draft-ietf-quic-transport.txt

QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                         M. Thomson, Ed.
Expires: August 18, 2019                                         Mozilla
                                                       February 14, 2019


           QUIC: A UDP-Based Multiplexed and Secure Transport
                    draft-ietf-quic-transport-latest

Abstract

   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.

Note to Readers

   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>.

Status of This Memo

   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 August 18, 2019.

Copyright Notice

   Copyright (c) 2019 IETF Trust and the persons identified as the
   document authors.  All rights reserved.




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   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
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   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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   5
     1.1.  Document Structure  . . . . . . . . . . . . . . . . . . .   6
     1.2.  Terms and Definitions . . . . . . . . . . . . . . . . . .   7
     1.3.  Notational Conventions  . . . . . . . . . . . . . . . . .   8
   2.  Streams . . . . . . . . . . . . . . . . . . . . . . . . . . .   8
     2.1.  Stream Types and Identifiers  . . . . . . . . . . . . . .   9
     2.2.  Sending and Receiving Data  . . . . . . . . . . . . . . .  10
     2.3.  Stream Prioritization . . . . . . . . . . . . . . . . . .  10
   3.  Stream States . . . . . . . . . . . . . . . . . . . . . . . .  11
     3.1.  Sending Stream States . . . . . . . . . . . . . . . . . .  11
     3.2.  Receiving Stream States . . . . . . . . . . . . . . . . .  13
     3.3.  Permitted Frame Types . . . . . . . . . . . . . . . . . .  16
     3.4.  Bidirectional Stream States . . . . . . . . . . . . . . .  16
     3.5.  Solicited State Transitions . . . . . . . . . . . . . . .  18
   4.  Flow Control  . . . . . . . . . . . . . . . . . . . . . . . .  19
     4.1.  Data Flow Control . . . . . . . . . . . . . . . . . . . .  19
     4.2.  Flow Credit Increments  . . . . . . . . . . . . . . . . .  20
     4.3.  Handling Stream Cancellation  . . . . . . . . . . . . . .  21
     4.4.  Stream Final Size . . . . . . . . . . . . . . . . . . . .  22
     4.5.  Controlling Concurrency . . . . . . . . . . . . . . . . .  22
   5.  Connections . . . . . . . . . . . . . . . . . . . . . . . . .  23
     5.1.  Connection ID . . . . . . . . . . . . . . . . . . . . . .  23
       5.1.1.  Issuing Connection IDs  . . . . . . . . . . . . . . .  24
       5.1.2.  Consuming and Retiring Connection IDs . . . . . . . .  25
     5.2.  Matching Packets to Connections . . . . . . . . . . . . .  25
       5.2.1.  Client Packet Handling  . . . . . . . . . . . . . . .  26
       5.2.2.  Server Packet Handling  . . . . . . . . . . . . . . .  27
     5.3.  Life of a QUIC Connection . . . . . . . . . . . . . . . .  27
   6.  Version Negotiation . . . . . . . . . . . . . . . . . . . . .  28
     6.1.  Sending Version Negotiation Packets . . . . . . . . . . .  28
     6.2.  Handling Version Negotiation Packets  . . . . . . . . . .  28
       6.2.1.  Version Negotiation Between Draft Versions  . . . . .  29
     6.3.  Using Reserved Versions . . . . . . . . . . . . . . . . .  29
   7.  Cryptographic and Transport Handshake . . . . . . . . . . . .  29
     7.1.  Example Handshake Flows . . . . . . . . . . . . . . . . .  31
     7.2.  Negotiating Connection IDs  . . . . . . . . . . . . . . .  32



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     7.3.  Transport Parameters  . . . . . . . . . . . . . . . . . .  33
       7.3.1.  Values of Transport Parameters for 0-RTT  . . . . . .  34
       7.3.2.  New Transport Parameters  . . . . . . . . . . . . . .  35
   8.  Address Validation  . . . . . . . . . . . . . . . . . . . . .  35
     8.1.  Address Validation During Connection Establishment  . . .  36
       8.1.1.  Address Validation using Retry Packets  . . . . . . .  37
       8.1.2.  Address Validation for Future Connections . . . . . .  37
       8.1.3.  Address Validation Token Integrity  . . . . . . . . .  39
     8.2.  Path Validation . . . . . . . . . . . . . . . . . . . . .  40
     8.3.  Initiating Path Validation  . . . . . . . . . . . . . . .  40
     8.4.  Path Validation Responses . . . . . . . . . . . . . . . .  41
     8.5.  Successful Path Validation  . . . . . . . . . . . . . . .  41
     8.6.  Failed Path Validation  . . . . . . . . . . . . . . . . .  41
   9.  Connection Migration  . . . . . . . . . . . . . . . . . . . .  42
     9.1.  Probing a New Path  . . . . . . . . . . . . . . . . . . .  43
     9.2.  Initiating Connection Migration . . . . . . . . . . . . .  43
     9.3.  Responding to Connection Migration  . . . . . . . . . . .  44
       9.3.1.  Peer Address Spoofing . . . . . . . . . . . . . . . .  44
       9.3.2.  On-Path Address Spoofing  . . . . . . . . . . . . . .  45
       9.3.3.  Off-Path Packet Forwarding  . . . . . . . . . . . . .  45
     9.4.  Loss Detection and Congestion Control . . . . . . . . . .  47
     9.5.  Privacy Implications of Connection Migration  . . . . . .  47
     9.6.  Server's Preferred Address  . . . . . . . . . . . . . . .  48
       9.6.1.  Communicating A Preferred Address . . . . . . . . . .  49
       9.6.2.  Responding to Connection Migration  . . . . . . . . .  49
       9.6.3.  Interaction of Client Migration and Preferred Address  50
     9.7.  Use of IPv6 Flow-Label and Migration  . . . . . . . . . .  50
   10. Connection Termination  . . . . . . . . . . . . . . . . . . .  51
     10.1.  Closing and Draining Connection States . . . . . . . . .  51
     10.2.  Idle Timeout . . . . . . . . . . . . . . . . . . . . . .  52
     10.3.  Immediate Close  . . . . . . . . . . . . . . . . . . . .  53
     10.4.  Stateless Reset  . . . . . . . . . . . . . . . . . . . .  54
       10.4.1.  Detecting a Stateless Reset  . . . . . . . . . . . .  56
       10.4.2.  Calculating a Stateless Reset Token  . . . . . . . .  57
       10.4.3.  Looping  . . . . . . . . . . . . . . . . . . . . . .  58
   11. Error Handling  . . . . . . . . . . . . . . . . . . . . . . .  58
     11.1.  Connection Errors  . . . . . . . . . . . . . . . . . . .  59
     11.2.  Stream Errors  . . . . . . . . . . . . . . . . . . . . .  59
   12. Packets and Frames  . . . . . . . . . . . . . . . . . . . . .  60
     12.1.  Protected Packets  . . . . . . . . . . . . . . . . . . .  60
     12.2.  Coalescing Packets . . . . . . . . . . . . . . . . . . .  61
     12.3.  Packet Numbers . . . . . . . . . . . . . . . . . . . . .  61
     12.4.  Frames and Frame Types . . . . . . . . . . . . . . . . .  63
   13. Packetization and Reliability . . . . . . . . . . . . . . . .  65
     13.1.  Packet Processing and Acknowledgment . . . . . . . . . .  66
       13.1.1.  Sending ACK Frames . . . . . . . . . . . . . . . . .  66
       13.1.2.  ACK Frames and Packet Protection . . . . . . . . . .  67
     13.2.  Retransmission of Information  . . . . . . . . . . . . .  67



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     13.3.  Explicit Congestion Notification . . . . . . . . . . . .  70
       13.3.1.  ECN Counts . . . . . . . . . . . . . . . . . . . . .  70
       13.3.2.  ECN Verification . . . . . . . . . . . . . . . . . .  71
   14. Packet Size . . . . . . . . . . . . . . . . . . . . . . . . .  72
     14.1.  Path Maximum Transmission Unit (PMTU)  . . . . . . . . .  73
     14.2.  ICMP Packet Too Big Messages . . . . . . . . . . . . . .  74
     14.3.  Datagram Packetization Layer PMTU Discovery  . . . . . .  75
   15. Versions  . . . . . . . . . . . . . . . . . . . . . . . . . .  75
   16. Variable-Length Integer Encoding  . . . . . . . . . . . . . .  76
   17. Packet Formats  . . . . . . . . . . . . . . . . . . . . . . .  77
     17.1.  Packet Number Encoding and Decoding  . . . . . . . . . .  77
     17.2.  Long Header Packets  . . . . . . . . . . . . . . . . . .  78
       17.2.1.  Version Negotiation Packet . . . . . . . . . . . . .  81
       17.2.2.  Initial Packet . . . . . . . . . . . . . . . . . . .  82
       17.2.3.  0-RTT  . . . . . . . . . . . . . . . . . . . . . . .  85
       17.2.4.  Handshake Packet . . . . . . . . . . . . . . . . . .  86
       17.2.5.  Retry Packet . . . . . . . . . . . . . . . . . . . .  87
     17.3.  Short Header Packets . . . . . . . . . . . . . . . . . .  90
   18. Transport Parameter Encoding  . . . . . . . . . . . . . . . .  91
     18.1.  Transport Parameter Definitions  . . . . . . . . . . . .  92
   19. Frame Types and Formats . . . . . . . . . . . . . . . . . . .  95
     19.1.  PADDING Frame  . . . . . . . . . . . . . . . . . . . . .  95
     19.2.  PING Frame . . . . . . . . . . . . . . . . . . . . . . .  96
     19.3.  ACK Frames . . . . . . . . . . . . . . . . . . . . . . .  96
       19.3.1.  ACK Ranges . . . . . . . . . . . . . . . . . . . . .  98
       19.3.2.  ECN Counts . . . . . . . . . . . . . . . . . . . . .  99
     19.4.  RESET_STREAM Frame . . . . . . . . . . . . . . . . . . . 100
     19.5.  STOP_SENDING Frame . . . . . . . . . . . . . . . . . . . 101
     19.6.  CRYPTO Frame . . . . . . . . . . . . . . . . . . . . . . 101
     19.7.  NEW_TOKEN Frame  . . . . . . . . . . . . . . . . . . . . 102
     19.8.  STREAM Frames  . . . . . . . . . . . . . . . . . . . . . 103
     19.9.  MAX_DATA Frame . . . . . . . . . . . . . . . . . . . . . 104
     19.10. MAX_STREAM_DATA Frame  . . . . . . . . . . . . . . . . . 105
     19.11. MAX_STREAMS Frames . . . . . . . . . . . . . . . . . . . 106
     19.12. DATA_BLOCKED Frame . . . . . . . . . . . . . . . . . . . 107
     19.13. STREAM_DATA_BLOCKED Frame  . . . . . . . . . . . . . . . 107
     19.14. STREAMS_BLOCKED Frames . . . . . . . . . . . . . . . . . 108
     19.15. NEW_CONNECTION_ID Frame  . . . . . . . . . . . . . . . . 108
     19.16. RETIRE_CONNECTION_ID Frame . . . . . . . . . . . . . . . 110
     19.17. PATH_CHALLENGE Frame . . . . . . . . . . . . . . . . . . 111
     19.18. PATH_RESPONSE Frame  . . . . . . . . . . . . . . . . . . 111
     19.19. CONNECTION_CLOSE Frames  . . . . . . . . . . . . . . . . 111
     19.20. Extension Frames . . . . . . . . . . . . . . . . . . . . 112
   20. Transport Error Codes . . . . . . . . . . . . . . . . . . . . 113
     20.1.  Application Protocol Error Codes . . . . . . . . . . . . 114
   21. Security Considerations . . . . . . . . . . . . . . . . . . . 114
     21.1.  Handshake Denial of Service  . . . . . . . . . . . . . . 114
     21.2.  Amplification Attack . . . . . . . . . . . . . . . . . . 115



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     21.3.  Optimistic ACK Attack  . . . . . . . . . . . . . . . . . 116
     21.4.  Slowloris Attacks  . . . . . . . . . . . . . . . . . . . 116
     21.5.  Stream Fragmentation and Reassembly Attacks  . . . . . . 116
     21.6.  Stream Commitment Attack . . . . . . . . . . . . . . . . 117
     21.7.  Explicit Congestion Notification Attacks . . . . . . . . 117
     21.8.  Stateless Reset Oracle . . . . . . . . . . . . . . . . . 117
     21.9.  Version Downgrade  . . . . . . . . . . . . . . . . . . . 118
   22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 118
     22.1.  QUIC Transport Parameter Registry  . . . . . . . . . . . 118
     22.2.  QUIC Frame Type Registry . . . . . . . . . . . . . . . . 120
     22.3.  QUIC Transport Error Codes Registry  . . . . . . . . . . 120
   23. References  . . . . . . . . . . . . . . . . . . . . . . . . . 122
     23.1.  Normative References . . . . . . . . . . . . . . . . . . 123
     23.2.  Informative References . . . . . . . . . . . . . . . . . 124
   Appendix A.  Sample Packet Number Decoding Algorithm  . . . . . . 126
   Appendix B.  Change Log . . . . . . . . . . . . . . . . . . . . . 126
     B.1.  Since draft-ietf-quic-transport-17  . . . . . . . . . . . 127
     B.2.  Since draft-ietf-quic-transport-16  . . . . . . . . . . . 127
     B.3.  Since draft-ietf-quic-transport-15  . . . . . . . . . . . 129
     B.4.  Since draft-ietf-quic-transport-14  . . . . . . . . . . . 129
     B.5.  Since draft-ietf-quic-transport-13  . . . . . . . . . . . 129
     B.6.  Since draft-ietf-quic-transport-12  . . . . . . . . . . . 130
     B.7.  Since draft-ietf-quic-transport-11  . . . . . . . . . . . 131
     B.8.  Since draft-ietf-quic-transport-10  . . . . . . . . . . . 131
     B.9.  Since draft-ietf-quic-transport-09  . . . . . . . . . . . 132
     B.10. Since draft-ietf-quic-transport-08  . . . . . . . . . . . 132
     B.11. Since draft-ietf-quic-transport-07  . . . . . . . . . . . 133
     B.12. Since draft-ietf-quic-transport-06  . . . . . . . . . . . 134
     B.13. Since draft-ietf-quic-transport-05  . . . . . . . . . . . 134
     B.14. Since draft-ietf-quic-transport-04  . . . . . . . . . . . 134
     B.15. Since draft-ietf-quic-transport-03  . . . . . . . . . . . 135
     B.16. Since draft-ietf-quic-transport-02  . . . . . . . . . . . 135
     B.17. Since draft-ietf-quic-transport-01  . . . . . . . . . . . 136
     B.18. Since draft-ietf-quic-transport-00  . . . . . . . . . . . 138
     B.19. Since draft-hamilton-quic-transport-protocol-01 . . . . . 138
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 139
   Contributors  . . . . . . . . . . . . . . . . . . . . . . . . . . 139
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . 139

1.  Introduction

   QUIC is a multiplexed and secure general-purpose transport protocol
   that provides:

   o  Stream multiplexing

   o  Stream and connection-level flow control




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   o  Low-latency connection establishment

   o  Connection migration and resilience to NAT rebinding

   o  Authenticated and encrypted header and payload

   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.

1.1.  Document Structure

   This document describes the core QUIC protocol and is structured as
   follows.

   o  Streams are the basic service abstraction that QUIC provides.

      *  Section 2 describes core concepts related to streams,

      *  Section 3 provides a reference model for stream states, and

      *  Section 4 outlines the operation of flow control.

   o  Connections are the context in which QUIC endpoints communicate.

      *  Section 5 describes core concepts related to connections,

      *  Section 6 describes version negotiation,

      *  Section 7 details the process for establishing connections,

      *  Section 8 specifies critical denial of service mitigation
         mechanisms,

      *  Section 9 describes how endpoints migrate a connection to a new
         network path,

      *  Section 10 lists the options for terminating an open
         connection, and

      *  Section 11 provides general guidance for error handling.

   o  Packets and frames are the basic unit used by QUIC to communicate.

      *  Section 12 describes concepts related to packets and frames,





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      *  Section 13 defines models for the transmission, retransmission,
         and acknowledgement of data, and

      *  Section 14 specifies rules for managing the size of packets.

   o  Finally, encoding details of QUIC protocol elements are described
      in:

      *  Section 15 (Versions),

      *  Section 16 (Integer Encoding),

      *  Section 17 (Packet Headers),

      *  Section 18 (Transport Parameters),

      *  Section 19 (Frames), and

      *  Section 20 (Errors).

   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].

1.2.  Terms and Definitions

   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.

   QUIC:  The transport protocol described by this document.  QUIC is a
      name, not an acronym.

   QUIC packet:  The smallest unit of QUIC that can be encapsulated in a
      UDP datagram.  Multiple QUIC packets can be encapsulated in a
      single UDP datagram.

   Endpoint:  An entity that can participate in a QUIC connection by
      generating, receiving, and processing QUIC packets.  There are
      only two types of endpoint in QUIC: client and server.




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   Client:  The endpoint initiating a QUIC connection.

   Server:  The endpoint accepting incoming QUIC connections.

   Connection ID:  An opaque identifier that is used to identify a QUIC
      connection at an endpoint.  Each endpoint sets a value for its
      peer to include in packets sent towards the endpoint.

   Stream:  A unidirectional or bidirectional channel of ordered bytes
      within a QUIC connection.  A QUIC connection can carry multiple
      simultaneous streams.

   Application:  An entity that uses QUIC to send and receive data.

1.3.  Notational Conventions

   Packet and frame diagrams in this document use the format described
   in Section 3.1 of [RFC2360], with the following additional
   conventions:

   [x]:  Indicates that x is optional

   x (A):  Indicates that x is A bits long

   x (A/B/C) ...:  Indicates that x is one of A, B, or C bits long

   x (i) ...:  Indicates that x uses the variable-length encoding in
      Section 16

   x (*) ...:  Indicates that x is variable-length

2.  Streams

   Streams in QUIC provide a lightweight, ordered byte-stream
   abstraction to an application.  An alternative view of QUIC streams
   is as an elastic "message" abstraction.

   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.



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   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.

2.1.  Stream Types and Identifiers

   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.  Stream IDs are unique to a stream.  A
   QUIC endpoint MUST NOT reuse a stream ID within a connection.  Stream
   IDs are encoded as variable-length integers (see Section 16).

   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 |
                +------+----------------------------------+

                         Table 1: Stream ID Types

   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.




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   The first bidirectional stream opened by the client has a stream ID
   of 0.

2.2.  Sending and Receiving Data

   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 that is visible to QUIC.  STREAM frame boundaries are not
   expected to be preserved when data is transmitted, when data is
   retransmitted after packet loss, or when data is 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.

2.3.  Stream Prioritization

   Stream multiplexing can have a significant effect on application
   performance if resources allocated to streams are correctly
   prioritized.

   QUIC does not provide frames 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




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   streams to dedicate resources to, the implementation SHOULD use the
   information provided by the application.

3.  Stream States

   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.

   Note:  These states are largely informative.  This document uses
      stream states to describe rules for when and how different types
      of frames can be sent and the reactions that are expected when
      different types of frames are received.  Though these state
      machines are intended to be useful in implementing QUIC, these
      states aren't intended to constrain implementations.  An
      implementation can define a different state machine as long as its
      behavior is consistent with an implementation that implements
      these states.

3.1.  Sending Stream States

   Figure 1 shows the states for the part of a stream that sends data to
   a peer.
















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          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 frame and enters this state, which
   can allow for better stream prioritization.




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   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.

3.2.  Receiving Stream States

   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



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   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".






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   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 creating a stream, 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".  In this state, the endpoint has all stream data.  Any STREAM
   or STREAM_DATA_BLOCKED frames it receives 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



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   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 immediately.  Alternatively, the
   RESET_STREAM signal might be suppressed or withheld if stream data is
   completely received and is buffered to be read by the application.
   In the latter case, the receiving part of the stream transitions from
   "Reset Recvd" to "Data Recvd".

   Once the application has been delivered 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.

3.3.  Permitted Frame Types

   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.

3.4.  Bidirectional Stream States

   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




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   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     | half-closed         |
   |                       | Read                | (remote)            |
   |                       |                     |                     |
   | Ready/Send/Data Sent  | Reset Recvd/Reset   | half-closed         |
   |                       | Read                | (remote)            |
   |                       |                     |                     |
   | Data Recvd            | Recv/Size Known     | half-closed (local) |
   |                       |                     |                     |
   | Reset Sent/Reset      | Recv/Size Known     | half-closed (local) |
   | Recvd                 |                     |                     |
   |                       |                     |                     |
   | Reset Sent/Reset      | Data Recvd/Data     | closed              |
   | Recvd                 | Read                |                     |
   |                       |                     |                     |
   | Reset Sent/Reset      | Reset Recvd/Reset   | closed              |
   | Recvd                 | Read                |                     |
   |                       |                     |                     |
   | Data Recvd            | Data Recvd/Data     | closed              |
   |                       | Read                |                     |
   |                       |                     |                     |
   | Data Recvd            | Reset Recvd/Reset   | closed              |
   |                       | Read                |                     |
   +-----------------------+---------------------+---------------------+

           Table 2: Possible Mapping of Stream States to HTTP/2

   Note (*1):  A stream is considered "idle" if it has not yet been
      created, or if the receiving part of the stream is in the "Recv"
      state without yet having received any frames.




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3.5.  Solicited State Transitions

   If an endpoint is no longer interested in the data it is receiving on
   a stream, it MAY send a STOP_SENDING frame identifying that stream 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
   will 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.







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4.  Flow Control

   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.

4.1.  Data Flow Control

   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:

   o  Stream flow control, which prevents a single stream from consuming
      the entire receive buffer for a connection by limiting the amount
      of data that can be sent on any stream.

   o  Connection flow control, which prevents senders from exceeding a
      receiver's buffer capacity for the connection, by limiting the
      total bytes of stream data sent in STREAM frames on all streams.

   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



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   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 at any time during the connection.  A receiver
   cannot renege on an advertisement however.  That is, 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
   STREAM_DATA_BLOCKED or DATA_BLOCKED frames 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 sends a single STREAM_DATA_BLOCKED or DATA_BLOCKED frame
   only once when it reaches a data limit.  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.

4.2.  Flow Credit Increments

   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.



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   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.

4.3.  Handling Stream Cancellation

   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.  If a RESET_STREAM frame is reordered with stream data for
   the same stream, the receiver's estimate of the number of bytes
   received on that stream can be lower than the sender's estimate of
   the number sent.  As a result, 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.



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4.4.  Stream Final Size

   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 largest byte offset sent on
   the stream.

   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.

4.5.  Controlling Concurrency

   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 18.1) and
   subsequently limits are advertised using MAX_STREAMS frames
   (Section 19.11).  Separate limits apply to unidirectional and
   bidirectional streams.

   Endpoints MUST NOT exceed the limit set by their peer.  An endpoint
   that receives a STREAM frame with a stream ID exceeding the limit it
   has sent MUST treat this as a stream error of type STREAM_LIMIT_ERROR
   (Section 11).

   A receiver cannot renege on an advertisement.  That is, once a
   receiver advertises a stream limit using the MAX_STREAMS frame,



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   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.

5.  Connections

   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.

5.1.  Connection ID

   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 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.





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   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 MAY be used when the connection ID is not
   needed for routing and the address/port tuple of packets is
   sufficient to identify a connection.  An endpoint whose peer has
   selected a zero-length connection ID MUST continue to use a zero-
   length connection ID for the lifetime of the connection and MUST NOT
   send packets from any other local address.

   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).

5.1.1.  Issuing Connection IDs

   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.



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   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).

   Endpoints store received connection IDs for future use.  An endpoint
   that receives excessive connection IDs MAY discard those it cannot
   store without sending a RETIRE_CONNECTION_ID frame.  An endpoint that
   issues connection IDs cannot expect its peer to store and use all
   issued connection IDs.

   An endpoint SHOULD ensure that its peer has a sufficient number of
   available and unused connection IDs.  While each endpoint
   independently chooses how many connection IDs to issue, endpoints
   SHOULD provide and maintain at least eight connection IDs.  The
   endpoint SHOULD do this by always supplying a new connection ID when
   a connection ID is retired by its peer or when the endpoint receives
   a packet with a previously unused connection ID.  Endpoints that
   initiate migration and require non-zero-length connection IDs SHOULD
   provide their peers with new connection IDs before migration, or risk
   the peer closing the connection.

5.1.2.  Consuming and Retiring Connection IDs

   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 won't 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.

5.2.  Matching Packets to Connections

   Incoming packets are classified on receipt.  Packets can either be
   associated with an existing connection, or - for servers -
   potentially create a new connection.




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   Hosts try to associate a packet with an existing connection.  If the
   packet has a 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 address/port tuple of a connection where the host did not
   require connection IDs, QUIC processes the packet as part of that
   connection.  Endpoints SHOULD either reject connection attempts that
   use the same addresses as existing connections, or use a non-zero-
   length Destination Connection ID so that packets can be correctly
   attributed to connections.

   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.

5.2.1.  Client Packet Handling

   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 address/port tuple to
   identify a connection.  Packets that don't match an existing
   connection are discarded.

   Due to packet reordering or loss, clients might receive packets for a
   connection that are encrypted with a key it has not yet computed.
   Clients 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.






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5.2.2.  Server Packet Handling

   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.

   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.

   Servers MUST drop other packets that contain unsupported versions.

   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 address tuple.  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 forbidden from sending Handshake packets prior
   to receiving a server response, so servers SHOULD ignore any such
   packets.

   Servers MUST drop incoming packets under all other circumstances.

5.3.  Life of a QUIC Connection

   TBD.








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6.  Version Negotiation

   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.

6.1.  Sending Version Negotiation Packets

   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.

   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.

6.2.  Handling Version Negotiation Packets

   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 21.9.





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6.2.1.  Version Negotiation Between Draft Versions

   [[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.

6.3.  Using Reserved Versions

   For a server to use a new version in the future, clients must
   correctly handle unsupported versions.  To help ensure this, a server
   SHOULD include a reserved version (see Section 15) while 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 reserved version number.  This can
   be used to solicit a list of supported versions from a server.

7.  Cryptographic and Transport Handshake

   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



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   the handshake protocol as possible.  The cryptographic handshake MUST
   provide the following properties:

   o  authenticated key exchange, where

      *  a server is always authenticated,

      *  a client is optionally authenticated,

      *  every connection produces distinct and unrelated keys,

      *  keying material is usable for packet protection for both 0-RTT
         and 1-RTT packets, and

      *  1-RTT keys have forward secrecy

   o  authenticated values for the transport parameters of the peer (see
      Section 7.3)

   o  authenticated negotiation of an application protocol (TLS uses
      ALPN [RFC7301] for this purpose)

   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 13.3.2.

   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.








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7.1.  Example Handshake Flows

   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[55, "..."], 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




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   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, CERT, CV, FIN]
                             <- 1-RTT[0]: STREAM[1, "..."] ACK[0]

   Initial[1]: ACK[0]
   Handshake[0]: CRYPTO[FIN], ACK[0]
   1-RTT[2]: STREAM[0, "..."] ACK[0] ->

                            1-RTT[1]: STREAM[55, "..."], ACK[1,2]
                                          <- Handshake[1]: ACK[0]

                     Figure 4: Example 0-RTT Handshake

7.2.  Negotiating Connection IDs

   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 which has not previously
   received a 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 SCIL field to indicate the length.



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   The first flight of 0-RTT packets use the same Destination and Source
   Connection ID values as the client's first Initial.

   The Destination Connection ID field in the server's Initial packet
   contains a connection ID that is chosen by the recipient of the
   packet (i.e., the client); the Source Connection ID includes the
   connection ID that the sender of the packet wishes to use (see
   Section 5.1).  The server MUST use consistent Source Connection IDs
   during the handshake.

   On 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.

7.3.  Transport Parameters

   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.




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   Definitions for each of the defined transport parameters are included
   in Section 18.1.  An endpoint MUST treat receipt of a transport
   parameter with an invalid value as a connection error of type
   TRANSPORT_PARAMETER_ERROR.  Any given parameter MUST appear at most
   once in a given transport parameters extension.  An endpoint MUST
   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 18.1) if it sent a Retry packet to enable validation of the
   Retry, as described in Section 17.2.5.

7.3.1.  Values of Transport Parameters for 0-RTT

   A client that attempts to send 0-RTT data MUST remember the transport
   parameters used by the server.  The transport parameters that the
   server advertises during connection establishment apply to all
   connections that are resumed using the keying material established
   during that handshake.  Remembered transport parameters apply to the
   new connection until the handshake completes and new transport
   parameters from the server can be provided.

   A server can remember the transport parameters 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
   transport parameters in determining whether to accept 0-RTT data.

   A server MAY accept 0-RTT and subsequently provide different values
   for transport parameters for use in the new connection.  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 18.1) that are
   smaller than the remembered value of those parameters.

   o  initial_max_data

   o  initial_max_stream_data_bidi_local

   o  initial_max_stream_data_bidi_remote

   o  initial_max_stream_data_uni

   o  initial_max_streams_bidi

   o  initial_max_streams_uni





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   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.

   The value of the server's previous preferred_address MUST NOT be used
   when establishing a new connection; rather, the client should wait to
   observe the server's new preferred_address value in the handshake.

   A server MUST either reject 0-RTT data or abort a handshake if the
   implied values for transport parameters cannot be supported.

   A client MUST NOT use updated values it learns either from the
   server's updated transport parameters or frames carried in 1-RTT when
   sending frames in 0-RTT packets.  Updated values of transport
   parameters from the handshake only apply 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; increased limits from frames cannot be applied, because
   the client will send frames that use any increased limits in 1-RTT
   packets (see Section 17.2.3).  A server MAY treat use of updated
   transport parameters in 0-RTT as a connection error of type
   PROTOCOL_VIOLATION.

7.3.2.  New Transport Parameters

   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.

   New transport parameters can be registered according to the rules in
   Section 22.1.

8.  Address Validation

   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



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   claims.  Address validation is performed both during connection
   establishment (see Section 8.1) and during connection migration (see
   Section 8.2).

8.1.  Address Validation During Connection Establishment

   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 pad UDP datagrams that contain only Initial packets to
   at least 1200 bytes.  Once a client has received an acknowledgment
   for a Handshake packet it MAY send smaller datagrams.  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 handshake 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.





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8.1.1.  Address Validation using Retry Packets

   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 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[0]: CRYPTO[CH] ->

                                    Initial[0]: CRYPTO[SH] ACK[0]
                          Handshake[0]: CRYPTO[EE, CERT, CV, FIN]
                                    <- 1-RTT[0]: STREAM[1, "..."]

                  Figure 5: Example Handshake with Retry

8.1.2.  Address Validation for Future Connections

   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



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   client MUST include the token in all Initial packets it sends, unless
   a Retry replaces the token with a newer token.  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 include an expiration time.
   The server MAY include either an explicit expiration time or an
   issued timestamp and dynamically calculate the expiration time.  It
   is also unlikely that the client port number is the same on two
   different connections; validating the port is therefore unlikely to
   be successful.

   A token SHOULD be constructed to be easily distinguishable from
   tokens that are sent in Retry packets as they are carried in the same
   field.

   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 it migrates prior to completing the handshake.

   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.





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   Note:  The rationale for treating the client as unvalidated rather
      than discarding the packet is that the client might have received
      the token in a previous connection using the NEW_TOKEN frame, and
      if the server has lost state, it might be unable to validate the
      token at all, leading to connection failure if the packet is
      discarded.  A server MAY encode tokens provided with NEW_TOKEN
      frames and Retry packets differently, and validate the latter more
      strictly.

   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.

8.1.3.  Address Validation Token Integrity

   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.






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8.2.  Path Validation

   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.

8.3.  Initiating Path Validation

   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.  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.





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   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.

8.4.  Path Validation Responses

   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.

   To ensure that packets can be both sent to and received from the
   peer, the PATH_RESPONSE MUST be sent on the same path as the
   triggering PATH_CHALLENGE.  That is, from the same local address on
   which the PATH_CHALLENGE was received, to the same remote address
   from which the PATH_CHALLENGE was received.

8.5.  Successful Path Validation

   A new address is considered valid when a PATH_RESPONSE frame is
   received that meets the following criteria:

   o  It 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.

   o  It was sent from the same remote address to which the
      corresponding PATH_CHALLENGE was sent.  If a PATH_RESPONSE frame
      is received from a different remote address than the one to which
      the PATH_CHALLENGE was sent, path validation is considered to have
      failed, even if the data matches that sent in the PATH_CHALLENGE.

   o  It was received on the same local address from which the
      corresponding PATH_CHALLENGE was sent.

   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.

8.6.  Failed Path Validation

   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



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   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.

9.  Connection Migration

   The use of a connection ID allows connections to survive changes to
   endpoint addresses (that is, IP address and/or 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.

   An endpoint MUST NOT initiate connection migration before the
   handshake is finished and the endpoint has 1-RTT keys.  The design of
   QUIC relies on endpoints retaining a stable address for the duration
   of the handshake.

   An endpoint also MUST NOT initiate connection migration if the peer
   sent the "disable_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
   MAY treat this as a connection error of type INVALID_MIGRATION.

   Not all changes of peer address are intentional 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.  NAT rebinding is not connection
   migration as defined in this section, though an endpoint SHOULD
   perform path validation (Section 8.2) if it detects a change in the
   IP address of its peer.





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   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.

9.1.  Probing a New Path

   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".

9.2.  Initiating Connection Migration

   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.





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   The new path might not have the same ECN capability.  Therefore, the
   endpoint verifies ECN capability as described in Section 13.3.

   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.

9.3.  Responding to Connection Migration

   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.

9.3.1.  Peer Address Spoofing

   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.





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   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.

9.3.2.  On-Path Address Spoofing

   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.

9.3.3.  Off-Path Packet Forwarding

   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



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   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.








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9.4.  Loss Detection and Congestion Control

   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
   SHOULD immediately reset the congestion controller and round-trip
   time estimator for the new path.

   An endpoint MUST NOT return to the send rate used for the previous
   path unless it is reasonably sure that the previous send rate is
   valid for the new path.  For instance, a change in the client's port
   number is likely indicative of a rebinding in a middlebox and not a
   complete change in path.  This determination likely depends on
   heuristics, which could be imperfect; if the new path capacity is
   significantly reduced, ultimately this relies on the congestion
   controller responding to congestion signals and reducing send rates
   appropriately.

   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.

9.5.  Privacy Implications of Connection Migration

   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,



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   as discussed in Section 5.1.  For this to be effective endpoints need
   to ensure that connections IDs they provide cannot be linked by any
   other entity.

   This eliminates the use of the connection ID for linking activity
   from the same connection on different networks.  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.

   Clients MAY move to a new connection ID at any time based on
   implementation-specific concerns.  For example, after a period of
   network inactivity NAT rebinding might occur when the client begins
   sending data again.

   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.

   Endpoints that use connection IDs with length greater than zero could
   have their activity correlated if their peers keep using the same
   destination connection ID after migration.  Endpoints that receive
   packets with a previously unused Destination Connection ID SHOULD
   change to sending packets with a connection ID that has not been used
   on any other network path.  The goal here is to ensure that packets
   sent on different paths cannot be correlated.  To fulfill this
   privacy requirement, endpoints that initiate migration and use
   connection IDs with length greater than zero SHOULD provide their
   peers with new connection IDs before migration.

   Caution:  If both endpoints change connection ID in response to
      seeing a change in connection ID from their peer, then this can
      trigger an infinite sequence of changes.

9.6.  Server's Preferred Address

   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




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   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.

9.6.1.  Communicating A Preferred Address

   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.

9.6.2.  Responding to Connection Migration

   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.



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9.6.3.  Interaction of Client Migration and Preferred Address

   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.

9.7.  Use of IPv6 Flow-Label and Migration

   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.





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10.  Connection Termination

   Connections should remain open until they become idle for a pre-
   negotiated period of time.  A QUIC connection, once established, can
   be terminated in one of three ways:

   o  idle timeout (Section 10.2)

   o  immediate close (Section 10.3)

   o  stateless reset (Section 10.4)

10.1.  Closing and Draining Connection States

   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 can confirm that its peer is also closing or draining.
   Receiving a CONNECTION_CLOSE frame is sufficient confirmation, as is
   receiving a stateless reset.  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 be



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   handled poorly.  Endpoints that have some alternative means to ensure
   that late-arriving packets on the connection do not create QUIC
   state, 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.

10.2.  Idle Timeout

   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 18.1) 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 an Probe Timeout



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   (PTO, see Section 6.2.2 of [QUIC-RECOVERY]), it is advisable to test
   for liveness before sending any data that cannot be retried safely.

10.3.  Immediate Close

   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.

   Note:  Allowing retransmission of a closing packet contradicts other
      advice in this document that recommends the creation of new packet
      numbers for every packet.  Sending new packet numbers is primarily
      of advantage to loss recovery and congestion control, which are
      not expected to be relevant for a closed connection.
      Retransmitting the final packet requires less state.

   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



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   requests that the connection be closed.  The application protocol can
   use an CONNECTION_CLOSE frame with an appropriate error code to
   signal closure.

   If the connection has been successfully established, endpoints MUST
   send any CONNECTION_CLOSE frames in a 1-RTT packet.  Prior to
   connection establishment a peer might not have 1-RTT keys, so
   endpoints SHOULD send CONNECTION_CLOSE frames in a Handshake packet.
   If the endpoint does not have Handshake keys, or it is not certain
   that the peer has Handshake keys, it MAY send CONNECTION_CLOSE frames
   in an Initial packet.  If multiple packets are sent, they can be
   coalesced (see Section 12.2) to facilitate retransmission.

10.4.  Stateless Reset

   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.  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 NEW_CONNECTION_ID frame sent by either peer, and
   servers can specify the stateless_reset_token transport parameter
   during the handshake (clients cannot because their transport
   parameters don't have confidentiality protection).  This value is
   protected by encryption, so only client and server know this 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:

















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    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 (182..)                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                                                               |
   +                   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.

   A stateless reset will be interpreted by a recipient as a packet with
   a short header.  For the packet to appear as valid, the Random Bits
   field needs to include at least 182 bits of data (or 23 bytes, less
   the two fixed bits).  This is intended to allow for a Destination
   Connection ID of the maximum length permitted, with a minimal packet
   number, and payload.  The Stateless Reset Token corresponds to the
   minimum expansion of the packet protection AEAD.  More unpredictable
   bytes might be necessary if the endpoint could have negotiated a
   packet protection scheme with a larger minimum AEAD expansion.

   An endpoint SHOULD NOT send a stateless reset that is significantly
   larger than the packet it receives.  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.

   An endpoint MAY send a stateless reset in response to a packet with a
   long header.  This would not be effective if the stateless reset
   token was not yet 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




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   establishment is complete or near completion, ignoring an unknown
   packet with a long header might be more effective.

   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:

   o  The packet might not reach the peer.  If the Destination
      Connection ID is critical for routing toward the peer, then this
      packet could be incorrectly routed.  This might also trigger
      another Stateless Reset in response, see Section 10.4.3.  A
      Stateless Reset that is not correctly routed is an ineffective
      error detection and recovery mechanism.  In this case, endpoints
      will need to rely on other methods - such as timers - to detect
      that the connection has failed.

   o  The randomly generated connection ID can be used by entities other
      than the peer to identify this as a potential stateless reset.  An
      endpoint that occasionally uses different connection IDs might
      introduce some uncertainty about this.

   Finally, the last 16 bytes of the packet are set to the value of the
   Stateless Reset Token.

   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.

10.4.1.  Detecting a Stateless Reset

   An endpoint detects a potential stateless reset when a incoming
   packet with a short header either cannot be associated with a
   connection, cannot be decrypted, or is marked as a duplicate packet.
   The endpoint then compares the last 16 bytes of the packet with the
   Stateless Reset Token provided by its peer, either in a
   NEW_CONNECTION_ID frame or the server's transport parameters.  If
   these values are identical, the endpoint MUST enter the draining




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   period and not send any further packets on this connection.  If the
   comparison fails, the packet can be discarded.

10.4.2.  Calculating a Stateless Reset Token

   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 cannot occur 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 21.8).  A connection ID from a
   connection that is reset by revealing the Stateless Reset Token
   cannot be reused for new connections at nodes that share a static
   key.





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   Note that Stateless Reset packets do not have any cryptographic
   protection.

10.4.3.  Looping

   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 the recommended minimum
   size of 39 bytes could mean that the packet could reveal to an
   observer that it is a Stateless Reset.  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 can increase the odds that a packet will trigger a
   Stateless Reset if it cannot be processed by padding it to at least
   40 bytes.

11.  Error Handling

   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.




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   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.

11.1.  Connection Errors

   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 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
   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.

11.2.  Stream Errors

   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, either
   directly or through the receipt of a STOP_SENDING frame from a peer.
   RESET_STREAM carries an application error code.  Resetting a stream
   without knowledge of the application protocol could cause the
   protocol to enter an unrecoverable state.  Application protocols



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   might require certain streams to be reliably delivered in order to
   guarantee consistent state between endpoints.

12.  Packets and Frames

   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.

12.1.  Protected Packets

   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



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   packet number increases with each packet sent in a given packet
   number space, see Section 12.3 for details.

12.2.  Coalescing Packets

   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 starting sending
   data.  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.

   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.  Though the values of some fields in the
   packet header might be redundant, no fields are omitted.  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.

12.3.  Packet Numbers

   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




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   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:

   o  Initial space: All Initial packets Section 17.2.2 are in this
      space.

   o  Handshake space: All Handshake packets Section 17.2.4 are in this
      space.

   o  Application data space: All 0-RTT and 1-RTT encrypted packets
      Section 12.1 are in this space.

   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 (that is, under the same cryptographic
   keys).  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.



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   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.

12.4.  Frames and Frame Types

   The payload of QUIC packets, after removing packet protection,
   commonly consists of a sequence of 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

   QUIC payloads MUST contain at least one frame, and MAY contain
   multiple frames and multiple frame types.

   Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC
   packet boundary.  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




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   The frame types defined in this specification are listed in Table 3.
   The Frame Type in STREAM 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 |
          +-------------+----------------------+---------------+

                           Table 3: Frame Types



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   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.

13.  Packetization and Reliability

   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.






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13.1.  Packet Processing and Acknowledgment

   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.

13.1.1.  Sending ACK Frames

   An endpoint MUST NOT send more than one packet containing only an ACK
   frame per received packet that contains frames other than ACK and
   PADDING frames.  An endpoint MUST NOT send a packet containing only
   an ACK frame in response to a packet containing only ACK or PADDING
   frames, even if there are packet gaps which precede the received
   packet.  This prevents an indefinite feedback loop of ACKs.  The
   endpoint MUST however acknowledge packets containing only ACK or
   PADDING frames when sending ACK frames in response to other 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.

   The receiver's delayed acknowledgment timer SHOULD NOT exceed the
   current RTT estimate or the value it indicates in the "max_ack_delay"
   transport parameter.  This ensures an acknowledgment is sent at least
   once per RTT when packets needing acknowledgement are received.  The
   sender can use the receiver's "max_ack_delay" value in determining
   timeouts for timer-based retransmission.

   Strategies and implications of the frequency of generating
   acknowledgments are discussed in more detail in [QUIC-RECOVERY].

   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.





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   Because ACK frames are not sent in response to ACK-only packets, a
   receiver that is only sending ACK frames will only receive
   acknowledgements for its packets if the sender includes them in
   packets with non-ACK frames.  A sender SHOULD bundle ACK frames with
   other frames when possible.

   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 [QUIC-RECOVERY] algorithms 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 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.

13.1.2.  ACK Frames and Packet Protection

   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.

   Endpoints SHOULD send acknowledgments for packets containing CRYPTO
   frames with a reduced delay; see Section 6.2.1 of [QUIC-RECOVERY].

13.2.  Retransmission of Information

   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.



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   o  Data sent in CRYPTO frames is retransmitted according to the rules
      in [QUIC-RECOVERY], until all data has been acknowledged.  Data in
      CRYPTO frames for Initial and Handshake packets is discarded when
      keys for the corresponding encryption level are discarded.

   o  Application data sent in STREAM frames is retransmitted in new
      STREAM frames unless the endpoint has sent a RESET_STREAM for that
      stream.  Once an endpoint sends a RESET_STREAM frame, no further
      STREAM frames are needed.

   o  The most recent set of acknowledgments are sent in ACK frames.  An
      ACK frame SHOULD contain all unacknowledged acknowledgments, as
      described in Section 13.1.1.

   o  Cancellation of stream transmission, as carried in a RESET_STREAM
      frame, is sent until acknowledged or until all stream data is
      acknowledged by the peer (that is, either the "Reset Recvd" or
      "Data Recvd" state is reached on the sending part of the stream).
      The content of a RESET_STREAM frame MUST NOT change when it is
      sent again.

   o  Similarly, a request to cancel stream transmission, as encoded in
      a STOP_SENDING frame, is sent until the receiving part of the
      stream enters either a "Data Recvd" or "Reset Recvd" state, see
      Section 3.5.

   o  Connection close signals, including packets that contain
      CONNECTION_CLOSE frames, are not sent again when packet loss is
      detected, but as described in Section 10.

   o  The current connection maximum data is sent in MAX_DATA frames.
      An updated value is sent in a MAX_DATA frame if the packet
      containing the most recently sent MAX_DATA frame is declared lost,
      or when the endpoint decides to update the limit.  Care is
      necessary to avoid sending this frame too often as the limit can
      increase frequently and cause an unnecessarily large number of
      MAX_DATA frames to be sent.

   o  The current maximum stream data offset is sent in MAX_STREAM_DATA
      frames.  Like MAX_DATA, an updated value is sent when the packet
      containing the most recent MAX_STREAM_DATA frame for a stream is
      lost or when the limit is updated, with care taken to prevent the
      frame from being sent too often.  An endpoint SHOULD stop sending
      MAX_STREAM_DATA frames when the receiving part of the stream
      enters a "Size Known" state.

   o  The limit on streams of a given type is sent in MAX_STREAMS
      frames.  Like MAX_DATA, an updated value is sent when a packet



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      containing the most recent MAX_STREAMS for a stream type frame is
      declared lost or when the limit is updated, with care taken to
      prevent the frame from being sent too often.

   o  Blocked signals are carried in DATA_BLOCKED, STREAM_DATA_BLOCKED,
      and STREAMS_BLOCKED frames.  DATA_BLOCKED frames have connection
      scope, STREAM_DATA_BLOCKED frames have stream scope, and
      STREAMS_BLOCKED frames are scoped to a specific stream type.  New
      frames are sent if packets containing the most recent frame for a
      scope is lost, but only while the endpoint is blocked on the
      corresponding limit.  These frames always include the limit that
      is causing blocking at the time that they are transmitted.

   o  A liveness or path validation check using PATH_CHALLENGE frames is
      sent periodically until a matching PATH_RESPONSE frame is received
      or until there is no remaining need for liveness or path
      validation checking.  PATH_CHALLENGE frames include a different
      payload each time they are sent.

   o  Responses to path validation using PATH_RESPONSE frames are sent
      just once.  A new PATH_CHALLENGE frame will be sent if another
      PATH_RESPONSE frame is needed.

   o  New connection IDs are sent in NEW_CONNECTION_ID frames and
      retransmitted if the packet containing them is lost.
      Retransmissions of this frame carry the same sequence number
      value.  Likewise, retired connection IDs are sent in
      RETIRE_CONNECTION_ID frames and retransmitted if the packet
      containing them is lost.

   o  PING and PADDING frames contain no information, so lost PING or
      PADDING frames do not require repair.

   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].





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13.3.  Explicit Congestion Notification

   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
   verifies the path, both during connection establishment and when
   migrating to a new path (see Section 9).

13.3.1.  ECN Counts

   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 13.1 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 21.7) 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 13.1 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.




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13.3.2.  ECN Verification

   Each endpoint independently verifies and enables use of ECN by
   setting the IP header ECN codepoint to ECN Capable Transport (ECT)
   for the path from it to the other peer.  Even if not setting ECN
   codepoints on packets it transmits, the endpoint SHOULD provide
   feedback about ECN markings received (if accessible).

   To verify both that a path supports ECN and the peer can provide ECN
   feedback, an endpoint sets the ECT(0) codepoint in the IP header of
   all outgoing packets [RFC8311].

   If an ECT codepoint set in the IP header is not corrupted by a
   network device, then a received packet contains either the codepoint
   sent by the peer or the Congestion Experienced (CE) codepoint set by
   a network device that is experiencing congestion.

   If a QUIC packet sent with an ECT codepoint is newly acknowledged by
   the peer in an ACK frame without ECN feedback, the endpoint stops
   setting ECT codepoints in subsequent IP packets, with the expectation
   that either the network path or the peer no longer supports ECN.

   Network devices that corrupt or apply non-standard ECN markings might
   result in reduced throughput or other undesirable side-effects.  To
   reduce this risk, an endpoint uses the following steps to verify the
   counts it receives in an ACK frame.

   o  The total increase in ECT(0), ECT(1), and CE counts MUST be no
      smaller than the total number of QUIC packets sent with an ECT
      codepoint that are newly acknowledged in this ACK frame.  This
      step detects any network remarking from ECT(0), ECT(1), or CE
      codepoints to Not-ECT.

   o  Any increase in either ECT(0) or ECT(1) counts, plus any increase
      in the CE count, MUST be no smaller than the number of packets
      sent with the corresponding ECT codepoint that are newly
      acknowledged in this ACK frame.  This step detects any erroneous
      network remarking from ECT(0) to ECT(1) (or vice versa).

   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 verification
   succeeds, the local reference counts MUST be increased to match the
   counts in the ACK frame.

   Processing counts out of order can result in verification failure.
   An endpoint SHOULD NOT perform this verification if the ACK frame is



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   received in a packet with packet number lower than a previously
   received ACK frame.  Verifying based on ACK frames that arrive out of
   order can result in disabling ECN unnecessarily.

   Upon successful verification, an endpoint continues to set ECT
   codepoints in subsequent packets with the expectation that the path
   is ECN-capable.

   If verification fails, then the endpoint ceases setting ECT
   codepoints in subsequent IP packets with the expectation that either
   the network path or the peer does not support ECN.

   If an endpoint sets ECT codepoints on outgoing IP packets and
   encounters a retransmission timeout due to the absence of
   acknowledgments from the peer (see [QUIC-RECOVERY]), or if an
   endpoint has reason to believe that an element on the network path
   might be corrupting ECN codepoints, the endpoint MAY cease setting
   ECT codepoints in subsequent packets.  Doing so allows the connection
   to be resilient to network elements that corrupt ECN codepoints in
   the IP header or drop packets with ECT or CE codepoints in the IP
   header.

14.  Packet Size

   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 combining the Initial packet with a 0-RTT
   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



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   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.

14.1.  Path Maximum Transmission Unit (PMTU)

   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.







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14.2.  ICMP Packet Too Big Messages

   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:

   o  An IPv4 endpoint could set the Don't Fragment (DF) bit on a small
      proportion of packets, so that most invalid ICMP messages arrive
      when there are no DF packets outstanding, and can therefore be
      identified as spurious.

   o  An endpoint could store additional information from the IP or UDP
      headers to use for validation (for example, the IP ID or UDP
      checksum).

   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.




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14.3.  Datagram Packetization Layer PMTU Discovery

   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.

15.  Versions

   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



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   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>.

16.  Variable-Length Integer Encoding

   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 |
          +------+--------+-------------+-----------------------+

                   Table 4: Summary of Integer Encodings





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   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.

17.  Packet Formats

   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.

17.1.  Packet Number Encoding and Decoding

   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



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   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.

17.2.  Long Header Packets

    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)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                    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:

   Header Form:  The most significant bit (0x80) of byte 0 (the first
      byte) is set to 1 for long headers.

   Fixed Bit:  The next bit (0x40) of byte 0 is set to 1.  Packets
      containing a zero value for this bit are not valid packets in this
      version and MUST be discarded.





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   Long Packet Type (T):  The next two bits (those with a mask of 0x30)
      of byte 0 contain a packet type.  Packet types are listed in
      Table 5.

   Type-Specific Bits (X):  The lower four bits (those with a mask of
      0x0f) of byte 0 are type-specific.

   Version:  The QUIC Version is a 32-bit field that follows the first
      byte.  This field indicates which version of QUIC is in use and
      determines how the rest of the protocol fields are interpreted.

   DCIL and SCIL:  The byte following the version contains the lengths
      of the two connection ID fields that follow it.  These lengths are
      encoded as two 4-bit unsigned integers.  The Destination
      Connection ID Length (DCIL) field occupies the 4 high bits of the
      byte and the Source Connection ID Length (SCIL) field occupies the
      4 low bits of the byte.  An encoded length of 0 indicates that the
      connection ID is also 0 bytes in length.  Non-zero encoded lengths
      are increased by 3 to get the full length of the connection ID,
      producing a length between 4 and 18 bytes inclusive.  For example,
      an byte with the value 0x50 describes an 8-byte Destination
      Connection ID and a zero-length Source Connection ID.

   Destination Connection ID:  The Destination Connection ID field
      follows the connection ID lengths and is either 0 bytes in length
      or between 4 and 18 bytes.  Section 7.2 describes the use of this
      field in more detail.

   Source Connection ID:  The Source Connection ID field follows the
      Destination Connection ID and is either 0 bytes in length or
      between 4 and 18 bytes.  Section 7.2 describes the use of this
      field in more detail.

   In this version of QUIC, the following packet types with the long
   header are defined:
















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                   +------+-----------+----------------+
                   | 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 |
                   +------+-----------+----------------+

                     Table 5: Long Header Packet Types

   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:

   Reserved Bits (R):  Two bits (those with a mask of 0x0c) of byte 0
      are reserved across multiple packet types.  These bits are
      protected using header protection (see Section 5.4 of [QUIC-TLS]).
      The value included prior to protection MUST be set to 0.  An
      endpoint MUST treat receipt of a packet that has a non-zero value
      for these bits, after removing both packet and header protection,
      as a connection error of type PROTOCOL_VIOLATION.  Discarding such
      a packet after only removing header protection can expose the
      endpoint to attacks (see Section 9.3 of [QUIC-TLS]).

   Packet Number Length (P):  In packet types which contain a Packet
      Number field, the least significant two bits (those with a mask of
      0x03) of byte 0 contain the length of the packet number, encoded
      as an unsigned, two-bit integer that is one less than the length
      of the packet number field in bytes.  That is, the length of the
      packet number field is the value of this field, plus one.  These
      bits are protected using header protection (see Section 5.4 of
      [QUIC-TLS]).

   Length:  The length of the remainder of the packet (that is, the
      Packet Number and Payload fields) in bytes, encoded as a variable-
      length integer (Section 16).




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   Packet Number:  The packet number field is 1 to 4 bytes long.  The
      packet number has confidentiality protection separate from packet
      protection, as described in Section 5.4 of [QUIC-TLS].  The length
      of the packet number field is encoded in the Packet Number Length
      bits of byte 0 (see above).

17.2.1.  Version Negotiation Packet

   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)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    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.

   The Version field of a Version Negotiation packet MUST be set to
   0x00000000.





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   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.

   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.

17.2.2.  Initial Packet

   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.




















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   +-+-+-+-+-+-+-+-+
   |1|1| 0 |R R|P P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         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.

   Token Length:  A variable-length integer specifying the length of the
      Token field, in bytes.  This value is zero if no token is present.
      Initial packets sent by the server MUST set the Token Length field
      to zero; clients that receive an Initial packet with a non-zero
      Token Length field MUST either discard the packet or generate a
      connection error of type PROTOCOL_VIOLATION.

   Token:  The value of the token that was previously provided in a
      Retry packet or NEW_TOKEN frame.

   Payload:  The payload of the 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.





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   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.

17.2.2.1.  Abandoning 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 4.10 of [QUIC-TLS]) along with any loss recovery and
   congestion control state (see Sections 5.3.1.2 and 6.9 of
   [QUIC-RECOVERY]).

   Any data in CRYPTO frames is discarded - and no longer retransmitted
   - when Initial keys are discarded.







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17.2.3.  0-RTT

   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)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           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 MAY 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.
   The keys used to protect 0-RTT packets will not change as a result of
   responding to a Retry packet unless the client also regenerates the
   cryptographic handshake message.  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.





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   Receiving a Retry packet, especially a Retry that changes the
   connection ID used for subsequent packets, indicates a strong
   possibility that 0-RTT packets could be lost.  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 SHOULD instead generate a fresh cryptographic handshake
   message and start packet numbers from 0.  This ensures that new 0-RTT
   packets will not use the same keys, avoiding any risk of key and
   nonce reuse; this also prevents 0-RTT packets from previous handshake
   attempts from being accepted as part of the connection.

   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 MAY 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).

17.2.4.  Handshake Packet

   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.

















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   +-+-+-+-+-+-+-+-+
   |1|1| 2 |R R|P P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           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.

17.2.5.  Retry Packet

   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 stateless retry (see
   Section 8.1).



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    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 | ODCIL |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Original Destination Connection ID (0/32..144)     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Retry Token (*)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                          Figure 13: Retry Packet

   A Retry packet (shown in Figure 13) does not contain any protected
   fields.  In addition to the long header, it contains these additional
   fields:

   ODCIL:  The four least-significant bits of the first byte of a Retry
      packet are not protected as they are for other packets with the
      long header, because Retry packets don't contain a protected
      payload.  These bits instead encode the length of the Original
      Destination Connection ID field.  The length uses the same
      encoding as the DCIL and SCIL fields.

   Original Destination Connection ID:  The Original Destination
      Connection ID contains the value of the Destination Connection ID
      from the Initial packet that this Retry is in response to.  The
      length of this field is given in ODCIL.

   Retry Token:  An opaque token that the server can use to validate the
      client's address.

   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.



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   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 can either reuse
   the cryptographic handshake message or construct a new one at its
   discretion.

   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
   that sends additional 0-RTT packets without constructing a new
   cryptographic handshake message MUST NOT reset the packet number to 0
   after a Retry packet, see Section 17.2.3.

   A server acknowledges the use of a Retry packet for a connection
   using the original_connection_id transport parameter (see
   Section 18.1).  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.



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   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.

17.3.  Short Header Packets

   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..144)           ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     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:

   Header Form:  The most significant bit (0x80) of byte 0 is set to 0
      for the short header.

   Fixed Bit:  The next bit (0x40) of byte 0 is set to 1.  Packets
      containing a zero value for this bit are not valid packets in this
      version and MUST be discarded.

   Spin Bit (S):  The sixth bit (0x20) of byte 0 is the Latency Spin
      Bit, set as described in [SPIN].

   Reserved Bits (R):  The next two bits (those with a mask of 0x18) of
      byte 0 are reserved.  These bits are protected using header
      protection (see Section 5.4 of [QUIC-TLS]).  The value included
      prior to protection MUST be set to 0.  An endpoint MUST treat
      receipt of a packet that has a non-zero value for these bits,
      after removing both packet and header protection, as a connection



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      error of type PROTOCOL_VIOLATION.  Discarding such a packet after
      only removing header protection can expose the endpoint to attacks
      (see Section 9.3 of [QUIC-TLS]).

   Key Phase (K):  The next bit (0x04) of byte 0 indicates the key
      phase, which allows a recipient of a packet to identify the packet
      protection keys that are used to protect the packet.  See
      [QUIC-TLS] for details.  This bit is protected using header
      protection (see Section 5.4 of [QUIC-TLS]).

   Packet Number Length (P):  The least significant two bits (those with
      a mask of 0x03) of byte 0 contain the length of the packet number,
      encoded as an unsigned, two-bit integer that is one less than the
      length of the packet number field in bytes.  That is, the length
      of the packet number field is the value of this field, plus one.
      These bits are protected using header protection (see Section 5.4
      of [QUIC-TLS]).

   Destination Connection ID:  The Destination Connection ID is a
      connection ID that is chosen by the intended recipient of the
      packet.  See Section 5.1 for more details.

   Packet Number:  The packet number field is 1 to 4 bytes long.  The
      packet number has confidentiality protection separate from packet
      protection, as described in Section 5.4 of [QUIC-TLS].  The length
      of the packet number field is encoded in Packet Number Length
      field.  See Section 17.1 for details.

   Protected Payload:  Packets with a short header always include a
      1-RTT protected payload.

   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.

18.  Transport Parameter Encoding

   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].










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      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_migration(12),
         preferred_address(13),
         (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.

18.1.  Transport Parameter Definitions

   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:





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   original_connection_id (0x0000):  The value of the Destination
      Connection ID field from the first Initial packet sent by the
      client.  This transport parameter is only sent by a server.  A
      server MUST include the original_connection_id transport parameter
      if it sent a Retry packet.

   idle_timeout (0x0001):  The idle timeout is a value in seconds that
      is encoded as an integer, see (Section 10.2).  If this parameter
      is absent or zero then the idle timeout is disabled.

   stateless_reset_token (0x0002):  A stateless reset token is used in
      verifying a stateless reset, see Section 10.4.  This parameter is
      a sequence of 16 bytes.  This transport parameter is only sent by
      a server.

   max_packet_size (0x0003):  The maximum packet size parameter is an
      integer value that limits the size of packets that the endpoint is
      willing to receive.  This indicates that packets larger than this
      limit will be dropped.  The default for this parameter is the
      maximum permitted UDP payload of 65527.  Values below 1200 are
      invalid.  This limit only applies to protected packets
      (Section 12.1).

   initial_max_data (0x0004):  The initial maximum data parameter is an
      integer value that contains the initial value for the maximum
      amount of data that can be sent on the connection.  This is
      equivalent to sending a MAX_DATA (Section 19.9) for the connection
      immediately after completing the handshake.

   initial_max_stream_data_bidi_local (0x0005):  This parameter is an
      integer value specifying the initial flow control limit for
      locally-initiated bidirectional streams.  This limit applies to
      newly created bidirectional streams opened by the endpoint that
      sends the transport parameter.  In client transport parameters,
      this applies to streams with an identifier with the least
      significant two bits set to 0x0; in server transport parameters,
      this applies to streams with the least significant two bits set to
      0x1.

   initial_max_stream_data_bidi_remote (0x0006):  This parameter is an
      integer value specifying the initial flow control limit for peer-
      initiated bidirectional streams.  This limit applies to newly
      created bidirectional streams opened by the endpoint that receives
      the transport parameter.  In client transport parameters, this
      applies to streams with an identifier with the least significant
      two bits set to 0x1; in server transport parameters, this applies
      to streams with the least significant two bits set to 0x0.




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   initial_max_stream_data_uni (0x0007):  This parameter is an integer
      value specifying the initial flow control limit for unidirectional
      streams.  This limit applies to newly created unidirectional
      streams opened by the endpoint that receives the transport
      parameter.  In client transport parameters, this applies to
      streams with an identifier with the least significant two bits set
      to 0x3; in server transport parameters, this applies to streams
      with the least significant two bits set to 0x2.

   initial_max_streams_bidi (0x0008):  The initial maximum bidirectional
      streams parameter is an integer value that contains the initial
      maximum number of bidirectional streams the peer may initiate.  If
      this parameter is absent or zero, the peer cannot open
      bidirectional streams until a MAX_STREAMS frame is sent.  Setting
      this parameter is equivalent to sending a MAX_STREAMS
      (Section 19.11) of the corresponding type with the same value.

   initial_max_streams_uni (0x0009):  The initial maximum unidirectional
      streams parameter is an integer value that contains the initial
      maximum number of unidirectional streams the peer may initiate.
      If this parameter is absent or zero, the peer cannot open
      unidirectional streams until a MAX_STREAMS frame is sent.  Setting
      this parameter is equivalent to sending a MAX_STREAMS
      (Section 19.11) of the corresponding type with the same value.

   ack_delay_exponent (0x000a):  The ACK delay exponent is an integer
      value indicating an exponent used to decode the ACK Delay field in
      the ACK frame (Section 19.3).  If this value is absent, a default
      value of 3 is assumed (indicating a multiplier of 8).  The default
      value is also used for ACK frames that are sent in Initial and
      Handshake packets.  Values above 20 are invalid.

   max_ack_delay (0x000b):  The maximum ACK delay is an integer value
      indicating the maximum amount of time in milliseconds by which the
      endpoint will delay sending acknowledgments.  This value SHOULD
      include the receiver's expected delays in alarms firing.  For
      example, if a receiver sets a timer for 5ms and alarms commonly
      fire up to 1ms late, then it should send a max_ack_delay of 6ms.
      If this value is absent, a default of 25 milliseconds is assumed.
      Values of 2^14 or greater are invalid.

   disable_migration (0x000c):  The disable migration transport
      parameter is included if the endpoint does not support connection
      migration (Section 9).  Peers of an endpoint that sets this
      transport parameter MUST NOT send any packets, including probing
      packets (Section 9.1), from a local address other than that used
      to perform the handshake.  This parameter is a zero-length value.




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   preferred_address (0x000d):  The server's preferred address is used
      to effect a change in server address at the end of the handshake,
      as described in Section 9.6.  The format of this transport
      parameter is the PreferredAddress struct shown in Figure 16.  This
      transport parameter is only sent by a server.  Servers MAY choose
      to only send a preferred address of one address family by sending
      an all-zero address and port (0.0.0.0:0 or ::.0) for the other
      family.

      struct {
        opaque ipv4Address[4];
        uint16 ipv4Port;
        opaque ipv6Address[16];
        uint16 ipv6Port;
        opaque connectionId<0..18>;
        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.

19.  Frame Types and Formats

   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.

19.1.  PADDING Frame

   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.



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19.2.  PING 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.

19.3.  ACK Frames

   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



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   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:

   Largest Acknowledged:  A variable-length integer representing the
      largest packet number the peer is acknowledging; this is usually
      the largest packet number that the peer has received prior to
      generating the ACK frame.  Unlike the packet number in the QUIC
      long or short header, the value in an ACK frame is not truncated.

   ACK Delay:  A variable-length integer including the time in
      microseconds that the largest acknowledged packet, as indicated in
      the Largest Acknowledged field, was received by this peer to when
      this ACK was sent.  The value of the ACK Delay field is scaled by
      multiplying the encoded value by 2 to the power of the value of
      the "ack_delay_exponent" transport parameter set by the sender of
      the ACK frame.  The "ack_delay_exponent" defaults to 3, or a
      multiplier of 8 (see Section 18.1).  Scaling in this fashion
      allows for a larger range of values with a shorter encoding at the
      cost of lower resolution.

   ACK Range Count:  A variable-length integer specifying the number of
      Gap and ACK Range fields in the frame.

   First ACK Range:  A variable-length integer indicating the number of
      contiguous packets preceding the Largest Acknowledged that are
      being acknowledged.  The First ACK Range is encoded as an ACK



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      Range (see Section 19.3.1) starting from the Largest Acknowledged.
      That is, the smallest packet acknowledged in the range is
      determined by subtracting the First ACK Range value from the
      Largest Acknowledged.

   ACK Ranges:  Contains additional ranges of packets which are
      alternately not acknowledged (Gap) and acknowledged (ACK Range),
      see Section 19.3.1.

   ECN Counts:  The three ECN Counts, see Section 19.3.2.

19.3.1.  ACK Ranges

   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 (repeated):  A variable-length integer indicating the number of
      contiguous unacknowledged packets preceding the packet number one
      lower than the smallest in the preceding ACK Range.






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   ACK Range (repeated):  A variable-length integer indicating the
      number of contiguous acknowledged packets preceding the largest
      packet number, as determined by the preceding Gap.

   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.

19.3.2.  ECN Counts

   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:



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    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:

   ECT(0) Count:  A variable-length integer representing the total
      number packets received with the ECT(0) codepoint.

   ECT(1) Count:  A variable-length integer representing the total
      number packets received with the ECT(1) codepoint.

   CE Count:  A variable-length integer representing the total number
      packets received with the CE codepoint.

   ECN counts are maintained separately for each packet number space.

19.4.  RESET_STREAM Frame

   An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly
   terminate 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 (16)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Final Size (i)                       ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+




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   RESET_STREAM frames contain the following fields:

   Stream ID:  A variable-length integer encoding of the Stream ID of
      the stream being terminated.

   Application Protocol Error Code:  A 16-bit application protocol error
      code (see Section 20.1) which indicates why the stream is being
      closed.

   Final Size:  A variable-length integer indicating the final size of
      the stream by the RESET_STREAM sender, in unit of bytes.

19.5.  STOP_SENDING Frame

   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 (16)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   STOP_SENDING frames contain the following fields:

   Stream ID:  A variable-length integer carrying the Stream ID of the
      stream being ignored.

   Application Error Code:  A 16-bit, application-specified reason the
      sender is ignoring the stream (see Section 20.1).

19.6.  CRYPTO Frame

   The CRYPTO frame (type=0x06) is used to transmit cryptographic
   handshake messages.  It can be sent in all packet types.  The CRYPTO
   frame offers the cryptographic protocol an in-order stream of bytes.



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   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:

   Offset:  A variable-length integer specifying the byte offset in the
      stream for the data in this CRYPTO frame.

   Length:  A variable-length integer specifying the length of the
      Crypto Data field in this CRYPTO frame.

   Crypto Data:  The cryptographic message data.

   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.

19.7.  NEW_TOKEN Frame

   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:





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    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:

   Token Length:  A variable-length integer specifying the length of the
      token in bytes.

   Token:  An opaque blob that the client may use with a future Initial
      packet.

19.8.  STREAM Frames

   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 determine the fields that are present in the frame.

   o  The OFF bit (0x04) in the frame type is set to indicate that there
      is an Offset field present.  When set to 1, the Offset field is
      present.  When set to 0, the Offset field is absent and the Stream
      Data starts at an offset of 0 (that is, the frame contains the
      first bytes of the stream, or the end of a stream that includes no
      data).

   o  The LEN bit (0x02) in the frame type is set to indicate that there
      is a Length field present.  If this bit is set to 0, the Length
      field is absent and the Stream Data field extends to the end of
      the packet.  If this bit is set to 1, the Length field is present.

   o  The FIN bit (0x01) of the frame type is set only on frames that
      contain the final size of the stream.  Setting this bit indicates
      that the frame marks the end of the stream.

   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:








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    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:

   Stream ID:  A variable-length integer indicating the stream ID of the
      stream (see Section 2.1).

   Offset:  A variable-length integer specifying the byte offset in the
      stream for the data in this STREAM frame.  This field is present
      when the OFF bit is set to 1.  When the Offset field is absent,
      the offset is 0.

   Length:  A variable-length integer specifying the length of the
      Stream Data field in this STREAM frame.  This field is present
      when the LEN bit is set to 1.  When the LEN bit is set to 0, the
      Stream Data field consumes all the remaining bytes in the packet.

   Stream Data:  The bytes from the designated stream to be delivered.

   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 - MUST
   be less than 2^62.

19.9.  MAX_DATA Frame

   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:






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    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:

   Maximum Data:  A variable-length integer indicating the maximum
      amount of data that can be sent on the entire connection, in units
      of bytes.

   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).

19.10.  MAX_STREAM_DATA Frame

   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:

   Stream ID:  The stream ID of the stream that is affected encoded as a
      variable-length integer.




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   Maximum Stream Data:  A variable-length integer indicating the
      maximum amount of data that can be sent on the identified stream,
      in units of bytes.

   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).

19.11.  MAX_STREAMS Frames

   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:

   Maximum Streams:  A count of the cumulative number of streams of the
      corresponding type that can be opened over the lifetime of the
      connection.

   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,



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   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.

19.12.  DATA_BLOCKED Frame

   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:

   Data Limit:  A variable-length integer indicating the connection-
      level limit at which blocking occurred.

19.13.  STREAM_DATA_BLOCKED Frame

   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)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+




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   STREAM_DATA_BLOCKED frames contain the following fields:

   Stream ID:  A variable-length integer indicating the stream which is
      flow control blocked.

   Stream Data Limit:  A variable-length integer indicating the offset
      of the stream at which the blocking occurred.

19.14.  STREAMS_BLOCKED Frames

   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:

   Stream Limit:  A variable-length integer indicating the stream limit
      at the time the frame was sent.

19.15.  NEW_CONNECTION_ID Frame

   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:










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    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)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   Length (8)  |                                               |
   +-+-+-+-+-+-+-+-+       Connection ID (32..144)                 +
   |                                                             ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                                                               |
   +                   Stateless Reset Token (128)                 +
   |                                                               |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   NEW_CONNECTION_ID frames contain the following fields:

   Sequence Number:  The sequence number assigned to the connection ID
      by the sender.  See Section 5.1.1.

   Length:  An 8-bit unsigned integer containing the length of the
      connection ID.  Values less than 4 and greater than 18 are invalid
      and MUST be treated as a connection error of type
      PROTOCOL_VIOLATION.

   Connection ID:  A connection ID of the specified length.

   Stateless Reset Token:  A 128-bit value that will be used for a
      stateless reset when the associated connection ID is used (see
      Section 10.4).

   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.




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   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.

19.16.  RETIRE_CONNECTION_ID Frame

   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:

   Sequence Number:  The sequence number of the connection ID being
      retired.  See Section 5.1.2.

   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.






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19.17.  PATH_CHALLENGE Frame

   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:

   Data:  This 8-byte field contains arbitrary data.

   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.

19.18.  PATH_RESPONSE Frame

   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.

19.19.  CONNECTION_CLOSE Frames

   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.



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   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 (16)     |      [ Frame Type (i) ]     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Reason Phrase Length (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Reason Phrase (*)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   CONNECTION_CLOSE frames contain the following fields:

   Error Code:  A 16-bit error code which indicates the reason for
      closing this connection.  A CONNECTION_CLOSE frame of type 0x1c
      uses codes from the space defined in Section 20.  A
      CONNECTION_CLOSE frame of type 0x1d uses codes from the
      application protocol error code space, see Section 20.1

   Frame Type:  A variable-length integer encoding the type of frame
      that triggered the error.  A value of 0 (equivalent to the mention
      of the PADDING frame) is used when the frame type is unknown.  The
      application-specific variant of CONNECTION_CLOSE (type 0x1d) does
      not include this field.

   Reason Phrase Length:  A variable-length integer specifying the
      length of the reason phrase in bytes.  Because a CONNECTION_CLOSE
      frame cannot be split between packets, any limits on packet size
      will also limit the space available for a reason phrase.

   Reason Phrase:  A human-readable explanation for why the connection
      was closed.  This can be zero length if the sender chooses to not
      give details beyond the Error Code.  This SHOULD be a UTF-8
      encoded string [RFC3629].

19.20.  Extension Frames

   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




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   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.

20.  Transport Error Codes

   QUIC error codes are 16-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.

   NO_ERROR (0x0):  An endpoint uses this with CONNECTION_CLOSE to
      signal that the connection is being closed abruptly in the absence
      of any error.

   INTERNAL_ERROR (0x1):  The endpoint encountered an internal error and
      cannot continue with the connection.

   SERVER_BUSY (0x2):  The server is currently busy and does not accept
      any new connections.

   FLOW_CONTROL_ERROR (0x3):  An endpoint received more data than it
      permitted in its advertised data limits (see Section 4).

   STREAM_LIMIT_ERROR (0x4):  An endpoint received a frame for a stream
      identifier that exceeded its advertised stream limit for the
      corresponding stream type.

   STREAM_STATE_ERROR (0x5):  An endpoint received a frame for a stream
      that was not in a state that permitted that frame (see Section 3).

   FINAL_SIZE_ERROR (0x6):  An endpoint received a STREAM frame
      containing data that exceeded the previously established final
      size.  Or an endpoint received a STREAM frame or a RESET_STREAM
      frame containing a final size that was lower than the size of
      stream data that was already received.  Or an endpoint received a
      STREAM frame or a RESET_STREAM frame containing a different final
      size to the one already established.





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   FRAME_ENCODING_ERROR (0x7):  An endpoint received a frame that was
      badly formatted.  For instance, a frame of an unknown type, or an
      ACK frame that has more acknowledgment ranges than the remainder
      of the packet could carry.

   TRANSPORT_PARAMETER_ERROR (0x8):  An endpoint received transport
      parameters that were badly formatted, included an invalid value,
      was absent even though it is mandatory, was present though it is
      forbidden, or is otherwise in error.

   PROTOCOL_VIOLATION (0xA):  An endpoint detected an error with
      protocol compliance that was not covered by more specific error
      codes.

   INVALID_MIGRATION (0xC):  A peer has migrated to a different network
      when the endpoint had disabled migration.

   CRYPTO_ERROR (0x1XX):  The cryptographic handshake failed.  A range
      of 256 values is reserved for carrying error codes specific to the
      cryptographic handshake that is used.  Codes for errors occurring
      when TLS is used for the crypto handshake are described in
      Section 4.8 of [QUIC-TLS].

   See Section 22.3 for details of registering new error codes.

20.1.  Application Protocol Error Codes

   Application protocol error codes are 16-bit unsigned integers, but
   the management of application error codes are left to application
   protocols.  Application protocol error codes are used for the
   RESET_STREAM frame (Section 19.4) and the CONNECTION_CLOSE frame with
   a type of 0x1d (Section 19.19).

21.  Security Considerations

21.1.  Handshake Denial of Service

   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.



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   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.

21.2.  Amplification Attack

   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).



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21.3.  Optimistic ACK Attack

   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).

21.4.  Slowloris Attacks

   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.

21.5.  Stream Fragmentation and Reassembly Attacks

   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,




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   delaying reassembly of STREAM frames, implementing heuristics based
   on the age and duration of reassembly holes, or some combination.

21.6.  Stream Commitment Attack

   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, transmission error may cause STREAM DATA frames opening
   streams to be received out of sequence.  A receiver is obligated to
   open intervening streams if a higher-numbered stream ID is received.
   Thus, on a new connection, opening stream 2000001 opens 1 million
   streams, as required by the specification.

   The number of active streams is limited by the concurrent stream
   limit transport parameter, as explained in Section 4.5.  If chosen
   judiciously, this limit mitigates 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.

21.7.  Explicit Congestion Notification Attacks

   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 receivers ignore ECN codepoints set in
   duplicate packets (see Section 13.3).

21.8.  Stateless Reset Oracle

   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



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   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.

21.9.  Version Downgrade

   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.

22.  IANA Considerations

22.1.  QUIC Transport Parameter Registry

   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:

   Value:  The numeric value of the assignment (registrations will be
      between 0x0000 and 0xfeff).

   Parameter Name:  A short mnemonic for the parameter.




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   Specification:  A reference to a publicly available specification for
      the value.

   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 18.1  |
     |        |                                     |               |
     | 0x0001 | idle_timeout                        | Section 18.1  |
     |        |                                     |               |
     | 0x0002 | stateless_reset_token               | Section 18.1  |
     |        |                                     |               |
     | 0x0003 | max_packet_size                     | Section 18.1  |
     |        |                                     |               |
     | 0x0004 | initial_max_data                    | Section 18.1  |
     |        |                                     |               |
     | 0x0005 | initial_max_stream_data_bidi_local  | Section 18.1  |
     |        |                                     |               |
     | 0x0006 | initial_max_stream_data_bidi_remote | Section 18.1  |
     |        |                                     |               |
     | 0x0007 | initial_max_stream_data_uni         | Section 18.1  |
     |        |                                     |               |
     | 0x0008 | initial_max_streams_bidi            | Section 18.1  |
     |        |                                     |               |
     | 0x0009 | initial_max_streams_uni             | Section 18.1  |
     |        |                                     |               |
     | 0x000a | ack_delay_exponent                  | Section 18.1  |
     |        |                                     |               |
     | 0x000b | max_ack_delay                       | Section 18.1  |
     |        |                                     |               |
     | 0x000c | disable_migration                   | Section 18.1  |
     |        |                                     |               |
     | 0x000d | preferred_address                   | Section 18.1  |
     +--------+-------------------------------------+---------------+

            Table 6: Initial QUIC Transport Parameters Entries







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22.2.  QUIC Frame Type Registry

   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:

   Value:  The numeric value of the assignment (registrations will be
      between 0x00 and 0x3fff).  A range of values MAY be assigned.

   Frame Name:  A short mnemonic for the frame type.

   Specification:  A reference to a publicly available specification for
      the value.

   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.

22.3.  QUIC Transport Error Codes Registry

   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 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].




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   Registrations MUST include the following fields:

   Value:  The numeric value of the assignment (registrations will be
      between 0x0000 and 0xfeff).

   Code:  A short mnemonic for the parameter.

   Description:  A brief description of the error code semantics, which
      MAY be a summary if a specification reference is provided.

   Specification:  A reference to a publicly available specification for
      the value.

   The initial contents of this registry are shown in Table 7.  Values
   from 0xFF00 to 0xFFFF are reserved for Private Use [RFC8126].




































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   +------+---------------------------+----------------+---------------+
   | Valu | Error                     | Description    | Specification |
   | e    |                           |                |               |
   +------+---------------------------+----------------+---------------+
   | 0x0  | NO_ERROR                  | No error       | Section 20    |
   |      |                           |                |               |
   | 0x1  | INTERNAL_ERROR            | Implementation | Section 20    |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x2  | SERVER_BUSY               | Server         | Section 20    |
   |      |                           | currently busy |               |
   |      |                           |                |               |
   | 0x3  | FLOW_CONTROL_ERROR        | Flow control   | Section 20    |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x4  | STREAM_LIMIT_ERROR        | Too many       | Section 20    |
   |      |                           | streams opened |               |
   |      |                           |                |               |
   | 0x5  | STREAM_STATE_ERROR        | Frame received | Section 20    |
   |      |                           | in invalid     |               |
   |      |                           | stream state   |               |
   |      |                           |                |               |
   | 0x6  | FINAL_SIZE_ERROR          | Change to      | Section 20    |
   |      |                           | final size     |               |
   |      |                           |                |               |
   | 0x7  | FRAME_ENCODING_ERROR      | Frame encoding | Section 20    |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x8  | TRANSPORT_PARAMETER_ERROR | Error in       | Section 20    |
   |      |                           | transport      |               |
   |      |                           | parameters     |               |
   |      |                           |                |               |
   | 0xA  | PROTOCOL_VIOLATION        | Generic        | Section 20    |
   |      |                           | protocol       |               |
   |      |                           | violation      |               |
   |      |                           |                |               |
   | 0xC  | INVALID_MIGRATION         | Violated       | Section 20    |
   |      |                           | disabled       |               |
   |      |                           | migration      |               |
   +------+---------------------------+----------------+---------------+

            Table 7: Initial QUIC Transport Error Codes Entries

23.  References







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23.1.  Normative References

   [DPLPMTUD]
              Fairhurst, G., Jones, T., Tuexen, M., and I. Ruengeler,
              "Packetization Layer Path MTU Discovery for Datagram
              Transports", draft-ietf-tsvwg-datagram-plpmtud-06 (work in
              progress), November 2018.

   [QUIC-RECOVERY]
              Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
              and Congestion Control", draft-ietf-quic-recovery (work in
              progress), February 2019.

   [QUIC-TLS]
              Thomson, M., Ed. and S. Turner, Ed., "Using Transport
              Layer Security (TLS) to Secure QUIC", draft-ietf-quic-tls
              (work in progress), February 2019.

   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,
              <https://www.rfc-editor.org/info/rfc1191>.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [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,
              <https://www.rfc-editor.org/info/rfc3168>.

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
              2003, <https://www.rfc-editor.org/info/rfc3629>.

   [RFC4086]  Eastlake 3rd, D., Schiller, J., and S. Crocker,
              "Randomness Requirements for Security", BCP 106, RFC 4086,
              DOI 10.17487/RFC4086, June 2005,
              <https://www.rfc-editor.org/info/rfc4086>.

   [RFC5116]  McGrew, D., "An Interface and Algorithms for Authenticated
              Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008,
              <https://www.rfc-editor.org/info/rfc5116>.







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   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
              "IPv6 Flow Label Specification", RFC 6437,
              DOI 10.17487/RFC6437, November 2011,
              <https://www.rfc-editor.org/info/rfc6437>.

   [RFC8085]  Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
              Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
              March 2017, <https://www.rfc-editor.org/info/rfc8085>.

   [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,
              <https://www.rfc-editor.org/info/rfc8126>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8201]  McCann, J., Deering, S., Mogul, J., and R. Hinden, Ed.,
              "Path MTU Discovery for IP version 6", STD 87, RFC 8201,
              DOI 10.17487/RFC8201, July 2017,
              <https://www.rfc-editor.org/info/rfc8201>.

   [RFC8311]  Black, D., "Relaxing Restrictions on Explicit Congestion
              Notification (ECN) Experimentation", RFC 8311,
              DOI 10.17487/RFC8311, January 2018,
              <https://www.rfc-editor.org/info/rfc8311>.

   [SPIN]     Trammell, B. and M. Kuehlewind, "The QUIC Latency Spin
              Bit", draft-ietf-quic-spin-exp-01 (work in progress),
              October 2018.

   [TLS13]    Rescorla, E., "The Transport Layer Security (TLS) Protocol
              Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
              <https://www.rfc-editor.org/info/rfc8446>.

23.2.  Informative References

   [EARLY-DESIGN]
              Roskind, J., "QUIC: Multiplexed Transport Over UDP",
              December 2013, <https://goo.gl/dMVtFi>.

   [HTTP2]    Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
              Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
              DOI 10.17487/RFC7540, May 2015,
              <https://www.rfc-editor.org/info/rfc7540>.





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   [QUIC-INVARIANTS]
              Thomson, M., "Version-Independent Properties of QUIC",
              draft-ietf-quic-invariants (work in progress), February
              2019.

   [RFC1812]  Baker, F., Ed., "Requirements for IP Version 4 Routers",
              RFC 1812, DOI 10.17487/RFC1812, June 1995,
              <https://www.rfc-editor.org/info/rfc1812>.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018,
              DOI 10.17487/RFC2018, October 1996,
              <https://www.rfc-editor.org/info/rfc2018>.

   [RFC2104]  Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
              Hashing for Message Authentication", RFC 2104,
              DOI 10.17487/RFC2104, February 1997,
              <https://www.rfc-editor.org/info/rfc2104>.

   [RFC2360]  Scott, G., "Guide for Internet Standards Writers", BCP 22,
              RFC 2360, DOI 10.17487/RFC2360, June 1998,
              <https://www.rfc-editor.org/info/rfc2360>.

   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",
              RFC 4303, DOI 10.17487/RFC4303, December 2005,
              <https://www.rfc-editor.org/info/rfc4303>.

   [RFC4443]  Conta, A., Deering, S., and M. Gupta, Ed., "Internet
              Control Message Protocol (ICMPv6) for the Internet
              Protocol Version 6 (IPv6) Specification", STD 89,
              RFC 4443, DOI 10.17487/RFC4443, March 2006,
              <https://www.rfc-editor.org/info/rfc4443>.

   [RFC4787]  Audet, F., Ed. and C. Jennings, "Network Address
              Translation (NAT) Behavioral Requirements for Unicast
              UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
              2007, <https://www.rfc-editor.org/info/rfc4787>.

   [RFC5869]  Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand
              Key Derivation Function (HKDF)", RFC 5869,
              DOI 10.17487/RFC5869, May 2010,
              <https://www.rfc-editor.org/info/rfc5869>.

   [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, <https://www.rfc-editor.org/info/rfc7301>.




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   [RFC8200]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", STD 86, RFC 8200,
              DOI 10.17487/RFC8200, July 2017,
              <https://www.rfc-editor.org/info/rfc8200>.

   [SLOWLORIS]
              RSnake Hansen, R., "Welcome to Slowloris...", June 2009,
              <https://web.archive.org/web/20150315054838/
              http://ha.ckers.org/slowloris/>.

Appendix A.  Sample Packet Number Decoding Algorithm

   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

Appendix B.  Change Log

      *RFC Editor's Note:* Please remove this section prior to
      publication of a final version of this document.

   Issue and pull request numbers are listed with a leading octothorp.





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B.1.  Since draft-ietf-quic-transport-17

   o  Stream-related errors now use STREAM_STATE_ERROR (#2305)

   o  Endpoints discard initial keys as soon as handshake keys are
      available (#1951, #2045)

   o  Expanded conditions for ignoring ICMP packet too big messages
      (#2108, #2161)

   o  Remove rate control from PATH_CHALLENGE/PATH_RESPONSE (#2129,
      #2241)

   o  Endpoints are permitted to discard malformed initial packets
      (#2141)

   o  Clarified ECN implementation and usage requirements (#2156, #2201)

   o  Disable ECN count verification for packets that arrive out of
      order (#2198, #2215)

   o  Use Probe Timeout (PTO) instead of RTO (#2206, #2238)

   o  Loosen constraints on retransmission of ACK ranges (#2199, #2245)

   o  Limit Retry and Version Negotiation to once per datagram (#2259,
      #2303)

   o  Set a maximum value for max_ack_delay transport parameter (#2282,
      #2301)

   o  Allow server preferred address for both IPv4 and IPv6 (#2122,
      #2296)

   o  Corrected requirements for migration to a preferred address
      (#2146, #2349)

B.2.  Since draft-ietf-quic-transport-16

   o  Stream limits are defined as counts, not maximums (#1850, #1906)

   o  Require amplification attack defense after closing (#1905, #1911)

   o  Remove reservation of application error code 0 for STOPPING
      (#1804, #1922)

   o  Renumbered frames (#1945)




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   o  Renumbered transport parameters (#1946)

   o  Numeric transport parameters are expressed as varints (#1608,
      #1947, #1955)

   o  Reorder the NEW_CONNECTION_ID frame (#1952, #1963)

   o  Rework the first byte (#2006)

      *  Fix the 0x40 bit

      *  Change type values for long header

      *  Add spin bit to short header (#631, #1988)

      *  Encrypt the remainder of the first byte (#1322)

      *  Move packet number length to first byte

      *  Move ODCIL to first byte of retry packets

      *  Simplify packet number protection (#1575)

   o  Allow STOP_SENDING to open a remote bidirectional stream (#1797,
      #2013)

   o  Added mitigation for off-path migration attacks (#1278, #1749,
      #2033)

   o  Don't let the PMTU to drop below 1280 (#2063, #2069)

   o  Require peers to replace retired connection IDs (#2085)

   o  Servers are required to ignore Version Negotiation packets (#2088)

   o  Tokens are repeated in all Initial packets (#2089)

   o  Clarified how PING frames are sent after loss (#2094)

   o  Initial keys are discarded once Handshake are available (#1951,
      #2045)

   o  ICMP PTB validation clarifications (#2161, #2109, #2108)








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B.3.  Since draft-ietf-quic-transport-15

   Substantial editorial reorganization; no technical changes.

B.4.  Since draft-ietf-quic-transport-14

   o  Merge ACK and ACK_ECN (#1778, #1801)

   o  Explicitly communicate max_ack_delay (#981, #1781)

   o  Validate original connection ID after Retry packets (#1710, #1486,
      #1793)

   o  Idle timeout is optional and has no specified maximum (#1765)

   o  Update connection ID handling; add RETIRE_CONNECTION_ID type
      (#1464, #1468, #1483, #1484, #1486, #1495, #1729, #1742, #1799,
      #1821)

   o  Include a Token in all Initial packets (#1649, #1794)

   o  Prevent handshake deadlock (#1764, #1824)

B.5.  Since draft-ietf-quic-transport-13

   o  Streams open when higher-numbered streams of the same type open
      (#1342, #1549)

   o  Split initial stream flow control limit into 3 transport
      parameters (#1016, #1542)

   o  All flow control transport parameters are optional (#1610)

   o  Removed UNSOLICITED_PATH_RESPONSE error code (#1265, #1539)

   o  Permit stateless reset in response to any packet (#1348, #1553)

   o  Recommended defense against stateless reset spoofing (#1386,
      #1554)

   o  Prevent infinite stateless reset exchanges (#1443, #1627)

   o  Forbid processing of the same packet number twice (#1405, #1624)

   o  Added a packet number decoding example (#1493)

   o  More precisely define idle timeout (#1429, #1614, #1652)




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   o  Corrected format of Retry packet and prevented looping (#1492,
      #1451, #1448, #1498)

   o  Permit 0-RTT after receiving Version Negotiation or Retry (#1507,
      #1514, #1621)

   o  Permit Retry in response to 0-RTT (#1547, #1552)

   o  Looser verification of ECN counters to account for ACK loss
      (#1555, #1481, #1565)

   o  Remove frame type field from APPLICATION_CLOSE (#1508, #1528)

B.6.  Since draft-ietf-quic-transport-12

   o  Changes to integration of the TLS handshake (#829, #1018, #1094,
      #1165, #1190, #1233, #1242, #1252, #1450, #1458)

      *  The cryptographic handshake uses CRYPTO frames, not stream 0

      *  QUIC packet protection is used in place of TLS record
         protection

      *  Separate QUIC packet number spaces are used for the handshake

      *  Changed Retry to be independent of the cryptographic handshake

      *  Added NEW_TOKEN frame and Token fields to Initial packet

      *  Limit the use of HelloRetryRequest to address TLS needs (like
         key shares)

   o  Enable server to transition connections to a preferred address
      (#560, #1251, #1373)

   o  Added ECN feedback mechanisms and handling; new ACK_ECN frame
      (#804, #805, #1372)

   o  Changed rules and recommendations for use of new connection IDs
      (#1258, #1264, #1276, #1280, #1419, #1452, #1453, #1465)

   o  Added a transport parameter to disable intentional connection
      migration (#1271, #1447)

   o  Packets from different connection ID can't be coalesced (#1287,
      #1423)





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   o  Fixed sampling method for packet number encryption; the length
      field in long headers includes the packet number field in addition
      to the packet payload (#1387, #1389)

   o  Stateless Reset is now symmetric and subject to size constraints
      (#466, #1346)

   o  Added frame type extension mechanism (#58, #1473)

B.7.  Since draft-ietf-quic-transport-11

   o  Enable server to transition connections to a preferred address
      (#560, #1251)

   o  Packet numbers are encrypted (#1174, #1043, #1048, #1034, #850,
      #990, #734, #1317, #1267, #1079)

   o  Packet numbers use a variable-length encoding (#989, #1334)

   o  STREAM frames can now be empty (#1350)

B.8.  Since draft-ietf-quic-transport-10

   o  Swap payload length and packed number fields in long header
      (#1294)

   o  Clarified that CONNECTION_CLOSE is allowed in Handshake packet
      (#1274)

   o  Spin bit reserved (#1283)

   o  Coalescing multiple QUIC packets in a UDP datagram (#1262, #1285)

   o  A more complete connection migration (#1249)

   o  Refine opportunistic ACK defense text (#305, #1030, #1185)

   o  A Stateless Reset Token isn't mandatory (#818, #1191)

   o  Removed implicit stream opening (#896, #1193)

   o  An empty STREAM frame can be used to open a stream without sending
      data (#901, #1194)

   o  Define stream counts in transport parameters rather than a maximum
      stream ID (#1023, #1065)

   o  STOP_SENDING is now prohibited before streams are used (#1050)



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   o  Recommend including ACK in Retry packets and allow PADDING (#1067,
      #882)

   o  Endpoints now become closing after an idle timeout (#1178, #1179)

   o  Remove implication that Version Negotiation is sent when a packet
      of the wrong version is received (#1197)

B.9.  Since draft-ietf-quic-transport-09

   o  Added PATH_CHALLENGE and PATH_RESPONSE frames to replace PING with
      Data and PONG frame.  Changed ACK frame type from 0x0e to 0x0d.
      (#1091, #725, #1086)

   o  A server can now only send 3 packets without validating the client
      address (#38, #1090)

   o  Delivery order of stream data is no longer strongly specified
      (#252, #1070)

   o  Rework of packet handling and version negotiation (#1038)

   o  Stream 0 is now exempt from flow control until the handshake
      completes (#1074, #725, #825, #1082)

   o  Improved retransmission rules for all frame types: information is
      retransmitted, not packets or frames (#463, #765, #1095, #1053)

   o  Added an error code for server busy signals (#1137)

   o  Endpoints now set the connection ID that their peer uses.
      Connection IDs are variable length.  Removed the
      omit_connection_id transport parameter and the corresponding short
      header flag. (#1089, #1052, #1146, #821, #745, #821, #1166, #1151)

B.10.  Since draft-ietf-quic-transport-08

   o  Clarified requirements for BLOCKED usage (#65, #924)

   o  BLOCKED frame now includes reason for blocking (#452, #924, #927,
      #928)

   o  GAP limitation in ACK Frame (#613)

   o  Improved PMTUD description (#614, #1036)

   o  Clarified stream state machine (#634, #662, #743, #894)




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   o  Reserved versions don't need to be generated deterministically
      (#831, #931)

   o  You don't always need the draining period (#871)

   o  Stateless reset clarified as version-specific (#930, #986)

   o  initial_max_stream_id_x transport parameters are optional (#970,
      #971)

   o  Ack Delay assumes a default value during the handshake (#1007,
      #1009)

   o  Removed transport parameters from NewSessionTicket (#1015)

B.11.  Since draft-ietf-quic-transport-07

   o  The long header now has version before packet number (#926, #939)

   o  Rename and consolidate packet types (#846, #822, #847)

   o  Packet types are assigned new codepoints and the Connection ID
      Flag is inverted (#426, #956)

   o  Removed type for Version Negotiation and use Version 0 (#963,
      #968)

   o  Streams are split into unidirectional and bidirectional (#643,
      #656, #720, #872, #175, #885)

      *  Stream limits now have separate uni- and bi-directional
         transport parameters (#909, #958)

      *  Stream limit transport parameters are now optional and default
         to 0 (#970, #971)

   o  The stream state machine has been split into read and write (#634,
      #894)

   o  Employ variable-length integer encodings throughout (#595)

   o  Improvements to connection close

      *  Added distinct closing and draining states (#899, #871)

      *  Draining period can terminate early (#869, #870)

      *  Clarifications about stateless reset (#889, #890)



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   o  Address validation for connection migration (#161, #732, #878)

   o  Clearly defined retransmission rules for BLOCKED (#452, #65, #924)

   o  negotiated_version is sent in server transport parameters (#710,
      #959)

   o  Increased the range over which packet numbers are randomized
      (#864, #850, #964)

B.12.  Since draft-ietf-quic-transport-06

   o  Replaced FNV-1a with AES-GCM for all "Cleartext" packets (#554)

   o  Split error code space between application and transport (#485)

   o  Stateless reset token moved to end (#820)

   o  1-RTT-protected long header types removed (#848)

   o  No acknowledgments during draining period (#852)

   o  Remove "application close" as a separate close type (#854)

   o  Remove timestamps from the ACK frame (#841)

   o  Require transport parameters to only appear once (#792)

B.13.  Since draft-ietf-quic-transport-05

   o  Stateless token is server-only (#726)

   o  Refactor section on connection termination (#733, #748, #328,
      #177)

   o  Limit size of Version Negotiation packet (#585)

   o  Clarify when and what to ack (#736)

   o  Renamed STREAM_ID_NEEDED to STREAM_ID_BLOCKED

   o  Clarify Keep-alive requirements (#729)

B.14.  Since draft-ietf-quic-transport-04

   o  Introduce STOP_SENDING frame, RESET_STREAM only resets in one
      direction (#165)




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   o  Removed GOAWAY; application protocols are responsible for graceful
      shutdown (#696)

   o  Reduced the number of error codes (#96, #177, #184, #211)

   o  Version validation fields can't move or change (#121)

   o  Removed versions from the transport parameters in a
      NewSessionTicket message (#547)

   o  Clarify the meaning of "bytes in flight" (#550)

   o  Public reset is now stateless reset and not visible to the path
      (#215)

   o  Reordered bits and fields in STREAM frame (#620)

   o  Clarifications to the stream state machine (#572, #571)

   o  Increased the maximum length of the Largest Acknowledged field in
      ACK frames to 64 bits (#629)

   o  truncate_connection_id is renamed to omit_connection_id (#659)

   o  CONNECTION_CLOSE terminates the connection like TCP RST (#330,
      #328)

   o  Update labels used in HKDF-Expand-Label to match TLS 1.3 (#642)

B.15.  Since draft-ietf-quic-transport-03

   o  Change STREAM and RESET_STREAM layout

   o  Add MAX_STREAM_ID settings

B.16.  Since draft-ietf-quic-transport-02

   o  The size of the initial packet payload has a fixed minimum (#267,
      #472)

   o  Define when Version Negotiation packets are ignored (#284, #294,
      #241, #143, #474)

   o  The 64-bit FNV-1a algorithm is used for integrity protection of
      unprotected packets (#167, #480, #481, #517)

   o  Rework initial packet types to change how the connection ID is
      chosen (#482, #442, #493)



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   o  No timestamps are forbidden in unprotected packets (#542, #429)

   o  Cryptographic handshake is now on stream 0 (#456)

   o  Remove congestion control exemption for cryptographic handshake
      (#248, #476)

   o  Version 1 of QUIC uses TLS; a new version is needed to use a
      different handshake protocol (#516)

   o  STREAM frames have a reduced number of offset lengths (#543, #430)

   o  Split some frames into separate connection- and stream- level
      frames (#443)

      *  WINDOW_UPDATE split into MAX_DATA and MAX_STREAM_DATA (#450)

      *  BLOCKED split to match WINDOW_UPDATE split (#454)

      *  Define STREAM_ID_NEEDED frame (#455)

   o  A NEW_CONNECTION_ID frame supports connection migration without
      linkability (#232, #491, #496)

   o  Transport parameters for 0-RTT are retained from a previous
      connection (#405, #513, #512)

      *  A client in 0-RTT no longer required to reset excess streams
         (#425, #479)

   o  Expanded security considerations (#440, #444, #445, #448)

B.17.  Since draft-ietf-quic-transport-01

   o  Defined short and long packet headers (#40, #148, #361)

   o  Defined a versioning scheme and stable fields (#51, #361)

   o  Define reserved version values for "greasing" negotiation (#112,
      #278)

   o  The initial packet number is randomized (#35, #283)

   o  Narrow the packet number encoding range requirement (#67, #286,
      #299, #323, #356)

   o  Defined client address validation (#52, #118, #120, #275)




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   o  Define transport parameters as a TLS extension (#49, #122)

   o  SCUP and COPT parameters are no longer valid (#116, #117)

   o  Transport parameters for 0-RTT are either remembered from before,
      or assume default values (#126)

   o  The server chooses connection IDs in its final flight (#119, #349,
      #361)

   o  The server echoes the Connection ID and packet number fields when
      sending a Version Negotiation packet (#133, #295, #244)

   o  Defined a minimum packet size for the initial handshake packet
      from the client (#69, #136, #139, #164)

   o  Path MTU Discovery (#64, #106)

   o  The initial handshake packet from the client needs to fit in a
      single packet (#338)

   o  Forbid acknowledgment of packets containing only ACK and PADDING
      (#291)

   o  Require that frames are processed when packets are acknowledged
      (#381, #341)

   o  Removed the STOP_WAITING frame (#66)

   o  Don't require retransmission of old timestamps for lost ACK frames
      (#308)

   o  Clarified that frames are not retransmitted, but the information
      in them can be (#157, #298)

   o  Error handling definitions (#335)

   o  Split error codes into four sections (#74)

   o  Forbid the use of Public Reset where CONNECTION_CLOSE is possible
      (#289)

   o  Define packet protection rules (#336)

   o  Require that stream be entirely delivered or reset, including
      acknowledgment of all STREAM frames or the RESET_STREAM, before it
      closes (#381)




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   o  Remove stream reservation from state machine (#174, #280)

   o  Only stream 1 does not contribute to connection-level flow control
      (#204)

   o  Stream 1 counts towards the maximum concurrent stream limit (#201,
      #282)

   o  Remove connection-level flow control exclusion for some streams
      (except 1) (#246)

   o  RESET_STREAM affects connection-level flow control (#162, #163)

   o  Flow control accounting uses the maximum data offset on each
      stream, rather than bytes received (#378)

   o  Moved length-determining fields to the start of STREAM and ACK
      (#168, #277)

   o  Added the ability to pad between frames (#158, #276)

   o  Remove error code and reason phrase from GOAWAY (#352, #355)

   o  GOAWAY includes a final stream number for both directions (#347)

   o  Error codes for RESET_STREAM and CONNECTION_CLOSE are now at a
      consistent offset (#249)

   o  Defined priority as the responsibility of the application protocol
      (#104, #303)

B.18.  Since draft-ietf-quic-transport-00

   o  Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flag

   o  Defined versioning

   o  Reworked description of packet and frame layout

   o  Error code space is divided into regions for each component

   o  Use big endian for all numeric values

B.19.  Since draft-hamilton-quic-transport-protocol-01

   o  Adopted as base for draft-ietf-quic-tls

   o  Updated authors/editors list



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   o  Added IANA Considerations section

   o  Moved Contributors and Acknowledgments to appendices

Acknowledgments

   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.

Contributors

   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.

Authors' Addresses

   Jana Iyengar (editor)
   Fastly

   Email: jri.ietf@gmail.com


   Martin Thomson (editor)
   Mozilla

   Email: mt@lowentropy.net











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