/or creation of a large and inefficient data structure.
-An adversarial receiver might intentionally not acknowledge packets containing stream data in order to force the sender to store the unacknowledged stream data for retransmission.
-The attack on receivers is mitigated if flow control windows correspond to available memory. However, some receivers will over-commit memory and advertise flow control offsets in the aggregate that exceed actual available memory. The over-commitment strategy can lead to better performance when endpoints are well behaved, but renders endpoints vulnerable to the stream fragmentation attack.
-QUIC deployments SHOULD provide mitigations against stream fragmentation attacks. Mitigations could consist of avoiding over-committing memory, limiting the size of tracking data structures, delaying reassembly of STREAM frames, implementing heuristics based on the age and duration of reassembly holes, or some combination.
-An adversarial endpoint can open lots of streams, exhausting state on an endpoint. The adversarial endpoint could repeat the process on a large number of connections, in a manner similar to SYN flooding attacks in TCP.
-Normally, clients will open streams sequentially, as explained in Section 2.1. However, when several streams are initiated at short intervals, loss or reordering may cause STREAM frames that open streams to be received out of sequence. On receiving a higher-numbered stream ID, a receiver is required to open all intervening streams of the same type (see Section 3.2). Thus, on a new connection, opening stream 4000000 opens 1 million and 1 client-initiated bidirectional streams.
-The number of active streams is limited by the initial_max_streams_bidi and initial_max_streams_uni transport parameters, as explained in Section 4.5. If chosen judiciously, these limits mitigate the effect of the stream commitment attack. However, setting the limit too low could affect performance when applications expect to open large number of streams.
-QUIC and TLS both contain messages that have legitimate uses in some contexts, but that can be abused to cause a peer to expend processing resources without having any observable impact on the state of the connection.
-Messages can also be used to change and revert state in small or inconsequential ways, such as by sending small increments to flow control limits.
-If processing costs are disproportionately large in comparison to bandwidth consumption or effect on state, then this could allow a malicious peer to exhaust processing capacity.
-While there are legitimate uses for all messages, implementations SHOULD track cost of processing relative to progress and treat excessive quantities of any non-productive packets as indicative of an attack. Endpoints MAY respond to this condition with a connection error, or by dropping packets.
-An on-path attacker could manipulate the value of ECN codepoints in the IP header to influence the sender’s rate. [RFC3168] discusses manipulations and their effects in more detail.
-An on-the-side attacker can duplicate and send packets with modified ECN codepoints to affect the sender’s rate. If duplicate packets are discarded by a receiver, an off-path attacker will need to race the duplicate packet against the original to be successful in this attack. Therefore, QUIC endpoints ignore the ECN codepoint field on an IP packet unless at least one QUIC packet in that IP packet is successfully processed; see Section 13.4.
-Stateless resets create a possible denial of service attack analogous to a TCP reset injection. This attack is possible if an attacker is able to cause a stateless reset token to be generated for a connection with a selected connection ID. An attacker that can cause this token to be generated can reset an active connection with the same connection ID.
-If a packet can be routed to different instances that share a static key, for example by changing an IP address or port, then an attacker can cause the server to send a stateless reset. To defend against this style of denial service, endpoints that share a static key for stateless reset (see Section 10.4.2) MUST be arranged so that packets with a given connection ID always arrive at an instance that has connection state, unless that connection is no longer active.
-In the case of a cluster that uses dynamic load balancing, it’s possible that a change in load balancer configuration could happen while an active instance retains connection state; even if an instance retains connection state, the change in routing and resulting stateless reset will result in the connection being terminated. If there is no chance in the packet being routed to the correct instance, it is better to send a stateless reset than wait for connections to time out. However, this is acceptable only if the routing cannot be influenced by an attacker.
-This document defines QUIC Version Negotiation packets Section 6, which can be used to negotiate the QUIC version used between two endpoints. However, this document does not specify how this negotiation will be performed between this version and subsequent future versions. In particular, Version Negotiation packets do not contain any mechanism to prevent version downgrade attacks. Future versions of QUIC that use Version Negotiation packets MUST define a mechanism that is robust against version downgrade attacks.
-Deployments should limit the ability of an attacker to target a new connection to a particular server instance. This means that client-controlled fields, such as the initial Destination Connection ID used on Initial and 0-RTT packets SHOULD NOT be used by themselves to make routing decisions. Ideally, routing decisions are made independently of client-selected values; a Source Connection ID can be selected to route later packets to the same server.
-IANA [SHALL add/has added] a registry for “QUIC Transport Parameters” under a “QUIC Protocol” heading.
-The “QUIC Transport Parameters” registry governs a 16-bit space. This space is split into two spaces that are governed by different policies. Values with the first byte in the range 0x00 to 0xfe (in hexadecimal) are assigned via the Specification Required policy [RFC8126]. Values with the first byte 0xff are reserved for Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -See Section 22.4 for details of registering new error codes.¶
+In defining these error codes, several principles are applied. Error conditions +that might require specific action on the part of a recipient are given unique +codes. Errors that represent common conditions are given specific codes. +Absent either of these conditions, error codes are used to identify a general +function of the stack, like flow control or transport parameter handling. +Finally, generic errors are provided for conditions where implementations are +unable or unwilling to use more specific codes.¶
+Application protocol error codes are 62-bit unsigned integers, but the +management of application error codes is left to application protocols. +Application protocol error codes are used for the RESET_STREAM frame +(Section 19.4), the STOP_SENDING frame (Section 19.5), and +the CONNECTION_CLOSE frame with a type of 0x1d (Section 19.19).¶
+As an encrypted and authenticated transport QUIC provides a range of protections +against denial of service. Once the cryptographic handshake is complete, QUIC +endpoints discard most packets that are not authenticated, greatly limiting the +ability of an attacker to interfere with existing connections.¶
+Once a connection is established QUIC endpoints might accept some +unauthenticated ICMP packets (see Section 14.2), but the use of these packets +is extremely limited. The only other type of packet that an endpoint might +accept is a stateless reset (Section 10.4) which relies on the token +being kept secret until it is used.¶
+During the creation of a connection, QUIC only provides protection against +attack from off the network path. All QUIC packets contain proof that the +recipient saw a preceding packet from its peer.¶
+The first mechanism used is the source and destination connection IDs, which are +required to match those set by a peer. Except for an Initial and stateless +reset packets, an endpoint only accepts packets that include a destination +connection that matches a connection ID the endpoint previously chose. This is +the only protection offered for Version Negotiation packets.¶
+The destination connection ID in an Initial packet is selected by a client to be +unpredictable, which serves an additional purpose. The packets that carry the +cryptographic handshake are protected with a key that is derived from this +connection ID and salt specific to the QUIC version. This allows endpoints to +use the same process for authenticating packets that they receive as they use +after the cryptographic handshake completes. Packets that cannot be +authenticated are discarded. Protecting packets in this fashion provides a +strong assurance that the sender of the packet saw the Initial packet and +understood it.¶
+These protections are not intended to be effective against an attacker that is +able to receive QUIC packets prior to the connection being established. Such an +attacker can potentially send packets that will be accepted by QUIC endpoints. +This version of QUIC attempts to detect this sort of attack, but it expects that +endpoints will fail to establish a connection rather than recovering. For the +most part, the cryptographic handshake protocol [QUIC-TLS] is responsible for +detecting tampering during the handshake.¶
+Endpoints are permitted to use other methods to detect and attempt to recover +from interference with the handshake. Invalid packets may be identified and +discarded using other methods, but no specific method is mandated in this +document.¶
+An attacker might be able to receive an address validation token +(Section 8) from a server and then release the IP address it used +to acquire that token. At a later time, the attacker may initiate a 0-RTT +connection with a server by spoofing this same address, which might now address +a different (victim) endpoint. The attacker can thus potentially cause the +server to send an initial congestion window's worth of data towards the victim.¶
+Servers SHOULD provide mitigations for this attack by limiting the usage and +lifetime of address validation tokens (see Section 8.1.3).¶
+An endpoint that acknowledges packets it has not received might cause a +congestion controller to permit sending at rates beyond what the network +supports. An endpoint MAY skip packet numbers when sending packets to detect +this behavior. An endpoint can then immediately close the connection with a +connection error of type PROTOCOL_VIOLATION (see Section 10.3).¶
+The attacks commonly known as Slowloris [SLOWLORIS] try to keep many +connections to the target endpoint open and hold them open as long as possible. +These attacks can be executed against a QUIC endpoint by generating the minimum +amount of activity necessary to avoid being closed for inactivity. This might +involve sending small amounts of data, gradually opening flow control windows in +order to control the sender rate, or manufacturing ACK frames that simulate a +high loss rate.¶
+QUIC deployments SHOULD provide mitigations for the Slowloris attacks, such as +increasing the maximum number of clients the server will allow, limiting the +number of connections a single IP address is allowed to make, imposing +restrictions on the minimum transfer speed a connection is allowed to have, and +restricting the length of time an endpoint is allowed to stay connected.¶
+An adversarial sender might intentionally send fragments of stream data in +order to cause disproportionate receive buffer memory commitment and/or +creation of a large and inefficient data structure.¶
+An adversarial receiver might intentionally not acknowledge packets +containing stream data in order to force the sender to store the +unacknowledged stream data for retransmission.¶
+The attack on receivers is mitigated if flow control windows correspond to +available memory. However, some receivers will over-commit memory and +advertise flow control offsets in the aggregate that exceed actual available +memory. The over-commitment strategy can lead to better performance when +endpoints are well behaved, but renders endpoints vulnerable to the stream +fragmentation attack.¶
+QUIC deployments SHOULD provide mitigations against stream fragmentation +attacks. Mitigations could consist of avoiding over-committing memory, +limiting the size of tracking data structures, delaying reassembly +of STREAM frames, implementing heuristics based on the age and +duration of reassembly holes, or some combination.¶
+An adversarial endpoint can open lots of streams, exhausting state on an +endpoint. The adversarial endpoint could repeat the process on a large number +of connections, in a manner similar to SYN flooding attacks in TCP.¶
+Normally, clients will open streams sequentially, as explained in Section 2.1. +However, when several streams are initiated at short intervals, loss or +reordering may cause STREAM frames that open streams to be received out of +sequence. On receiving a higher-numbered stream ID, a receiver is required to +open all intervening streams of the same type (see Section 3.2). +Thus, on a new connection, opening stream 4000000 opens 1 million and 1 +client-initiated bidirectional streams.¶
+The number of active streams is limited by the initial_max_streams_bidi and +initial_max_streams_uni transport parameters, as explained in +Section 4.5. If chosen judiciously, these limits mitigate the +effect of the stream commitment attack. However, setting the limit too low +could affect performance when applications expect to open large number of +streams.¶
+QUIC and TLS both contain messages that have legitimate uses in some contexts, +but that can be abused to cause a peer to expend processing resources without +having any observable impact on the state of the connection.¶
+Messages can also be used to change and revert state in small or inconsequential +ways, such as by sending small increments to flow control limits.¶
+If processing costs are disproportionately large in comparison to bandwidth +consumption or effect on state, then this could allow a malicious peer to +exhaust processing capacity.¶
+While there are legitimate uses for all messages, implementations SHOULD track +cost of processing relative to progress and treat excessive quantities of any +non-productive packets as indicative of an attack. Endpoints MAY respond to +this condition with a connection error, or by dropping packets.¶
+An on-path attacker could manipulate the value of ECN codepoints in the IP +header to influence the sender's rate. [RFC3168] discusses manipulations and +their effects in more detail.¶
+An on-the-side attacker can duplicate and send packets with modified ECN +codepoints to affect the sender's rate. If duplicate packets are discarded by a +receiver, an off-path attacker will need to race the duplicate packet against +the original to be successful in this attack. Therefore, QUIC endpoints ignore +the ECN codepoint field on an IP packet unless at least one QUIC packet in that +IP packet is successfully processed; see Section 13.4.¶
+Stateless resets create a possible denial of service attack analogous to a TCP +reset injection. This attack is possible if an attacker is able to cause a +stateless reset token to be generated for a connection with a selected +connection ID. An attacker that can cause this token to be generated can reset +an active connection with the same connection ID.¶
+If a packet can be routed to different instances that share a static key, for +example by changing an IP address or port, then an attacker can cause the server +to send a stateless reset. To defend against this style of denial service, +endpoints that share a static key for stateless reset (see Section 10.4.2) MUST +be arranged so that packets with a given connection ID always arrive at an +instance that has connection state, unless that connection is no longer active.¶
+In the case of a cluster that uses dynamic load balancing, it's possible that a +change in load balancer configuration could happen while an active instance +retains connection state; even if an instance retains connection state, the +change in routing and resulting stateless reset will result in the connection +being terminated. If there is no chance in the packet being routed to the +correct instance, it is better to send a stateless reset than wait for +connections to time out. However, this is acceptable only if the routing cannot +be influenced by an attacker.¶
+This document defines QUIC Version Negotiation packets in +Section 6, which can be used to negotiate the QUIC version used +between two endpoints. However, this document does not specify how this +negotiation will be performed between this version and subsequent future +versions. In particular, Version Negotiation packets do not contain any +mechanism to prevent version downgrade attacks. Future versions of QUIC that +use Version Negotiation packets MUST define a mechanism that is robust against +version downgrade attacks.¶
+Deployments should limit the ability of an attacker to target a new connection +to a particular server instance. This means that client-controlled fields, such +as the initial Destination Connection ID used on Initial and 0-RTT packets +SHOULD NOT be used by themselves to make routing decisions. Ideally, routing +decisions are made independently of client-selected values; a Source Connection +ID can be selected to route later packets to the same server.¶
+This document establishes several registries for the management of codepoints in +QUIC. These registries operate on a common set of policies as defined in +Section 22.1.¶
+All QUIC registries allow for both provisional and permanent registration of +codepoints. This section documents policies that are common to these +registries.¶
+Provisional registration of codepoints are intended to allow for private use and +experimentation with extensions to QUIC. Provisional registrations only require +the inclusion of the codepoint value and contact information. However, +provisional registrations could be reclaimed and reassigned for another purpose.¶
+Provisional registrations require Expert Review, as defined in Section 4.5 of +[RFC8126]. Designated expert(s) are advised that only registrations for an +excessive proportion of remaining codepoint space or the very first unassigned +value (see Section 22.1.2) can be rejected.¶
+Provisional registrations will include a date field that indicates when the +registration was last updated. A request to update the date on any provisional +registration can be made without review from the designated expert(s).¶
+All QUIC registries include the following fields to support provisional +registration:¶
+The nominated expert(s) verify that a specification exists and is readily accessible. Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are shown in Table 6.
- - -| Value | -Parameter Name | -Specification | -
|---|---|---|
| 0x0000 | -original_connection_id | -Section 18.2 | -
| 0x0001 | -idle_timeout | -Section 18.2 | -
| 0x0002 | -stateless_reset_token | -Section 18.2 | -
| 0x0003 | -max_packet_size | -Section 18.2 | -
| 0x0004 | -initial_max_data | -Section 18.2 | -
| 0x0005 | -initial_max_stream_data_bidi_local | -Section 18.2 | -
| 0x0006 | -initial_max_stream_data_bidi_remote | -Section 18.2 | -
| 0x0007 | -initial_max_stream_data_uni | -Section 18.2 | -
| 0x0008 | -initial_max_streams_bidi | -Section 18.2 | -
| 0x0009 | -initial_max_streams_uni | -Section 18.2 | -
| 0x000a | -ack_delay_exponent | -Section 18.2 | -
| 0x000b | -max_ack_delay | -Section 18.2 | -
| 0x000c | -disable_active_migration | -Section 18.2 | -
| 0x000d | -preferred_address | -Section 18.2 | -
| 0x000e | -active_connection_id_limit | -Section 18.2 | -
Additionally, each value of the format 31 * N + 27 for integer values of N (that is, 27, 58, 89, …) MUST NOT be assigned by IANA.
-IANA [SHALL add/has added] a registry for “QUIC Frame Types” under a “QUIC Protocol” heading.
-The “QUIC Frame Types” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -Provisional registrations MAY omit the Specification and Notes fields, plus any +additional fields that might be required for a permanent registration. The Date +field is not required as part of requesting a registration as it is set to the +date the registration is created or updated.¶
+New uses of codepoints from QUIC registries SHOULD use a randomly selected +codepoint that excludes both existing allocations and the first unallocated +codepoint in the selected space. Requests for multiple codepoints MAY use a +contiguous range. This minimizes the risk that differing semantics are +attributed to the same codepoint by different implementations. Use of the first +codepoint in a range is intended for use by specifications that are developed +through the standards process [STD] and its allocation MUST be +negotiated with IANA before use.¶
+For codepoints that are encoded in variable-length integers +(Section 16), such as frame types, codepoints that encode to four or +eight bytes (that is, values 2^14 and above) SHOULD be used unless the usage is +especially sensitive to having a longer encoding.¶
+Applications to register codepoints in QUIC registries MAY include a codepoint +as part of the registration. IANA MUST allocate the selected codepoint unless +that codepoint is already assigned or the codepoint is the first unallocated +codepoint in the registry.¶
+A request might be made to remove an unused provisional registration from the +registry to reclaim space in a registry, or portion of the registry (such as the +64-16383 range for codepoints that use variable-length encodings). This SHOULD +be done only for the codepoints with the earliest recorded date and entries that +have been updated less than a year prior SHOULD NOT be reclaimed.¶
+A request to remove a codepoint MUST be reviewed by the designated expert(s). +The expert(s) MUST attempt to determine whether the codepoint is still in use. +Experts are advised to contact the listed contacts for the registration, plus as +wide a set of protocol implementers as possible in order to determine whether +any use of the codepoint is known. The expert(s) are advised to allow at least +four weeks for responses.¶
+If any use of the codepoints is identified by this search or a request to update +the registration is made, the codepoint MUST NOT be reclaimed. Instead, the +date on the registration is updated. A note might be added for the registration +recording relevant information that was learned.¶
+If no use of the codepoint was identified and no request was made to update the +registration, the codepoint MAY be removed from the registry.¶
+This process also applies to requests to change a provisional registration into +a permanent registration, except that the goal is not to determine whether there +is no use of the codepoint, but to determine that the registration is an +accurate representation of any deployed usage.¶
+Permanent registrations in QUIC registries use the Specification Required policy +[RFC8126], unless otherwise specified. The designated expert(s) verify that +a specification exists and is readily accessible. Expert(s) are encouraged to +be biased towards approving registrations unless they are abusive, frivolous, or +actively harmful (not merely aesthetically displeasing, or architecturally +dubious). The creation of a registry MAY specify additional constraints on +permanent registrations.¶
+The creation of a registries MAY identify a range of codepoints where +registrations are governed by a different registration policy. For instance, +the registries for 62-bit codepoints in this document have stricter policies for +codepoints in the range from 0 to 63.¶
+Any stricter requirements for permanent registrations do not prevent provisional +registrations for affected codepoints. For instance, a provisional registration +for a frame type Section 22.3 of 61 could be requested.¶
+All registrations made by Standards Track publications MUST be permanent.¶
+All registrations in this document are assigned a permanent status and list as +contact both the IESG (ietf@ietf.org) and the QUIC working group +(quic@ietf.org).¶
+IANA [SHALL add/has added] a registry for "QUIC Transport Parameters" under a +"QUIC" heading.¶
+The "QUIC Transport Parameters" registry governs a 16-bit space. This registry +follows the registration policy from Section 22.1. Permanent registrations +in this registry are assigned using the Specification Required policy +[RFC8126].¶
+In addition to the fields in Section 22.1.1, permanent registrations in +this registry MUST include the following fields:¶
+The nominated expert(s) verify that a specification exists and is readily accessible. Specifications for new registrations need to describe the means by which an endpoint might determine that it can send the identified type of frame. An accompanying transport parameter registration (see Section 22.1) is expected for most registrations. The specification needs to describe the format and assigned semantics of any fields in the frame.
-Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are tabulated in Table 3.
-IANA [SHALL add/has added] a registry for “QUIC Transport Error Codes” under a “QUIC Protocol” heading.
-The “QUIC Transport Error Codes” registry governs a 62-bit space. This space is split into three spaces that are governed by different policies. Values between 0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review policies [RFC8126]. Values from 0x40 to 0x3fff operate on the Specification Required policy [RFC8126]. All other values are assigned to Private Use [RFC8126].
-Registrations MUST include the following fields:
- - -The initial contents of this registry are shown in Table 6.¶
+| Value | +Parameter Name | +Specification | +
|---|---|---|
| 0x0000 | +original_connection_id | ++ Section 18.2 + | +
| 0x0001 | +idle_timeout | ++ Section 18.2 + | +
| 0x0002 | +stateless_reset_token | ++ Section 18.2 + | +
| 0x0003 | +max_packet_size | ++ Section 18.2 + | +
| 0x0004 | +initial_max_data | ++ Section 18.2 + | +
| 0x0005 | +initial_max_stream_data_bidi_local | ++ Section 18.2 + | +
| 0x0006 | +initial_max_stream_data_bidi_remote | ++ Section 18.2 + | +
| 0x0007 | +initial_max_stream_data_uni | ++ Section 18.2 + | +
| 0x0008 | +initial_max_streams_bidi | ++ Section 18.2 + | +
| 0x0009 | +initial_max_streams_uni | ++ Section 18.2 + | +
| 0x000a | +ack_delay_exponent | ++ Section 18.2 + | +
| 0x000b | +max_ack_delay | ++ Section 18.2 + | +
| 0x000c | +disable_active_migration | ++ Section 18.2 + | +
| 0x000d | +preferred_address | ++ Section 18.2 + | +
| 0x000e | +active_connection_id_limit | ++ Section 18.2 + | +
Additionally, each value of the format 31 * N + 27 for integer values of N
+(that is, 27, 58, 89, ...) are reserved and MUST NOT be assigned by IANA.¶
IANA [SHALL add/has added] a registry for "QUIC Frame Types" under a +"QUIC" heading.¶
+The "QUIC Frame Types" registry governs a 62-bit space. This registry follows +the registration policy from Section 22.1. Permanent registrations in this +registry are assigned using the Specification Required policy [RFC8126], +except for values between 0x00 and 0x3f (in hexadecimal; inclusive), which are +assigned using Standards Action or IESG Approval as defined in Section 4.9 and +4.10 of [RFC8126].¶
+In addition to the fields in Section 22.1.1, permanent registrations in +this registry MUST include the following fields:¶
+The nominated expert(s) verify that a specification exists and is readily accessible. Expert(s) are encouraged to be biased towards approving registrations unless they are abusive, frivolous, or actively harmful (not merely aesthetically displeasing, or architecturally dubious).
-The initial contents of this registry are shown in Table 7.
- - -| Value | -Error | -Description | -Specification | -
|---|---|---|---|
| 0x0 | -NO_ERROR | -No error | -Section 20 | -
| 0x1 | -INTERNAL_ERROR | -Implementation error | -Section 20 | -
| 0x2 | -SERVER_BUSY | -Server currently busy | -Section 20 | -
| 0x3 | -FLOW_CONTROL_ERROR | -Flow control error | -Section 20 | -
| 0x4 | -STREAM_LIMIT_ERROR | -Too many streams opened | -Section 20 | -
| 0x5 | -STREAM_STATE_ERROR | -Frame received in invalid stream state | -Section 20 | -
| 0x6 | -FINAL_SIZE_ERROR | -Change to final size | -Section 20 | -
| 0x7 | -FRAME_ENCODING_ERROR | -Frame encoding error | -Section 20 | -
| 0x8 | -TRANSPORT_PARAMETER_ERROR | -Error in transport parameters | -Section 20 | -
| 0xA | -PROTOCOL_VIOLATION | -Generic protocol violation | -Section 20 | -
| 0xD | -CRYPTO_BUFFER_EXCEEDED | -CRYPTO data buffer overflowed | -Section 20 | -
| [DPLPMTUD] | --Fairhurst, G., Jones, T., Tuexen, M., Ruengeler, I. and T. Voelker, "Packetization Layer Path MTU Discovery for Datagram Transports", Internet-Draft draft-ietf-tsvwg-datagram-plpmtud-08, June 2019. | -
| [QUIC-RECOVERY] | --Iyengar, J. and I. Swett, "QUIC Loss Detection and Congestion Control", Internet-Draft draft-ietf-quic-recovery, November 2019. | -
| [QUIC-TLS] | --Thomson, M. and S. Turner, "Using Transport Layer Security (TLS) to Secure QUIC", Internet-Draft draft-ietf-quic-tls, November 2019. | -
| [RFC1191] | --Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, DOI 10.17487/RFC1191, November 1990. | -
| [RFC2119] | --Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. | -
| [RFC3168] | --Ramakrishnan, K., Floyd, S. and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, DOI 10.17487/RFC3168, September 2001. | -
| [RFC3629] | --Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November 2003. | -
| [RFC4086] | --Eastlake 3rd, D., Schiller, J. and S. Crocker, "Randomness Requirements for Security", BCP 106, RFC 4086, DOI 10.17487/RFC4086, June 2005. | -
| [RFC5116] | --McGrew, D., "An Interface and Algorithms for Authenticated Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008. | -
| [RFC6437] | --Amante, S., Carpenter, B., Jiang, S. and J. Rajahalme, "IPv6 Flow Label Specification", RFC 6437, DOI 10.17487/RFC6437, November 2011. | -
| [RFC8085] | --Eggert, L., Fairhurst, G. and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, March 2017. | -
| [RFC8126] | --Cotton, M., Leiba, B. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 8126, DOI 10.17487/RFC8126, June 2017. | -
| [RFC8174] | --Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017. | -
| [RFC8201] | --McCann, J., Deering, S., Mogul, J. and R. Hinden, "Path MTU Discovery for IP version 6", STD 87, RFC 8201, DOI 10.17487/RFC8201, July 2017. | -
| [RFC8311] | --Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018. | -
| [TLS13] | --Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018. | -
| [EARLY-DESIGN] | --Roskind, J., "QUIC: Multiplexed Transport Over UDP", December 2013. | -
| [HTTP2] | --Belshe, M., Peon, R. and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015. | -
| [QUIC-INVARIANTS] | --Thomson, M., "Version-Independent Properties of QUIC", Internet-Draft draft-ietf-quic-invariants, November 2019. | -
| [QUIC-MANAGEABILITY] | --Kuehlewind, M. and B. Trammell, "Manageability of the QUIC Transport Protocol", Internet-Draft draft-ietf-quic-manageability-05, July 2019. | -
| [RFC1812] | --Baker, F., "Requirements for IP Version 4 Routers", RFC 1812, DOI 10.17487/RFC1812, June 1995. | -
| [RFC2018] | --Mathis, M., Mahdavi, J., Floyd, S. and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, DOI 10.17487/RFC2018, October 1996. | -
| [RFC2104] | --Krawczyk, H., Bellare, M. and R. Canetti, "HMAC: Keyed-Hashing for Message Authentication", RFC 2104, DOI 10.17487/RFC2104, February 1997. | -
| [RFC2360] | --Scott, G., "Guide for Internet Standards Writers", BCP 22, RFC 2360, DOI 10.17487/RFC2360, June 1998. | -
| [RFC4303] | --Kent, S., "IP Encapsulating Security Payload (ESP)", RFC 4303, DOI 10.17487/RFC4303, December 2005. | -
| [RFC4443] | --Conta, A., Deering, S. and M. Gupta, "Internet Control Message Protocol (ICMPv6) for the Internet Protocol Version 6 (IPv6) Specification", STD 89, RFC 4443, DOI 10.17487/RFC4443, March 2006. | -
| [RFC4787] | --Audet, F. and C. Jennings, "Network Address Translation (NAT) Behavioral Requirements for Unicast UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January 2007. | -
| [RFC5681] | --Allman, M., Paxson, V. and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009. | -
| [RFC5869] | --Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand Key Derivation Function (HKDF)", RFC 5869, DOI 10.17487/RFC5869, May 2010. | -
| [RFC7301] | --Friedl, S., Popov, A., Langley, A. and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014. | -
| [RFC8200] | --Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", STD 86, RFC 8200, DOI 10.17487/RFC8200, July 2017. | -
| [SLOWLORIS] | --RSnake Hansen, R., "Welcome to Slowloris...", June 2009. | -
The following pseudo-code shows how an implementation can decode packet numbers after header protection has been removed.
+In addition to the advice in Section 22.1, specifications for new permanent +registrations SHOULD describe the means by which an endpoint might determine +that it can send the identified type of frame. An accompanying transport +parameter registration (see Section 22.2) is expected for most +registrations. Specifications for permanent registrations also needs to +describe the format and assigned semantics of any fields in the frame.¶
+The initial contents of this registry are tabulated in Table 3.¶
+IANA [SHALL add/has added] a registry for "QUIC Transport Error Codes" under a +"QUIC" heading.¶
+The "QUIC Transport Error Codes" registry governs a 62-bit space. This space is +split into three spaces that are governed by different policies. Permanent +registrations in this registry are assigned using the Specification Required +policy [RFC8126], except for values between 0x00 and 0x3f (in hexadecimal; +inclusive), which are assigned using Standards Action or IESG Approval as +defined in Section 4.9 and 4.10 of [RFC8126].¶
+In addition to the fields in Section 22.1.1, permanent registrations in +this registry MUST include the following fields:¶
+The initial contents of this registry are shown in Table 7.¶
+| Value | +Error | +Description | +Specification | +
|---|---|---|---|
| 0x0 | +NO_ERROR | +No error | ++ Section 20 + | +
| 0x1 | +INTERNAL_ERROR | +Implementation error | ++ Section 20 + | +
| 0x2 | +SERVER_BUSY | +Server currently busy | ++ Section 20 + | +
| 0x3 | +FLOW_CONTROL_ERROR | +Flow control error | ++ Section 20 + | +
| 0x4 | +STREAM_LIMIT_ERROR | +Too many streams opened | ++ Section 20 + | +
| 0x5 | +STREAM_STATE_ERROR | +Frame received in invalid stream state | ++ Section 20 + | +
| 0x6 | +FINAL_SIZE_ERROR | +Change to final size | ++ Section 20 + | +
| 0x7 | +FRAME_ENCODING_ERROR | +Frame encoding error | ++ Section 20 + | +
| 0x8 | +TRANSPORT_PARAMETER_ERROR | +Error in transport parameters | ++ Section 20 + | +
| 0x9 | +CONNECTION_ID_LIMIT_ERROR | +Too many connection IDs received | ++ Section 20 + | +
| 0xA | +PROTOCOL_VIOLATION | +Generic protocol violation | ++ Section 20 + | +
| 0xB | +INVALID_TOKEN | +Invalid Token Received | ++ Section 20 + | +
| 0xD | +CRYPTO_BUFFER_EXCEEDED | +CRYPTO data buffer overflowed | ++ Section 20 + | +
The pseudo-code in Figure 37 shows how an implementation can decode +packet numbers after header protection has been removed.¶
+
DecodePacketNumber(largest_pn, truncated_pn, pn_nbits):
expected_pn = largest_pn + 1
@@ -3987,532 +8702,1008 @@
return candidate_pn - pn_win
return candidate_pn
-Issue and pull request numbers are listed with a leading octothorp.
-Issue and pull request numbers are listed with a leading octothorp.¶
+Stateless reset changes (#2152, #2993)¶
+Rework the first byte (#2006)¶
+Substantial editorial reorganization; no technical changes.
-Substantial editorial reorganization; no technical changes.¶
+Changes to integration of the TLS handshake (#829, #1018, #1094, #1165, #1190, +#1233, #1242, #1252, #1450, #1458)¶
+Streams are split into unidirectional and bidirectional (#643, #656, #720, +#872, #175, #885)¶
Improvements to connection close¶
Split some frames into separate connection- and stream- level frames +(#443)¶
Transport parameters for 0-RTT are retained from a previous connection (#405, +#513, #512)¶
Special thanks are due to the following for helping shape pre-IETF QUIC and its deployment: Chris Bentzel, Misha Efimov, Roberto Peon, Alistair Riddoch, Siddharth Vijayakrishnan, and Assar Westerlund.
-This document has benefited immensely from various private discussions and public ones on the quic@ietf.org and proto-quic@chromium.org mailing lists. Our thanks to all.
-The original authors of this specification were Ryan Hamilton, Jana Iyengar, Ian Swett, and Alyssa Wilk.
-The original design and rationale behind this protocol draw significantly from work by Jim Roskind [EARLY-DESIGN]. In alphabetical order, the contributors to the pre-IETF QUIC project at Google are: Britt Cyr, Jeremy Dorfman, Ryan Hamilton, Jana Iyengar, Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley, Jim Roskind, Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti, Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale Worley, Fan Yang, Dan Zhang, Daniel Ziegler.
-Special thanks are due to the following for helping shape pre-IETF QUIC and its +deployment: Chris Bentzel, Misha Efimov, Roberto Peon, Alistair Riddoch, +Siddharth Vijayakrishnan, and Assar Westerlund.¶
+This document has benefited immensely from various private discussions and +public ones on the quic@ietf.org and proto-quic@chromium.org mailing lists. Our +thanks to all.¶
+The original authors of this specification were Ryan Hamilton, Jana Iyengar, Ian +Swett, and Alyssa Wilk.¶
+The original design and rationale behind this protocol draw significantly from +work by Jim Roskind [EARLY-DESIGN]. In alphabetical order, the contributors to +the pre-IETF QUIC project at Google are: Britt Cyr, Jeremy Dorfman, Ryan +Hamilton, Jana Iyengar, Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley, +Jim Roskind, Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti, +Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale Worley, Fan +Yang, Dan Zhang, Daniel Ziegler.¶
+