draft-ietf-quic-recovery.txt

QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                           I. Swett, Ed.
Expires: 18 February 2021                                         Google
                                                          17 August 2020


               QUIC Loss Detection and Congestion Control
                    draft-ietf-quic-recovery-latest

Abstract

   This document describes loss detection and congestion control
   mechanisms for QUIC.

Note to Readers

   Discussion of this draft takes place on the QUIC working group
   mailing list (quic@ietf.org (mailto:quic@ietf.org)), which is
   archived at https://mailarchive.ietf.org/arch/
   search/?email_list=quic.

   Working Group information can be found at https://github.com/quicwg;
   source code and issues list for this draft can be found at
   https://github.com/quicwg/base-drafts/labels/-recovery.

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 18 February 2021.

Copyright Notice

   Copyright (c) 2020 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
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   extracted from this document must include Simplified BSD License text
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   provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Conventions and Definitions . . . . . . . . . . . . . . . . .   4
   3.  Design of the QUIC Transmission Machinery . . . . . . . . . .   5
   4.  Relevant Differences Between QUIC and TCP . . . . . . . . . .   5
     4.1.  Separate Packet Number Spaces . . . . . . . . . . . . . .   6
     4.2.  Monotonically Increasing Packet Numbers . . . . . . . . .   6
     4.3.  Clearer Loss Epoch  . . . . . . . . . . . . . . . . . . .   6
     4.4.  No Reneging . . . . . . . . . . . . . . . . . . . . . . .   7
     4.5.  More ACK Ranges . . . . . . . . . . . . . . . . . . . . .   7
     4.6.  Explicit Correction For Delayed Acknowledgements  . . . .   7
     4.7.  Probe Timeout Replaces RTO and TLP  . . . . . . . . . . .   7
     4.8.  The Minimum Congestion Window is Two Packets  . . . . . .   8
   5.  Estimating the Round-Trip Time  . . . . . . . . . . . . . . .   8
     5.1.  Generating RTT samples  . . . . . . . . . . . . . . . . .   8
     5.2.  Estimating min_rtt  . . . . . . . . . . . . . . . . . . .   9
     5.3.  Estimating smoothed_rtt and rttvar  . . . . . . . . . . .   9
   6.  Loss Detection  . . . . . . . . . . . . . . . . . . . . . . .  11
     6.1.  Acknowledgement-Based Detection . . . . . . . . . . . . .  11
       6.1.1.  Packet Threshold  . . . . . . . . . . . . . . . . . .  12
       6.1.2.  Time Threshold  . . . . . . . . . . . . . . . . . . .  12
     6.2.  Probe Timeout . . . . . . . . . . . . . . . . . . . . . .  13
       6.2.1.  Computing PTO . . . . . . . . . . . . . . . . . . . .  13
       6.2.2.  Handshakes and New Paths  . . . . . . . . . . . . . .  15
       6.2.3.  Speeding Up Handshake Completion  . . . . . . . . . .  16
       6.2.4.  Sending Probe Packets . . . . . . . . . . . . . . . .  16
     6.3.  Handling Retry Packets  . . . . . . . . . . . . . . . . .  17
     6.4.  Discarding Keys and Packet State  . . . . . . . . . . . .  18
   7.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .  18
     7.1.  Explicit Congestion Notification  . . . . . . . . . . . .  19
     7.2.  Initial and Minimum Congestion Window . . . . . . . . . .  19
     7.3.  Slow Start  . . . . . . . . . . . . . . . . . . . . . . .  19
     7.4.  Congestion Avoidance  . . . . . . . . . . . . . . . . . .  20
     7.5.  Recovery Period . . . . . . . . . . . . . . . . . . . . .  20
     7.6.  Ignoring Loss of Undecryptable Packets  . . . . . . . . .  20
     7.7.  Probe Timeout . . . . . . . . . . . . . . . . . . . . . .  21
     7.8.  Persistent Congestion . . . . . . . . . . . . . . . . . .  21
     7.9.  Pacing  . . . . . . . . . . . . . . . . . . . . . . . . .  23



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     7.10. Under-utilizing the Congestion Window . . . . . . . . . .  24
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  24
     8.1.  Congestion Signals  . . . . . . . . . . . . . . . . . . .  24
     8.2.  Traffic Analysis  . . . . . . . . . . . . . . . . . . . .  24
     8.3.  Misreporting ECN Markings . . . . . . . . . . . . . . . .  25
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  25
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  25
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  25
     10.2.  Informative References . . . . . . . . . . . . . . . . .  26
   Appendix A.  Loss Recovery Pseudocode . . . . . . . . . . . . . .  27
     A.1.  Tracking Sent Packets . . . . . . . . . . . . . . . . . .  27
       A.1.1.  Sent Packet Fields  . . . . . . . . . . . . . . . . .  28
     A.2.  Constants of Interest . . . . . . . . . . . . . . . . . .  28
     A.3.  Variables of interest . . . . . . . . . . . . . . . . . .  29
     A.4.  Initialization  . . . . . . . . . . . . . . . . . . . . .  29
     A.5.  On Sending a Packet . . . . . . . . . . . . . . . . . . .  30
     A.6.  On Receiving a Datagram . . . . . . . . . . . . . . . . .  30
     A.7.  On Receiving an Acknowledgment  . . . . . . . . . . . . .  31
     A.8.  Setting the Loss Detection Timer  . . . . . . . . . . . .  32
     A.9.  On Timeout  . . . . . . . . . . . . . . . . . . . . . . .  34
     A.10. Detecting Lost Packets  . . . . . . . . . . . . . . . . .  35
   Appendix B.  Congestion Control Pseudocode  . . . . . . . . . . .  36
     B.1.  Constants of interest . . . . . . . . . . . . . . . . . .  36
     B.2.  Variables of interest . . . . . . . . . . . . . . . . . .  37
     B.3.  Initialization  . . . . . . . . . . . . . . . . . . . . .  37
     B.4.  On Packet Sent  . . . . . . . . . . . . . . . . . . . . .  38
     B.5.  On Packet Acknowledgement . . . . . . . . . . . . . . . .  38
     B.6.  On New Congestion Event . . . . . . . . . . . . . . . . .  39
     B.7.  Process ECN Information . . . . . . . . . . . . . . . . .  39
     B.8.  On Packets Lost . . . . . . . . . . . . . . . . . . . . .  39
     B.9.  Upon dropping Initial or Handshake keys . . . . . . . . .  40
   Appendix C.  Change Log . . . . . . . . . . . . . . . . . . . . .  40
     C.1.  Since draft-ietf-quic-recovery-28 . . . . . . . . . . . .  41
     C.2.  Since draft-ietf-quic-recovery-27 . . . . . . . . . . . .  41
     C.3.  Since draft-ietf-quic-recovery-26 . . . . . . . . . . . .  41
     C.4.  Since draft-ietf-quic-recovery-25 . . . . . . . . . . . .  41
     C.5.  Since draft-ietf-quic-recovery-24 . . . . . . . . . . . .  41
     C.6.  Since draft-ietf-quic-recovery-23 . . . . . . . . . . . .  41
     C.7.  Since draft-ietf-quic-recovery-22 . . . . . . . . . . . .  42
     C.8.  Since draft-ietf-quic-recovery-21 . . . . . . . . . . . .  42
     C.9.  Since draft-ietf-quic-recovery-20 . . . . . . . . . . . .  42
     C.10. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . .  42
     C.11. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . .  43
     C.12. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . .  43
     C.13. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . .  44
     C.14. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . .  44
     C.15. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . .  44
     C.16. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . .  45



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     C.17. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . .  45
     C.18. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . .  45
     C.19. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . .  45
     C.20. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . .  45
     C.21. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . .  45
     C.22. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . .  46
     C.23. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . .  46
     C.24. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . .  46
     C.25. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . .  46
     C.26. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . .  46
     C.27. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . .  46
     C.28. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . .  46
     C.29. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . .  47
   Appendix D.  Contributors . . . . . . . . . . . . . . . . . . . .  47
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . .  47
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  47

1.  Introduction

   QUIC is a new multiplexed and secure transport protocol atop UDP,
   specified in [QUIC-TRANSPORT].  This document describes congestion
   control and loss recovery for QUIC.  Mechanisms described in this
   document follow the spirit of existing TCP congestion control and
   loss recovery mechanisms, described in RFCs, various Internet-drafts,
   or academic papers, and also those prevalent in TCP implementations.

2.  Conventions and Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

   Definitions of terms that are used in this document:

   Ack-eliciting frames:  All frames other than ACK, PADDING, and
      CONNECTION_CLOSE are considered ack-eliciting.

   Ack-eliciting packets:  Packets that contain ack-eliciting frames
      elicit an ACK from the receiver within the maximum acknowledgement
      delay and are called ack-eliciting packets.

   In-flight:  Packets are considered in-flight when they are ack-
      eliciting or contain a PADDING frame, and they have been sent but
      are not acknowledged, declared lost, or discarded along with old
      keys.




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3.  Design of the QUIC Transmission Machinery

   All transmissions in QUIC are sent with a packet-level header, which
   indicates the encryption level and includes a packet sequence number
   (referred to below as a packet number).  The encryption level
   indicates the packet number space, as described in [QUIC-TRANSPORT].
   Packet numbers never repeat within a packet number space for the
   lifetime of a connection.  Packet numbers are sent in monotonically
   increasing order within a space, preventing ambiguity.

   This design obviates the need for disambiguating between
   transmissions and retransmissions; this eliminates significant
   complexity from QUIC's interpretation of TCP loss detection
   mechanisms.

   QUIC packets can contain multiple frames of different types.  The
   recovery mechanisms ensure that data and frames that need reliable
   delivery are acknowledged or declared lost and sent in new packets as
   necessary.  The types of frames contained in a packet affect recovery
   and congestion control logic:

   *  All packets are acknowledged, though packets that contain no ack-
      eliciting frames are only acknowledged along with ack-eliciting
      packets.

   *  Long header packets that contain CRYPTO frames are critical to the
      performance of the QUIC handshake and use shorter timers for
      acknowledgement.

   *  Packets containing frames besides ACK or CONNECTION_CLOSE frames
      count toward congestion control limits and are considered in-
      flight.

   *  PADDING frames cause packets to contribute toward bytes in flight
      without directly causing an acknowledgment to be sent.

4.  Relevant Differences Between QUIC and TCP

   Readers familiar with TCP's loss detection and congestion control
   will find algorithms here that parallel well-known TCP ones.
   However, protocol differences between QUIC and TCP contribute to
   algorithmic differences.  These protocol differences are briefly
   described below.








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4.1.  Separate Packet Number Spaces

   QUIC uses separate packet number spaces for each encryption level,
   except 0-RTT and all generations of 1-RTT keys use the same packet
   number space.  Separate packet number spaces ensures acknowledgement
   of packets sent with one level of encryption will not cause spurious
   retransmission of packets sent with a different encryption level.
   Congestion control and round-trip time (RTT) measurement are unified
   across packet number spaces.

4.2.  Monotonically Increasing Packet Numbers

   TCP conflates transmission order at the sender with delivery order at
   the receiver, which results in retransmissions of the same data
   carrying the same sequence number, and consequently leads to
   "retransmission ambiguity".  QUIC separates the two.  QUIC uses a
   packet number to indicate transmission order.  Application data is
   sent in one or more streams and delivery order is determined by
   stream offsets encoded within STREAM frames.

   QUIC's packet number is strictly increasing within a packet number
   space, and directly encodes transmission order.  A higher packet
   number signifies that the packet was sent later, and a lower packet
   number signifies that the packet was sent earlier.  When a packet
   containing ack-eliciting frames is detected lost, QUIC includes
   necessary frames in a new packet with a new packet number, removing
   ambiguity about which packet is acknowledged when an ACK is received.
   Consequently, more accurate RTT measurements can be made, spurious
   retransmissions are trivially detected, and mechanisms such as Fast
   Retransmit can be applied universally, based only on packet number.

   This design point significantly simplifies loss detection mechanisms
   for QUIC.  Most TCP mechanisms implicitly attempt to infer
   transmission ordering based on TCP sequence numbers - a non-trivial
   task, especially when TCP timestamps are not available.

4.3.  Clearer Loss Epoch

   QUIC starts a loss epoch when a packet is lost and ends one when any
   packet sent after the epoch starts is acknowledged.  TCP waits for
   the gap in the sequence number space to be filled, and so if a
   segment is lost multiple times in a row, the loss epoch may not end
   for several round trips.  Because both should reduce their congestion
   windows only once per epoch, QUIC will do it once for every round
   trip that experiences loss, while TCP may only do it once across
   multiple round trips.





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4.4.  No Reneging

   QUIC ACKs contain information that is similar to TCP SACK, but QUIC
   does not allow any acknowledged packet to be reneged, greatly
   simplifying implementations on both sides and reducing memory
   pressure on the sender.

4.5.  More ACK Ranges

   QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges.  In
   high loss environments, this speeds recovery, reduces spurious
   retransmits, and ensures forward progress without relying on
   timeouts.

4.6.  Explicit Correction For Delayed Acknowledgements

   QUIC endpoints measure the delay incurred between when a packet is
   received and when the corresponding acknowledgment is sent, allowing
   a peer to maintain a more accurate round-trip time estimate; see
   Section 13.2 of [QUIC-TRANSPORT].

4.7.  Probe Timeout Replaces RTO and TLP

   QUIC uses a probe timeout (PTO; see Section 6.2), with a timer based
   on TCP's RTO computation.  QUIC's PTO includes the peer's maximum
   expected acknowledgement delay instead of using a fixed minimum
   timeout.  QUIC does not collapse the congestion window until
   persistent congestion (Section 7.8) is declared, unlike TCP, which
   collapses the congestion window upon expiry of an RTO.  Instead of
   collapsing the congestion window and declaring everything in-flight
   lost, QUIC allows probe packets to temporarily exceed the congestion
   window whenever the timer expires.

   In doing this, QUIC avoids unnecessary congestion window reductions,
   obviating the need for correcting mechanisms such as F-RTO
   ([RFC5682]).  Since QUIC does not collapse the congestion window on a
   PTO expiration, a QUIC sender is not limited from sending more in-
   flight packets after a PTO expiration if it still has available
   congestion window.  This occurs when a sender is application-limited
   and the PTO timer expires.  This is more aggressive than TCP's RTO
   mechanism when application-limited, but identical when not
   application-limited.

   A single packet loss at the tail does not indicate persistent
   congestion, so QUIC specifies a time-based definition to ensure one
   or more packets are sent prior to a dramatic decrease in congestion
   window; see Section 7.8.




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4.8.  The Minimum Congestion Window is Two Packets

   TCP uses a minimum congestion window of one packet.  However, loss of
   that single packet means that the sender needs to waiting for a PTO
   (Section 6.2) to recover, which can be much longer than a round-trip
   time.  Sending a single ack-eliciting packet also increases the
   chances of incurring additional latency when a receiver delays its
   acknowledgement.

   QUIC therefore recommends that the minimum congestion window be two
   packets.  While this increases network load, it is considered safe,
   since the sender will still reduce its sending rate exponentially
   under persistent congestion (Section 6.2).

5.  Estimating the Round-Trip Time

   At a high level, an endpoint measures the time from when a packet was
   sent to when it is acknowledged as a round-trip time (RTT) sample.
   The endpoint uses RTT samples and peer-reported host delays (see
   Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical
   description of the network path's RTT.  An endpoint computes the
   following three values for each path: the minimum value observed over
   the lifetime of the path (min_rtt), an exponentially-weighted moving
   average (smoothed_rtt), and the mean deviation (referred to as
   "variation" in the rest of this document) in the observed RTT samples
   (rttvar).

5.1.  Generating RTT samples

   An endpoint generates an RTT sample on receiving an ACK frame that
   meets the following two conditions:

   *  the largest acknowledged packet number is newly acknowledged, and

   *  at least one of the newly acknowledged packets was ack-eliciting.

   The RTT sample, latest_rtt, is generated as the time elapsed since
   the largest acknowledged packet was sent:

   latest_rtt = ack_time - send_time_of_largest_acked

   An RTT sample is generated using only the largest acknowledged packet
   in the received ACK frame.  This is because a peer reports
   acknowledgment delays for only the largest acknowledged packet in an
   ACK frame.  While the reported acknowledgment delay is not used by
   the RTT sample measurement, it is used to adjust the RTT sample in
   subsequent computations of smoothed_rtt and rttvar (Section 5.3).




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   To avoid generating multiple RTT samples for a single packet, an ACK
   frame SHOULD NOT be used to update RTT estimates if it does not newly
   acknowledge the largest acknowledged packet.

   An RTT sample MUST NOT be generated on receiving an ACK frame that
   does not newly acknowledge at least one ack-eliciting packet.  A peer
   usually does not send an ACK frame when only non-ack-eliciting
   packets are received.  Therefore an ACK frame that contains
   acknowledgements for only non-ack-eliciting packets could include an
   arbitrarily large ACK Delay value.  Ignoring such ACK frames avoids
   complications in subsequent smoothed_rtt and rttvar computations.

   A sender might generate multiple RTT samples per RTT when multiple
   ACK frames are received within an RTT.  As suggested in [RFC6298],
   doing so might result in inadequate history in smoothed_rtt and
   rttvar.  Ensuring that RTT estimates retain sufficient history is an
   open research question.

5.2.  Estimating min_rtt

   min_rtt is the minimum RTT observed for a given network path.
   min_rtt is set to the latest_rtt on the first RTT sample, and to the
   lesser of min_rtt and latest_rtt on subsequent samples.  In this
   document, min_rtt is used by loss detection to reject implausibly
   small rtt samples.

   An endpoint uses only locally observed times in computing the min_rtt
   and does not adjust for acknowledgment delays reported by the peer.
   Doing so allows the endpoint to set a lower bound for the
   smoothed_rtt based entirely on what it observes (see Section 5.3),
   and limits potential underestimation due to erroneously-reported
   delays by the peer.

   The RTT for a network path may change over time.  If a path's actual
   RTT decreases, the min_rtt will adapt immediately on the first low
   sample.  If the path's actual RTT increases, the min_rtt will not
   adapt to it, allowing future RTT samples that are smaller than the
   new RTT to be included in smoothed_rtt.

5.3.  Estimating smoothed_rtt and rttvar

   smoothed_rtt is an exponentially-weighted moving average of an
   endpoint's RTT samples, and rttvar is the variation in the RTT
   samples, estimated using a mean variation.

   The calculation of smoothed_rtt uses path latency after adjusting RTT
   samples for acknowledgement delays.  These delays are computed using
   the ACK Delay field of the ACK frame as described in Section 19.3 of



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   [QUIC-TRANSPORT].  For packets sent in the Application Data packet
   number space, a peer limits any delay in sending an acknowledgement
   for an ack-eliciting packet to no greater than the value it
   advertised in the max_ack_delay transport parameter.  Consequently,
   when a peer reports an acknowledgment delay that is greater than its
   max_ack_delay, the delay is attributed to reasons out of the peer's
   control, such as scheduler latency at the peer or loss of previous
   ACK frames.  Any delays beyond the peer's max_ack_delay are therefore
   considered effectively part of path delay and incorporated into the
   smoothed_rtt estimate.

   When adjusting an RTT sample using peer-reported acknowledgement
   delays, an endpoint:

   *  MUST ignore the ACK Delay field of the ACK frame for packets sent
      in the Initial and Handshake packet number space.

   *  MUST use the lesser of the value reported in ACK Delay field of
      the ACK frame and the peer's max_ack_delay transport parameter.

   *  MUST NOT apply the adjustment if the resulting RTT sample is
      smaller than the min_rtt.  This limits the underestimation that a
      misreporting peer can cause to the smoothed_rtt.

   smoothed_rtt and rttvar are computed as follows, similar to
   [RFC6298].

   When there are no samples for a network path, and on the first RTT
   sample for the network path:

   smoothed_rtt = rtt_sample
   rttvar = rtt_sample / 2

   Before any RTT samples are available, the initial RTT is used as
   rtt_sample.  On the first RTT sample for the network path, that
   sample is used as rtt_sample.  This ensures that the first
   measurement erases the history of any persisted or default values.

   On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:

   ack_delay = acknowledgement delay from ACK frame
   ack_delay = min(ack_delay, max_ack_delay)
   adjusted_rtt = latest_rtt
   if (min_rtt + ack_delay < latest_rtt):
     adjusted_rtt = latest_rtt - ack_delay
   smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
   rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
   rttvar = 3/4 * rttvar + 1/4 * rttvar_sample



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6.  Loss Detection

   QUIC senders use acknowledgements to detect lost packets, and a probe
   time out (see Section 6.2) to ensure acknowledgements are received.
   This section provides a description of these algorithms.

   If a packet is lost, the QUIC transport needs to recover from that
   loss, such as by retransmitting the data, sending an updated frame,
   or discarding the frame.  For more information, see Section 13.3 of
   [QUIC-TRANSPORT].

   Loss detection is separate per packet number space, unlike RTT
   measurement and congestion control, because RTT and congestion
   control are properties of the path, whereas loss detection also
   relies upon key availability.

6.1.  Acknowledgement-Based Detection

   Acknowledgement-based loss detection implements the spirit of TCP's
   Fast Retransmit ([RFC5681]), Early Retransmit ([RFC5827]), FACK
   ([FACK]), SACK loss recovery ([RFC6675]), and RACK ([RACK]).  This
   section provides an overview of how these algorithms are implemented
   in QUIC.

   A packet is declared lost if it meets all the following conditions:

   *  The packet is unacknowledged, in-flight, and was sent prior to an
      acknowledged packet.

   *  Either its packet number is kPacketThreshold smaller than an
      acknowledged packet (Section 6.1.1), or it was sent long enough in
      the past (Section 6.1.2).

   The acknowledgement indicates that a packet sent later was delivered,
   and the packet and time thresholds provide some tolerance for packet
   reordering.

   Spuriously declaring packets as lost leads to unnecessary
   retransmissions and may result in degraded performance due to the
   actions of the congestion controller upon detecting loss.
   Implementations can detect spurious retransmissions and increase the
   reordering threshold in packets or time to reduce future spurious
   retransmissions and loss events.  Implementations with adaptive time
   thresholds MAY choose to start with smaller initial reordering
   thresholds to minimize recovery latency.






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6.1.1.  Packet Threshold

   The RECOMMENDED initial value for the packet reordering threshold
   (kPacketThreshold) is 3, based on best practices for TCP loss
   detection ([RFC5681], [RFC6675]).  In order to remain similar to TCP,
   implementations SHOULD NOT use a packet threshold less than 3; see
   [RFC5681].

   Some networks may exhibit higher degrees of packet reordering,
   causing a sender to detect spurious losses.  Additionally, packet
   reordering could be more common with QUIC than TCP, because network
   elements that could observe and fix the order of reordered TCP
   packets cannot do that for QUIC.  Algorithms that increase the
   reordering threshold after spuriously detecting losses, such as RACK
   [RACK], have proven to be useful in TCP and are expected to at least
   as useful in QUIC.

6.1.2.  Time Threshold

   Once a later packet within the same packet number space has been
   acknowledged, an endpoint SHOULD declare an earlier packet lost if it
   was sent a threshold amount of time in the past.  To avoid declaring
   packets as lost too early, this time threshold MUST be set to at
   least the local timer granularity, as indicated by the kGranularity
   constant.  The time threshold is:

   max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity)

   If packets sent prior to the largest acknowledged packet cannot yet
   be declared lost, then a timer SHOULD be set for the remaining time.

   Using max(smoothed_rtt, latest_rtt) protects from the two following
   cases:

   *  the latest RTT sample is lower than the smoothed RTT, perhaps due
      to reordering where the acknowledgement encountered a shorter
      path;

   *  the latest RTT sample is higher than the smoothed RTT, perhaps due
      to a sustained increase in the actual RTT, but the smoothed RTT
      has not yet caught up.

   The RECOMMENDED time threshold (kTimeThreshold), expressed as a
   round-trip time multiplier, is 9/8.  The RECOMMENDED value of the
   timer granularity (kGranularity) is 1ms.






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   Implementations MAY experiment with absolute thresholds, thresholds
   from previous connections, adaptive thresholds, or including RTT
   variation.  Smaller thresholds reduce reordering resilience and
   increase spurious retransmissions, and larger thresholds increase
   loss detection delay.

6.2.  Probe Timeout

   A Probe Timeout (PTO) triggers sending one or two probe datagrams
   when ack-eliciting packets are not acknowledged within the expected
   period of time or the server may not have validated the client's
   address.  A PTO enables a connection to recover from loss of tail
   packets or acknowledgements.

   A PTO timer expiration event does not indicate packet loss and MUST
   NOT cause prior unacknowledged packets to be marked as lost.  When an
   acknowledgement is received that newly acknowledges packets, loss
   detection proceeds as dictated by packet and time threshold
   mechanisms; see Section 6.1.

   As with loss detection, the probe timeout is per packet number space.
   The PTO algorithm used in QUIC implements the reliability functions
   of Tail Loss Probe ([RACK]), RTO ([RFC5681]), and F-RTO ([RFC5682])
   algorithms for TCP.  The timeout computation is based on TCP's
   retransmission timeout period ([RFC6298]).

6.2.1.  Computing PTO

   When an ack-eliciting packet is transmitted, the sender schedules a
   timer for the PTO period as follows:

   PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay

   The PTO period is the amount of time that a sender ought to wait for
   an acknowledgement of a sent packet.  This time period includes the
   estimated network round-trip time (smoothed_rtt), the variation in
   the estimate (4*rttvar), and max_ack_delay, to account for the
   maximum time by which a receiver might delay sending an
   acknowledgement.  When the PTO is armed for Initial or Handshake
   packet number spaces, the max_ack_delay is 0, as specified in 13.2.1
   of [QUIC-TRANSPORT].

   The PTO value MUST be set to at least kGranularity, to avoid the
   timer expiring immediately.

   A sender recomputes and may need to reset its PTO timer every time an
   ack-eliciting packet is sent or acknowledged, when the handshake is
   confirmed, or when Initial or Handshake keys are discarded.  This



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   ensures the PTO is always set based on the latest RTT information and
   for the last sent ack-eliciting packet in the correct packet number
   space.  When the PTO expires and there are no ack-eliciting packets
   in flight, the PTO is set from that moment.

   When ack-eliciting packets in multiple packet number spaces are in
   flight, the timer MUST be set for the packet number space with the
   earliest timeout, with one exception.  The Application Data packet
   number space (Section 4.1.1 of [QUIC-TLS]) MUST be ignored until the
   handshake completes.  Not arming the PTO for Application Data
   prevents a client from retransmitting a 0-RTT packet on a PTO
   expiration before confirming that the server is able to decrypt 0-RTT
   packets, and prevents a server from sending a 1-RTT packet on a PTO
   expiration before it has the keys to process an acknowledgement.

   When a PTO timer expires, the PTO backoff MUST be increased,
   resulting in the PTO period being set to twice its current value.
   The PTO backoff factor is reset when an acknowledgement is received,
   except in the following case.  A server might take longer to respond
   to packets during the handshake than otherwise.  To protect such a
   server from repeated client probes, the PTO backoff is not reset at a
   client that is not yet certain that the server has finished
   validating the client's address.  That is, a client does not reset
   the PTO backoff factor on receiving acknowledgements until it
   receives a HANDSHAKE_DONE frame or an acknowledgement for one of its
   Handshake or 1-RTT packets.

   This exponential reduction in the sender's rate is important because
   consecutive PTOs might be caused by loss of packets or
   acknowledgements due to severe congestion.  Even when there are ack-
   eliciting packets in-flight in multiple packet number spaces, the
   exponential increase in probe timeout occurs across all spaces to
   prevent excess load on the network.  For example, a timeout in the
   Initial packet number space doubles the length of the timeout in the
   Handshake packet number space.

   The life of a connection that is experiencing consecutive PTOs is
   limited by the endpoint's idle timeout.

   The probe timer MUST NOT be set if the time threshold (Section 6.1.2)
   loss detection timer is set.  The time threshold loss detection timer
   is expected to both expire earlier than the PTO and be less likely to
   spuriously retransmit data.








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6.2.2.  Handshakes and New Paths

   Resumed connections over the same network MAY use the previous
   connection's final smoothed RTT value as the resumed connection's
   initial RTT.  When no previous RTT is available, the initial RTT
   SHOULD be set to 333ms, resulting in a 1 second initial timeout, as
   recommended in [RFC6298].

   A connection MAY use the delay between sending a PATH_CHALLENGE and
   receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in
   Appendix A.2) for a new path, but the delay SHOULD NOT be considered
   an RTT sample.

   Initial packets and Handshake packets could be never acknowledged,
   but they are removed from bytes in flight when the Initial and
   Handshake keys are discarded, as described below in Section 6.4.
   When Initial or Handshake keys are discarded, the PTO and loss
   detection timers MUST be reset, because discarding keys indicates
   forward progress and the loss detection timer might have been set for
   a now discarded packet number space.

6.2.2.1.  Before Address Validation

   Until the server has validated the client's address on the path, the
   amount of data it can send is limited to three times the amount of
   data received, as specified in Section 8.1 of [QUIC-TRANSPORT].  If
   no additional data can be sent, the server's PTO timer MUST NOT be
   armed until datagrams have been received from the client, because
   packets sent on PTO count against the anti-amplification limit.  Note
   that the server could fail to validate the client's address even if
   0-RTT is accepted.

   Since the server could be blocked until more datagrams are received
   from the client, it is the client's responsibility to send packets to
   unblock the server until it is certain that the server has finished
   its address validation (see Section 8 of [QUIC-TRANSPORT]).  That is,
   the client MUST set the probe timer if the client has not received an
   acknowledgement for one of its Handshake or 1-RTT packets and has not
   received a HANDSHAKE_DONE frame, even if there are no packets in
   flight.  When the PTO fires, the client MUST send a Handshake packet
   if it has Handshake keys, otherwise it MUST send an Initial packet in
   a UDP datagram of at least 1200 bytes.









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6.2.3.  Speeding Up Handshake Completion

   When a server receives an Initial packet containing duplicate CRYPTO
   data, it can assume the client did not receive all of the server's
   CRYPTO data sent in Initial packets, or the client's estimated RTT is
   too small.  When a client receives Handshake or 1-RTT packets prior
   to obtaining Handshake keys, it may assume some or all of the
   server's Initial packets were lost.

   To speed up handshake completion under these conditions, an endpoint
   MAY send a packet containing unacknowledged CRYPTO data earlier than
   the PTO expiry, subject to the address validation limits in
   Section 8.1 of [QUIC-TRANSPORT].

   Endpoints can also use coalesced packets to ensure that each datagram
   elicits at least one acknowledgement.  For example, a client can
   coalesce an Initial packet containing PING and PADDING frames with a
   0-RTT data packet and a server can coalesce an Initial packet
   containing a PING frame with one or more packets in its first flight.

6.2.4.  Sending Probe Packets

   When a PTO timer expires, a sender MUST send at least one ack-
   eliciting packet in the packet number space as a probe.  An endpoint
   MAY send up to two full-sized datagrams containing ack-eliciting
   packets, to avoid an expensive consecutive PTO expiration due to a
   single lost datagram or transmit data from multiple packet number
   spaces.  All probe packets sent on a PTO MUST be ack-eliciting.

   In addition to sending data in the packet number space for which the
   timer expired, the sender SHOULD send ack-eliciting packets from
   other packet number spaces with in-flight data, coalescing packets if
   possible.  This is particularly valuable when the server has both
   Initial and Handshake data in-flight or the client has both Handshake
   and Application Data in-flight, because the peer might only have
   receive keys for one of the two packet number spaces.

   If the sender wants to elicit a faster acknowledgement on PTO, it can
   skip a packet number to eliminate the acknowledgment delay.

   When the PTO timer expires, an ack-eliciting packet MUST be sent.  An
   endpoint SHOULD include new data in this packet.  Previously sent
   data MAY be sent if no new data can be sent.  Implementations MAY use
   alternative strategies for determining the content of probe packets,
   including sending new or retransmitted data based on the
   application's priorities.





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   It is possible the sender has no new or previously-sent data to send.
   As an example, consider the following sequence of events: new
   application data is sent in a STREAM frame, deemed lost, then
   retransmitted in a new packet, and then the original transmission is
   acknowledged.  When there is no data to send, the sender SHOULD send
   a PING or other ack-eliciting frame in a single packet, re-arming the
   PTO timer.

   Alternatively, instead of sending an ack-eliciting packet, the sender
   MAY mark any packets still in flight as lost.  Doing so avoids
   sending an additional packet, but increases the risk that loss is
   declared too aggressively, resulting in an unnecessary rate reduction
   by the congestion controller.

   Consecutive PTO periods increase exponentially, and as a result,
   connection recovery latency increases exponentially as packets
   continue to be dropped in the network.  Sending two packets on PTO
   expiration increases resilience to packet drops, thus reducing the
   probability of consecutive PTO events.

   When the PTO timer expires multiple times and new data cannot be
   sent, implementations must choose between sending the same payload
   every time or sending different payloads.  Sending the same payload
   may be simpler and ensures the highest priority frames arrive first.
   Sending different payloads each time reduces the chances of spurious
   retransmission.

6.3.  Handling Retry Packets

   A Retry packet causes a client to send another Initial packet,
   effectively restarting the connection process.  A Retry packet
   indicates that the Initial was received, but not processed.  A Retry
   packet cannot be treated as an acknowledgment, because it does not
   indicate that a packet was processed or specify the packet number.

   Clients that receive a Retry packet reset congestion control and loss
   recovery state, including resetting any pending timers.  Other
   connection state, in particular cryptographic handshake messages, is
   retained; see Section 17.2.5 of [QUIC-TRANSPORT].

   The client MAY compute an RTT estimate to the server as the time
   period from when the first Initial was sent to when a Retry or a
   Version Negotiation packet is received.  The client MAY use this
   value in place of its default for the initial RTT estimate.







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6.4.  Discarding Keys and Packet State

   When packet protection keys are discarded (see Section 4.8 of
   [QUIC-TLS]), all packets that were sent with those keys can no longer
   be acknowledged because their acknowledgements cannot be processed
   anymore.  The sender MUST discard all recovery state associated with
   those packets and MUST remove them from the count of bytes in flight.

   Endpoints stop sending and receiving Initial packets once they start
   exchanging Handshake packets; see Section 17.2.2.1 of
   [QUIC-TRANSPORT].  At this point, recovery state for all in-flight
   Initial packets is discarded.

   When 0-RTT is rejected, recovery state for all in-flight 0-RTT
   packets is discarded.

   If a server accepts 0-RTT, but does not buffer 0-RTT packets that
   arrive before Initial packets, early 0-RTT packets will be declared
   lost, but that is expected to be infrequent.

   It is expected that keys are discarded after packets encrypted with
   them would be acknowledged or declared lost.  However, Initial
   secrets are discarded as soon as handshake keys are proven to be
   available to both client and server; see Section 4.9.1 of [QUIC-TLS].

7.  Congestion Control

   This document specifies a congestion controller for QUIC similar to
   TCP NewReno ([RFC6582]).

   The signals QUIC provides for congestion control are generic and are
   designed to support different algorithms.  Endpoints can unilaterally
   choose a different algorithm to use, such as Cubic ([RFC8312]).

   If an endpoint uses a different controller than that specified in
   this document, the chosen controller MUST conform to the congestion
   control guidelines specified in Section 3.1 of [RFC8085].

   Similar to TCP, packets containing only ACK frames do not count
   towards bytes in flight and are not congestion controlled.  Unlike
   TCP, QUIC can detect the loss of these packets and MAY use that
   information to adjust the congestion controller or the rate of ACK-
   only packets being sent, but this document does not describe a
   mechanism for doing so.

   The algorithm in this document specifies and uses the controller's
   congestion window in bytes.




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   An endpoint MUST NOT send a packet if it would cause bytes_in_flight
   (see Appendix B.2) to be larger than the congestion window, unless
   the packet is sent on a PTO timer expiration; see Section 6.2.

7.1.  Explicit Congestion Notification

   If a path has been verified to support ECN ([RFC3168], [RFC8311]),
   QUIC treats a Congestion Experienced (CE) codepoint in the IP header
   as a signal of congestion.  This document specifies an endpoint's
   response when its peer receives packets with the ECN-CE codepoint.

7.2.  Initial and Minimum Congestion Window

   QUIC begins every connection in slow start with the congestion window
   set to an initial value.  Endpoints SHOULD use an initial congestion
   window of 10 times the maximum datagram size (max_datagram_size),
   limited to the larger of 14720 or twice the maximum datagram size.
   This follows the analysis and recommendations in [RFC6928],
   increasing the byte limit to account for the smaller 8 byte overhead
   of UDP compared to the 20 byte overhead for TCP.

   Prior to validating the client's address, the server can be further
   limited by the anti-amplification limit as specified in Section 8.1
   of [QUIC-TRANSPORT].  Though the anti-amplification limit can prevent
   the congestion window from being fully utilized and therefore slow
   down the increase in congestion window, it does not directly affect
   the congestion window.

   The minimum congestion window is the smallest value the congestion
   window can decrease to as a response to loss, ECN-CE, or persistent
   congestion.  The RECOMMENDED value is 2 * max_datagram_size.

7.3.  Slow Start

   The congestion controller is in slow start any time the congestion
   window is below the slow start threshold.  The congestion controller
   begins in slow start.

   While the congestion controller is in slow start, the congestion
   window increases by the number of bytes acknowledged when each
   acknowledgment is processed.  This results in exponential growth of
   the congestion window.

   The congestion controller exits slow start and enters congestion
   avoidance when a packet is lost or when the ECN-CE count increases.






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   The congestion controller re-enters slow start any time the
   congestion window is less than the slow start threshold, which only
   occurs after persistent congestion is declared.

7.4.  Congestion Avoidance

   The congestion controller is in congestion avoidance any time the
   congestion window is at or above the slow start threshold.

   Congestion avoidance uses an Additive Increase Multiplicative
   Decrease (AIMD) approach that increases the congestion window by one
   maximum packet size per congestion window acknowledged.

   When first entering congestion avoidance, when a packet is declared
   lost, or when the ECN-CE count is detected, the congestion controller
   halves the congestion window, reduces the slow start threshold to the
   resulting size, and enters the recovery period.

7.5.  Recovery Period

   A recovery period is entered when loss or ECN-CE marking of a packet
   is detected in congestion avoidance after the congestion window and
   slow start threshold have been decreased.  A recovery period ends
   when a packet sent during the recovery period is acknowledged.  This
   is slightly different from TCP's definition of recovery, which ends
   when the lost packet that started recovery is acknowledged.

   The recovery period aims to limit congestion window reduction to once
   per round trip.  Therefore during recovery, the congestion window
   remains unchanged irrespective of new losses or increases in the ECN-
   CE counter.

   When entering recovery, a single packet MAY be sent even if bytes in
   flight now exceeds the recently reduced congestion window.  This
   speeds up loss recovery if the data in the lost packet is
   retransmitted and is similar to TCP as described in Section 5 of
   [RFC6675].  If further packets are lost while the sender is in
   recovery, sending any packets in response MUST obey the congestion
   window limit.

7.6.  Ignoring Loss of Undecryptable Packets

   During the handshake, some packet protection keys might not be
   available when a packet arrives and the receiver can choose to drop
   the packet.  In particular, Handshake and 0-RTT packets cannot be
   processed until the Initial packets arrive and 1-RTT packets cannot
   be processed until the handshake completes.  Endpoints MAY ignore the
   loss of Handshake, 0-RTT, and 1-RTT packets that might have arrived



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   before the peer had packet protection keys to process those packets.
   Endpoints MUST NOT ignore the loss of packets that were sent after
   the earliest acknowledged packet in a given packet number space.

7.7.  Probe Timeout

   Probe packets MUST NOT be blocked by the congestion controller.  A
   sender MUST however count these packets as being additionally in
   flight, since these packets add network load without establishing
   packet loss.  Note that sending probe packets might cause the
   sender's bytes in flight to exceed the congestion window until an
   acknowledgement is received that establishes loss or delivery of
   packets.

7.8.  Persistent Congestion

   When an ACK frame is received that establishes loss of all in-flight
   packets sent over a long enough period of time, the network is
   considered to be experiencing persistent congestion.  Commonly, this
   can be established by consecutive PTOs, but since the PTO timer is
   reset when a new ack-eliciting packet is sent, an explicit duration
   must be used to account for those cases where PTOs do not occur or
   are substantially delayed.  The rationale for this threshold is to
   enable a sender to use initial PTOs for aggressive probing, as TCP
   does with Tail Loss Probe (TLP; see [RACK]), before establishing
   persistent congestion, as TCP does with a Retransmission Timeout
   (RTO; see [RFC5681]).  The RECOMMENDED value for
   kPersistentCongestionThreshold is 3, which is approximately
   equivalent to two TLPs before an RTO in TCP.

   The persistent congestion period SHOULD NOT start until there is at
   least one RTT sample.  Prior to an RTT sample, the duration cannot be
   correctly calculated.  Waiting for one RTT sample also avoids
   spuriously declaring persistent congestion when the initial RTT is
   larger than the actual RTT.

   This duration is computed as follows:

   (smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay) *
       kPersistentCongestionThreshold

   Unlike the PTO computation in Section 6.2, persistent congestion
   includes the max_ack_delay irrespective of the packet number spaces
   in which losses are established.

   The following example illustrates how persistent congestion can be
   established.  Assume:




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   smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay = 2
   kPersistentCongestionThreshold = 3

   Consider the following sequence of events:

                  +========+============================+
                  | Time   | Action                     |
                  +========+============================+
                  | t=0    | Send packet #1 (app data)  |
                  +--------+----------------------------+
                  | t=1    | Send packet #2 (app data)  |
                  +--------+----------------------------+
                  | t=1.2  | Recv acknowledgement of #1 |
                  +--------+----------------------------+
                  | t=2    | Send packet #3 (app data)  |
                  +--------+----------------------------+
                  | t=3    | Send packet #4 (app data)  |
                  +--------+----------------------------+
                  | t=4    | Send packet #5 (app data)  |
                  +--------+----------------------------+
                  | t=5    | Send packet #6 (app data)  |
                  +--------+----------------------------+
                  | t=6    | Send packet #7 (app data)  |
                  +--------+----------------------------+
                  | t=8    | Send packet #8 (PTO 1)     |
                  +--------+----------------------------+
                  | t=12   | Send packet #9 (PTO 2)     |
                  +--------+----------------------------+
                  | t=12.2 | Recv acknowledgement of #9 |
                  +--------+----------------------------+

                                  Table 1

   Packets 2 through 8 are declared lost when the acknowledgement for
   packet 9 is received at t = 12.2.

   The congestion period is calculated as the time between the oldest
   and newest lost packets: 8 - 1 = 7.  The duration for establishing
   persistent congestion is: 2 * 3 = 6.  Because the threshold was
   reached and because none of the packets between the oldest and the
   newest lost packets were acknowledged, the network is considered to
   have experienced persistent congestion.

   While this example shows the occurrence of PTOs, they are not
   required for persistent congestion to be established.






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   When persistent congestion is established, the sender's congestion
   window MUST be reduced to the minimum congestion window
   (kMinimumWindow).  This response of collapsing the congestion window
   on persistent congestion is functionally similar to a sender's
   response on a Retransmission Timeout (RTO) in TCP ([RFC5681]) after
   Tail Loss Probes (TLP; see [RACK]).

7.9.  Pacing

   This document does not specify a pacer, but it is RECOMMENDED that a
   sender pace sending of all in-flight packets based on input from the
   congestion controller.  Sending multiple packets into the network
   without any delay between them creates a packet burst that might
   cause short-term congestion and losses.  Implementations MUST either
   use pacing or another method to limit such bursts to the initial
   congestion window; see Section 7.2.

   An implementation should take care to architect its congestion
   controller to work well with a pacer.  For instance, a pacer might
   wrap the congestion controller and control the availability of the
   congestion window, or a pacer might pace out packets handed to it by
   the congestion controller.

   Timely delivery of ACK frames is important for efficient loss
   recovery.  Packets containing only ACK frames SHOULD therefore not be
   paced, to avoid delaying their delivery to the peer.

   Endpoints can implement pacing as they choose.  A perfectly paced
   sender spreads packets exactly evenly over time.  For a window-based
   congestion controller, such as the one in this document, that rate
   can be computed by averaging the congestion window over the round-
   trip time.  Expressed as a rate in bytes:

   rate = N * congestion_window / smoothed_rtt

   Or, expressed as an inter-packet interval:

   interval = smoothed_rtt * packet_size / congestion_window / N

   Using a value for "N" that is small, but at least 1 (for example,
   1.25) ensures that variations in round-trip time do not result in
   under-utilization of the congestion window.  Values of 'N' larger
   than 1 ultimately result in sending packets as acknowledgments are
   received rather than when timers fire, provided the congestion window
   is fully utilized and acknowledgments arrive at regular intervals.






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   Practical considerations, such as packetization, scheduling delays,
   and computational efficiency, can cause a sender to deviate from this
   rate over time periods that are much shorter than a round-trip time.

   One possible implementation strategy for pacing uses a leaky bucket
   algorithm, where the capacity of the "bucket" is limited to the
   maximum burst size and the rate the "bucket" fills is determined by
   the above function.

7.10.  Under-utilizing the Congestion Window

   When bytes in flight is smaller than the congestion window and
   sending is not pacing limited, the congestion window is under-
   utilized.  When this occurs, the congestion window SHOULD NOT be
   increased in either slow start or congestion avoidance.  This can
   happen due to insufficient application data or flow control limits.

   A sender MAY use the pipeACK method described in Section 4.3 of
   [RFC7661] to determine if the congestion window is sufficiently
   utilized.

   A sender that paces packets (see Section 7.9) might delay sending
   packets and not fully utilize the congestion window due to this
   delay.  A sender SHOULD NOT consider itself application limited if it
   would have fully utilized the congestion window without pacing delay.

   A sender MAY implement alternative mechanisms to update its
   congestion window after periods of under-utilization, such as those
   proposed for TCP in [RFC7661].

8.  Security Considerations

8.1.  Congestion Signals

   Congestion control fundamentally involves the consumption of signals
   - both loss and ECN codepoints - from unauthenticated entities.  On-
   path attackers can spoof or alter these signals.  An attacker can
   cause endpoints to reduce their sending rate by dropping packets, or
   alter send rate by changing ECN codepoints.

8.2.  Traffic Analysis

   Packets that carry only ACK frames can be heuristically identified by
   observing packet size.  Acknowledgement patterns may expose
   information about link characteristics or application behavior.  To
   reduce leaked information, endpoints can bundle acknowledgments with
   other frames, or they can use PADDING frames at a potential cost to
   performance.



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8.3.  Misreporting ECN Markings

   A receiver can misreport ECN markings to alter the congestion
   response of a sender.  Suppressing reports of ECN-CE markings could
   cause a sender to increase their send rate.  This increase could
   result in congestion and loss.

   A sender MAY attempt to detect suppression of reports by marking
   occasional packets that they send with ECN-CE.  If a packet sent with
   ECN-CE is not reported as having been CE marked when the packet is
   acknowledged, then the sender SHOULD disable ECN for that path.

   Reporting additional ECN-CE markings will cause a sender to reduce
   their sending rate, which is similar in effect to advertising reduced
   connection flow control limits and so no advantage is gained by doing
   so.

   Endpoints choose the congestion controller that they use.  Though
   congestion controllers generally treat reports of ECN-CE markings as
   equivalent to loss ([RFC8311]), the exact response for each
   controller could be different.  Failure to correctly respond to
   information about ECN markings is therefore difficult to detect.

9.  IANA Considerations

   This document has no IANA actions.

10.  References

10.1.  Normative References

   [QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
              QUIC", Work in Progress, Internet-Draft, draft-ietf-quic-
              tls-latest, 17 August 2020,
              <https://tools.ietf.org/html/draft-ietf-quic-tls-latest>.

   [QUIC-TRANSPORT]
              Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", Work in Progress,
              Internet-Draft, draft-ietf-quic-transport-latest, 17
              August 2020, <https://tools.ietf.org/html/draft-ietf-quic-
              transport-latest>.

   [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>.




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   [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>.

   [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>.

10.2.  Informative References

   [FACK]     Mathis, M. and J. Mahdavi, "Forward Acknowledgement:
              Refining TCP Congestion Control", ACM SIGCOMM , August
              1996.

   [RACK]     Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "RACK-
              TLP: a time-based efficient loss detection for TCP", Work
              in Progress, Internet-Draft, draft-ietf-tcpm-rack-09, 13
              July 2020, <http://www.ietf.org/internet-drafts/draft-
              ietf-tcpm-rack-09.txt>.

   [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>.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
              <https://www.rfc-editor.org/info/rfc5681>.

   [RFC5682]  Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
              "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
              Spurious Retransmission Timeouts with TCP", RFC 5682,
              DOI 10.17487/RFC5682, September 2009,
              <https://www.rfc-editor.org/info/rfc5682>.

   [RFC5827]  Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
              P. Hurtig, "Early Retransmit for TCP and Stream Control
              Transmission Protocol (SCTP)", RFC 5827,
              DOI 10.17487/RFC5827, May 2010,
              <https://www.rfc-editor.org/info/rfc5827>.

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,
              <https://www.rfc-editor.org/info/rfc6298>.






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   [RFC6582]  Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
              NewReno Modification to TCP's Fast Recovery Algorithm",
              RFC 6582, DOI 10.17487/RFC6582, April 2012,
              <https://www.rfc-editor.org/info/rfc6582>.

   [RFC6675]  Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
              and Y. Nishida, "A Conservative Loss Recovery Algorithm
              Based on Selective Acknowledgment (SACK) for TCP",
              RFC 6675, DOI 10.17487/RFC6675, August 2012,
              <https://www.rfc-editor.org/info/rfc6675>.

   [RFC6928]  Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
              "Increasing TCP's Initial Window", RFC 6928,
              DOI 10.17487/RFC6928, April 2013,
              <https://www.rfc-editor.org/info/rfc6928>.

   [RFC7661]  Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
              TCP to Support Rate-Limited Traffic", RFC 7661,
              DOI 10.17487/RFC7661, October 2015,
              <https://www.rfc-editor.org/info/rfc7661>.

   [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>.

   [RFC8312]  Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
              R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
              RFC 8312, DOI 10.17487/RFC8312, February 2018,
              <https://www.rfc-editor.org/info/rfc8312>.

Appendix A.  Loss Recovery Pseudocode

   We now describe an example implementation of the loss detection
   mechanisms described in Section 6.

A.1.  Tracking Sent Packets

   To correctly implement congestion control, a QUIC sender tracks every
   ack-eliciting packet until the packet is acknowledged or lost.  It is
   expected that implementations will be able to access this information
   by packet number and crypto context and store the per-packet fields
   (Appendix A.1.1) for loss recovery and congestion control.

   After a packet is declared lost, the endpoint can track it for an
   amount of time comparable to the maximum expected packet reordering,
   such as 1 RTT.  This allows for detection of spurious
   retransmissions.



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   Sent packets are tracked for each packet number space, and ACK
   processing only applies to a single space.

A.1.1.  Sent Packet Fields

   packet_number:  The packet number of the sent packet.

   ack_eliciting:  A boolean that indicates whether a packet is ack-
      eliciting.  If true, it is expected that an acknowledgement will
      be received, though the peer could delay sending the ACK frame
      containing it by up to the max_ack_delay.

   in_flight:  A boolean that indicates whether the packet counts
      towards bytes in flight.

   sent_bytes:  The number of bytes sent in the packet, not including
      UDP or IP overhead, but including QUIC framing overhead.

   time_sent:  The time the packet was sent.

A.2.  Constants of Interest

   Constants used in loss recovery are based on a combination of RFCs,
   papers, and common practice.

   kPacketThreshold:  Maximum reordering in packets before packet
      threshold loss detection considers a packet lost.  The value
      recommended in Section 6.1.1 is 3.

   kTimeThreshold:  Maximum reordering in time before time threshold
      loss detection considers a packet lost.  Specified as an RTT
      multiplier.  The value recommended in Section 6.1.2 is 9/8.

   kGranularity:  Timer granularity.  This is a system-dependent value,
      and Section 6.1.2 recommends a value of 1ms.

   kInitialRtt:  The RTT used before an RTT sample is taken.  The value
      recommended in Section 6.2.2 is 333ms.

   kPacketNumberSpace:  An enum to enumerate the three packet number
      spaces.

     enum kPacketNumberSpace {
       Initial,
       Handshake,
       ApplicationData,
     }




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A.3.  Variables of interest

   Variables required to implement the congestion control mechanisms are
   described in this section.

   latest_rtt:  The most recent RTT measurement made when receiving an
      ack for a previously unacked packet.

   smoothed_rtt:  The smoothed RTT of the connection, computed as
      described in Section 5.3.

   rttvar:  The RTT variation, computed as described in Section 5.3.

   min_rtt:  The minimum RTT seen in the connection, ignoring
      acknowledgment delay, as described in Section 5.2.

   max_ack_delay:  The maximum amount of time by which the receiver
      intends to delay acknowledgments for packets in the Application
      Data packet number space.  The actual ack_delay in a received ACK
      frame may be larger due to late timers, reordering, or lost ACK
      frames.

   loss_detection_timer:  Multi-modal timer used for loss detection.

   pto_count:  The number of times a PTO has been sent without receiving
      an ack.

   time_of_last_ack_eliciting_packet[kPacketNumberSpace]:  The time the
      most recent ack-eliciting packet was sent.

   largest_acked_packet[kPacketNumberSpace]:  The largest packet number
      acknowledged in the packet number space so far.

   loss_time[kPacketNumberSpace]:  The time at which the next packet in
      that packet number space will be considered lost based on
      exceeding the reordering window in time.

   sent_packets[kPacketNumberSpace]:  An association of packet numbers
      in a packet number space to information about them.  Described in
      detail above in Appendix A.1.

A.4.  Initialization

   At the beginning of the connection, initialize the loss detection
   variables as follows:






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      loss_detection_timer.reset()
      pto_count = 0
      latest_rtt = 0
      smoothed_rtt = kInitialRtt
      rttvar = kInitialRtt / 2
      min_rtt = 0
      max_ack_delay = 0
      for pn_space in [ Initial, Handshake, ApplicationData ]:
        largest_acked_packet[pn_space] = infinite
        time_of_last_ack_eliciting_packet[pn_space] = 0
        loss_time[pn_space] = 0

A.5.  On Sending a Packet

   After a packet is sent, information about the packet is stored.  The
   parameters to OnPacketSent are described in detail above in
   Appendix A.1.1.

   Pseudocode for OnPacketSent follows:

    OnPacketSent(packet_number, pn_space, ack_eliciting,
                 in_flight, sent_bytes):
      sent_packets[pn_space][packet_number].packet_number =
                                               packet_number
      sent_packets[pn_space][packet_number].time_sent = now()
      sent_packets[pn_space][packet_number].ack_eliciting =
                                               ack_eliciting
      sent_packets[pn_space][packet_number].in_flight = in_flight
      if (in_flight):
        if (ack_eliciting):
          time_of_last_ack_eliciting_packet[pn_space] = now()
        OnPacketSentCC(sent_bytes)
        sent_packets[pn_space][packet_number].sent_bytes =
          sent_bytes
        SetLossDetectionTimer()

A.6.  On Receiving a Datagram

   When a server is blocked by anti-amplification limits, receiving a
   datagram unblocks it, even if none of the packets in the datagram are
   successfully processed.  In such a case, the PTO timer will need to
   be re-armed.

   Pseudocode for OnDatagramReceived follows:







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   OnDatagramReceived(datagram):
     // If this datagram unblocks the server, arm the
     // PTO timer to avoid deadlock.
     if (server was at anti-amplification limit):
       SetLossDetectionTimer()

A.7.  On Receiving an Acknowledgment

   When an ACK frame is received, it may newly acknowledge any number of
   packets.

   Pseudocode for OnAckReceived and UpdateRtt follow:

   IncludesAckEliciting(packets):
     for packet in packets:
       if (packet.ack_eliciting):
         return true
     return false

   OnAckReceived(ack, pn_space):
     if (largest_acked_packet[pn_space] == infinite):
       largest_acked_packet[pn_space] = ack.largest_acked
     else:
       largest_acked_packet[pn_space] =
           max(largest_acked_packet[pn_space], ack.largest_acked)

     // DetectAndRemoveAckedPackets finds packets that are newly
     // acknowledged and removes them from sent_packets.
     newly_acked_packets =
         DetectAndRemoveAckedPackets(ack, pn_space)
     // Nothing to do if there are no newly acked packets.
     if (newly_acked_packets.empty()):
       return

     // If the largest acknowledged is newly acked and
     // at least one ack-eliciting was newly acked, update the RTT.
     if (newly_acked_packets.largest().packet_number ==
             ack.largest_acked &&
         IncludesAckEliciting(newly_acked_packets)):
       latest_rtt =
         now() - newly_acked_packets.largest().time_sent
       ack_delay = 0
       if (pn_space == ApplicationData):
         ack_delay = ack.ack_delay
       UpdateRtt(ack_delay)

     // Process ECN information if present.
     if (ACK frame contains ECN information):



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         ProcessECN(ack, pn_space)

     lost_packets = DetectAndRemoveLostPackets(pn_space)
     if (!lost_packets.empty()):
       OnPacketsLost(lost_packets)
     OnPacketsAcked(newly_acked_packets)

     // Reset pto_count unless the client is unsure if
     // the server has validated the client's address.
     if (PeerCompletedAddressValidation()):
       pto_count = 0
     SetLossDetectionTimer()


   UpdateRtt(ack_delay):
     if (is first RTT sample):
       min_rtt = latest_rtt
       smoothed_rtt = latest_rtt
       rttvar = latest_rtt / 2
       return

     // min_rtt ignores acknowledgment delay.
     min_rtt = min(min_rtt, latest_rtt)
     // Limit ack_delay by max_ack_delay
     ack_delay = min(ack_delay, max_ack_delay)
     // Adjust for acknowledgment delay if plausible.
     adjusted_rtt = latest_rtt
     if (latest_rtt > min_rtt + ack_delay):
       adjusted_rtt = latest_rtt - ack_delay

     rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt)
     smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt

A.8.  Setting the Loss Detection Timer

   QUIC loss detection uses a single timer for all timeout loss
   detection.  The duration of the timer is based on the timer's mode,
   which is set in the packet and timer events further below.  The
   function SetLossDetectionTimer defined below shows how the single
   timer is set.

   This algorithm may result in the timer being set in the past,
   particularly if timers wake up late.  Timers set in the past fire
   immediately.

   Pseudocode for SetLossDetectionTimer follows:





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   GetLossTimeAndSpace():
     time = loss_time[Initial]
     space = Initial
     for pn_space in [ Handshake, ApplicationData ]:
       if (time == 0 || loss_time[pn_space] < time):
         time = loss_time[pn_space];
         space = pn_space
     return time, space

   GetPtoTimeAndSpace():
     duration = (smoothed_rtt + max(4 * rttvar, kGranularity))
         * (2 ^ pto_count)
     // Arm PTO from now when there are no inflight packets.
     if (no in-flight packets):
       assert(!PeerCompletedAddressValidation())
       if (has handshake keys):
         return (now() + duration), Handshake
       else:
         return (now() + duration), Initial
     pto_timeout = infinite
     pto_space = Initial
     for space in [ Initial, Handshake, ApplicationData ]:
       if (no in-flight packets in space):
           continue;
       if (space == ApplicationData):
         // Skip Application Data until handshake complete.
         if (handshake is not complete):
           return pto_timeout, pto_space
         // Include max_ack_delay and backoff for Application Data.
         duration += max_ack_delay * (2 ^ pto_count)

       t = time_of_last_ack_eliciting_packet[space] + duration
       if (t < pto_timeout):
         pto_timeout = t
         pto_space = space
     return pto_timeout, pto_space

   PeerCompletedAddressValidation():
     // Assume clients validate the server's address implicitly.
     if (endpoint is server):
       return true
     // Servers complete address validation when a
     // protected packet is received.
     return has received Handshake ACK ||
          has received 1-RTT ACK ||
          has received HANDSHAKE_DONE

   SetLossDetectionTimer():



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     earliest_loss_time, _ = GetLossTimeAndSpace()
     if (earliest_loss_time != 0):
       // Time threshold loss detection.
       loss_detection_timer.update(earliest_loss_time)
       return

     if (server is at anti-amplification limit):
       // The server's timer is not set if nothing can be sent.
       loss_detection_timer.cancel()
       return

     if (no ack-eliciting packets in flight &&
         PeerCompletedAddressValidation()):
       // There is nothing to detect lost, so no timer is set.
       // However, the client needs to arm the timer if the
       // server might be blocked by the anti-amplification limit.
       loss_detection_timer.cancel()
       return

     // Determine which PN space to arm PTO for.
     timeout, _ = GetPtoTimeAndSpace()
     loss_detection_timer.update(timeout)

A.9.  On Timeout

   When the loss detection timer expires, the timer's mode determines
   the action to be performed.

   Pseudocode for OnLossDetectionTimeout follows:






















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   OnLossDetectionTimeout():
     earliest_loss_time, pn_space = GetLossTimeAndSpace()
     if (earliest_loss_time != 0):
       // Time threshold loss Detection
       lost_packets = DetectAndRemoveLostPackets(pn_space)
       assert(!lost_packets.empty())
       OnPacketsLost(lost_packets)
       SetLossDetectionTimer()
       return

     if (bytes_in_flight > 0):
       // PTO. Send new data if available, else retransmit old data.
       // If neither is available, send a single PING frame.
       _, pn_space = GetPtoTimeAndSpace()
       SendOneOrTwoAckElicitingPackets(pn_space)
     else:
       assert(!PeerCompletedAddressValidation())
       // Client sends an anti-deadlock packet: Initial is padded
       // to earn more anti-amplification credit,
       // a Handshake packet proves address ownership.
       if (has Handshake keys):
         SendOneAckElicitingHandshakePacket()
       else:
         SendOneAckElicitingPaddedInitialPacket()

     pto_count++
     SetLossDetectionTimer()

A.10.  Detecting Lost Packets

   DetectAndRemoveLostPackets is called every time an ACK is received or
   the time threshold loss detection timer expires.  This function
   operates on the sent_packets for that packet number space and returns
   a list of packets newly detected as lost.

   Pseudocode for DetectAndRemoveLostPackets follows:















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   DetectAndRemoveLostPackets(pn_space):
     assert(largest_acked_packet[pn_space] != infinite)
     loss_time[pn_space] = 0
     lost_packets = {}
     loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)

     // Minimum time of kGranularity before packets are deemed lost.
     loss_delay = max(loss_delay, kGranularity)

     // Packets sent before this time are deemed lost.
     lost_send_time = now() - loss_delay

     foreach unacked in sent_packets[pn_space]:
       if (unacked.packet_number > largest_acked_packet[pn_space]):
         continue

       // Mark packet as lost, or set time when it should be marked.
       if (unacked.time_sent <= lost_send_time ||
           largest_acked_packet[pn_space] >=
             unacked.packet_number + kPacketThreshold):
         sent_packets[pn_space].remove(unacked.packet_number)
         if (unacked.in_flight):
           lost_packets.insert(unacked)
       else:
         if (loss_time[pn_space] == 0):
           loss_time[pn_space] = unacked.time_sent + loss_delay
         else:
           loss_time[pn_space] = min(loss_time[pn_space],
                                     unacked.time_sent + loss_delay)
     return lost_packets

Appendix B.  Congestion Control Pseudocode

   We now describe an example implementation of the congestion
   controller described in Section 7.

B.1.  Constants of interest

   Constants used in congestion control are based on a combination of
   RFCs, papers, and common practice.

   kInitialWindow:  Default limit on the initial bytes in flight as
      described in Section 7.2.

   kMinimumWindow:  Minimum congestion window in bytes as described in
      Section 7.2.

   kLossReductionFactor:  Reduction in congestion window when a new loss



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      event is detected.  Section 7 recommends a value is 0.5.

   kPersistentCongestionThreshold:  Period of time for persistent
      congestion to be established, specified as a PTO multiplier.
      Section 7.8 recommends a value of 3.

B.2.  Variables of interest

   Variables required to implement the congestion control mechanisms are
   described in this section.

   max_datagram_size:  The sender's current maximum payload size.  Does
      not include UDP or IP overhead.  The max datagram size is used for
      congestion window computations.  An endpoint sets the value of
      this variable based on its PMTU (see Section 14.1 of
      [QUIC-TRANSPORT]), with a minimum value of 1200 bytes.

   ecn_ce_counters[kPacketNumberSpace]:  The highest value reported for
      the ECN-CE counter in the packet number space by the peer in an
      ACK frame.  This value is used to detect increases in the reported
      ECN-CE counter.

   bytes_in_flight:  The sum of the size in bytes of all sent packets
      that contain at least one ack-eliciting or PADDING frame, and have
      not been acknowledged or declared lost.  The size does not include
      IP or UDP overhead, but does include the QUIC header and AEAD
      overhead.  Packets only containing ACK frames do not count towards
      bytes_in_flight to ensure congestion control does not impede
      congestion feedback.

   congestion_window:  Maximum number of bytes-in-flight that may be
      sent.

   congestion_recovery_start_time:  The time when QUIC first detects
      congestion due to loss or ECN, causing it to enter congestion
      recovery.  When a packet sent after this time is acknowledged,
      QUIC exits congestion recovery.

   ssthresh:  Slow start threshold in bytes.  When the congestion window
      is below ssthresh, the mode is slow start and the window grows by
      the number of bytes acknowledged.

B.3.  Initialization

   At the beginning of the connection, initialize the congestion control
   variables as follows:





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      congestion_window = kInitialWindow
      bytes_in_flight = 0
      congestion_recovery_start_time = 0
      ssthresh = infinite
      for pn_space in [ Initial, Handshake, ApplicationData ]:
        ecn_ce_counters[pn_space] = 0

B.4.  On Packet Sent

   Whenever a packet is sent, and it contains non-ACK frames, the packet
   increases bytes_in_flight.

      OnPacketSentCC(bytes_sent):
        bytes_in_flight += bytes_sent

B.5.  On Packet Acknowledgement

   Invoked from loss detection's OnAckReceived and is supplied with the
   newly acked_packets from sent_packets.

     InCongestionRecovery(sent_time):
       return sent_time <= congestion_recovery_start_time

     OnPacketsAcked(acked_packets):
       for acked_packet in acked_packets:
         OnPacketAcked(acked_packet)

     OnPacketAcked(acked_packet):
       // Remove from bytes_in_flight.
       bytes_in_flight -= acked_packet.sent_bytes

       // Do not increase congestion_window if application
       // limited or flow control limited.
       if (IsAppOrFlowControlLimited())
         return

       // Do not increase congestion window in recovery period.
       if (InCongestionRecovery(acked_packet.time_sent)):
         return

       if (congestion_window < ssthresh):
         // Slow start.
         congestion_window += acked_packet.sent_bytes
       else:
         // Congestion avoidance.
         congestion_window +=
           max_datagram_size * acked_packet.sent_bytes
           / congestion_window



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B.6.  On New Congestion Event

   Invoked from ProcessECN and OnPacketsLost when a new congestion event
   is detected.  May start a new recovery period and reduces the
   congestion window.

      OnCongestionEvent(sent_time):
        // Start a new congestion event if packet was sent after the
        // start of the previous congestion recovery period.
        if (!InCongestionRecovery(sent_time)):
          congestion_recovery_start_time = now()
          congestion_window *= kLossReductionFactor
          congestion_window = max(congestion_window, kMinimumWindow)
          ssthresh = congestion_window
          // A packet can be sent to speed up loss recovery.
          MaybeSendOnePacket()

B.7.  Process ECN Information

   Invoked when an ACK frame with an ECN section is received from the
   peer.

      ProcessECN(ack, pn_space):
        // If the ECN-CE counter reported by the peer has increased,
        // this could be a new congestion event.
        if (ack.ce_counter > ecn_ce_counters[pn_space]):
          ecn_ce_counters[pn_space] = ack.ce_counter
          sent_time = sent_packets[ack.largest_acked].time_sent
          OnCongestionEvent(sent_time)

B.8.  On Packets Lost

   Invoked when DetectAndRemoveLostPackets deems packets lost.


















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      InPersistentCongestion(largest_lost):
        // Persistent congestion cannot be declared on the
        // first RTT sample.
        if (is first RTT sample):
          return false
        pto = smoothed_rtt + max(4 * rttvar, kGranularity) +
          max_ack_delay
        congestion_period = pto * kPersistentCongestionThreshold
        // Determine if all packets in the time period before the
        // largest newly lost packet, including the edges and
        // across all packet number spaces, are marked lost.
        return AreAllPacketsLost(largest_lost, congestion_period)

      OnPacketsLost(lost_packets):
        // Remove lost packets from bytes_in_flight.
        for lost_packet in lost_packets:
          bytes_in_flight -= lost_packet.sent_bytes
        OnCongestionEvent(lost_packets.largest().time_sent)

        // Collapse congestion window if persistent congestion
        if (InPersistentCongestion(lost_packets.largest())):
          congestion_window = kMinimumWindow

B.9.  Upon dropping Initial or Handshake keys

   When Initial or Handshake keys are discarded, packets from the space
   are discarded and loss detection state is updated.

   Pseudocode for OnPacketNumberSpaceDiscarded follows:

   OnPacketNumberSpaceDiscarded(pn_space):
     assert(pn_space != ApplicationData)
     // Remove any unacknowledged packets from flight.
     foreach packet in sent_packets[pn_space]:
       if packet.in_flight
         bytes_in_flight -= size
     sent_packets[pn_space].clear()
     // Reset the loss detection and PTO timer
     time_of_last_ack_eliciting_packet[pn_space] = 0
     loss_time[pn_space] = 0
     pto_count = 0
     SetLossDetectionTimer()

Appendix C.  Change Log

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




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

C.1.  Since draft-ietf-quic-recovery-28

   *  Refactored pseudocode to correct PTO calculation (#3564, #3674,
      #3681)

C.2.  Since draft-ietf-quic-recovery-27

   *  Added recommendations for speeding up handshake under some loss
      conditions (#3078, #3080)

   *  PTO count is reset when handshake progress is made (#3272, #3415)

   *  PTO count is not reset by a client when the server might be
      awaiting address validation (#3546, #3551)

   *  Recommend repairing losses immediately after entering the recovery
      period (#3335, #3443)

   *  Clarified what loss conditions can be ignored during the handshake
      (#3456, #3450)

   *  Allow, but don't recommend, using RTT from previous connection to
      seed RTT (#3464, #3496)

   *  Recommend use of adaptive loss detection thresholds (#3571, #3572)

C.3.  Since draft-ietf-quic-recovery-26

   No changes.

C.4.  Since draft-ietf-quic-recovery-25

   No significant changes.

C.5.  Since draft-ietf-quic-recovery-24

   *  Require congestion control of some sort (#3247, #3244, #3248)

   *  Set a minimum reordering threshold (#3256, #3240)

   *  PTO is specific to a packet number space (#3067, #3074, #3066)

C.6.  Since draft-ietf-quic-recovery-23

   *  Define under-utilizing the congestion window (#2630, #2686, #2675)




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   *  PTO MUST send data if possible (#3056, #3057)

   *  Connection Close is not ack-eliciting (#3097, #3098)

   *  MUST limit bursts to the initial congestion window (#3160)

   *  Define the current max_datagram_size for congestion control
      (#3041, #3167)

C.7.  Since draft-ietf-quic-recovery-22

   *  PTO should always send an ack-eliciting packet (#2895)

   *  Unify the Handshake Timer with the PTO timer (#2648, #2658, #2886)

   *  Move ACK generation text to transport draft (#1860, #2916)

C.8.  Since draft-ietf-quic-recovery-21

   *  No changes

C.9.  Since draft-ietf-quic-recovery-20

   *  Path validation can be used as initial RTT value (#2644, #2687)

   *  max_ack_delay transport parameter defaults to 0 (#2638, #2646)

   *  ACK delay only measures intentional delays induced by the
      implementation (#2596, #2786)

C.10.  Since draft-ietf-quic-recovery-19

   *  Change kPersistentThreshold from an exponent to a multiplier
      (#2557)

   *  Send a PING if the PTO timer fires and there's nothing to send
      (#2624)

   *  Set loss delay to at least kGranularity (#2617)

   *  Merge application limited and sending after idle sections.  Always
      limit burst size instead of requiring resetting CWND to initial
      CWND after idle (#2605)

   *  Rewrite RTT estimation, allow RTT samples where a newly acked
      packet is ack-eliciting but the largest_acked is not (#2592)





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   *  Don't arm the handshake timer if there is no handshake data
      (#2590)

   *  Clarify that the time threshold loss alarm takes precedence over
      the crypto handshake timer (#2590, #2620)

   *  Change initial RTT to 500ms to align with RFC6298 (#2184)

C.11.  Since draft-ietf-quic-recovery-18

   *  Change IW byte limit to 14720 from 14600 (#2494)

   *  Update PTO calculation to match RFC6298 (#2480, #2489, #2490)

   *  Improve loss detection's description of multiple packet number
      spaces and pseudocode (#2485, #2451, #2417)

   *  Declare persistent congestion even if non-probe packets are sent
      and don't make persistent congestion more aggressive than RTO
      verified was (#2365, #2244)

   *  Move pseudocode to the appendices (#2408)

   *  What to send on multiple PTOs (#2380)

C.12.  Since draft-ietf-quic-recovery-17

   *  After Probe Timeout discard in-flight packets or send another
      (#2212, #1965)

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

   *  0-RTT state is discarded when 0-RTT is rejected (#2300)

   *  Loss detection timer is cancelled when ack-eliciting frames are in
      flight (#2117, #2093)

   *  Packets are declared lost if they are in flight (#2104)

   *  After becoming idle, either pace packets or reset the congestion
      controller (#2138, 2187)

   *  Process ECN counts before marking packets lost (#2142)

   *  Mark packets lost before resetting crypto_count and pto_count
      (#2208, #2209)




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   *  Congestion and loss recovery state are discarded when keys are
      discarded (#2327)

C.13.  Since draft-ietf-quic-recovery-16

   *  Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP
      and min crypto timeouts; eliminate timeout validation (#2114,
      #2166, #2168, #1017)

   *  Redefine how congestion avoidance in terms of when the period
      starts (#1928, #1930)

   *  Document what needs to be tracked for packets that are in flight
      (#765, #1724, #1939)

   *  Integrate both time and packet thresholds into loss detection
      (#1969, #1212, #934, #1974)

   *  Reduce congestion window after idle, unless pacing is used (#2007,
      #2023)

   *  Disable RTT calculation for packets that don't elicit
      acknowledgment (#2060, #2078)

   *  Limit ack_delay by max_ack_delay (#2060, #2099)

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

   *  Reorder ECN and loss detection in pseudocode (#2142)

   *  Only cancel loss detection timer if ack-eliciting packets are in
      flight (#2093, #2117)

C.14.  Since draft-ietf-quic-recovery-14

   *  Used max_ack_delay from transport params (#1796, #1782)

   *  Merge ACK and ACK_ECN (#1783)

C.15.  Since draft-ietf-quic-recovery-13

   *  Corrected the lack of ssthresh reduction in CongestionEvent
      pseudocode (#1598)

   *  Considerations for ECN spoofing (#1426, #1626)





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   *  Clarifications for PADDING and congestion control (#837, #838,
      #1517, #1531, #1540)

   *  Reduce early retransmission timer to RTT/8 (#945, #1581)

   *  Packets are declared lost after an RTO is verified (#935, #1582)

C.16.  Since draft-ietf-quic-recovery-12

   *  Changes to manage separate packet number spaces and encryption
      levels (#1190, #1242, #1413, #1450)

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

C.17.  Since draft-ietf-quic-recovery-11

   No significant changes.

C.18.  Since draft-ietf-quic-recovery-10

   *  Improved text on ack generation (#1139, #1159)

   *  Make references to TCP recovery mechanisms informational (#1195)

   *  Define time_of_last_sent_handshake_packet (#1171)

   *  Added signal from TLS the data it includes needs to be sent in a
      Retry packet (#1061, #1199)

   *  Minimum RTT (min_rtt) is initialized with an infinite value
      (#1169)

C.19.  Since draft-ietf-quic-recovery-09

   No significant changes.

C.20.  Since draft-ietf-quic-recovery-08

   *  Clarified pacing and RTO (#967, #977)

C.21.  Since draft-ietf-quic-recovery-07

   *  Include ACK delay in RTO(and TLP) computations (#981)

   *  ACK delay in SRTT computation (#961)

   *  Default RTT and Slow Start (#590)



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   *  Many editorial fixes.

C.22.  Since draft-ietf-quic-recovery-06

   No significant changes.

C.23.  Since draft-ietf-quic-recovery-05

   *  Add more congestion control text (#776)

C.24.  Since draft-ietf-quic-recovery-04

   No significant changes.

C.25.  Since draft-ietf-quic-recovery-03

   No significant changes.

C.26.  Since draft-ietf-quic-recovery-02

   *  Integrate F-RTO (#544, #409)

   *  Add congestion control (#545, #395)

   *  Require connection abort if a skipped packet was acknowledged
      (#415)

   *  Simplify RTO calculations (#142, #417)

C.27.  Since draft-ietf-quic-recovery-01

   *  Overview added to loss detection

   *  Changes initial default RTT to 100ms

   *  Added time-based loss detection and fixes early retransmit

   *  Clarified loss recovery for handshake packets

   *  Fixed references and made TCP references informative

C.28.  Since draft-ietf-quic-recovery-00

   *  Improved description of constants and ACK behavior







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C.29.  Since draft-iyengar-quic-loss-recovery-01

   *  Adopted as base for draft-ietf-quic-recovery

   *  Updated authors/editors list

   *  Added table of contents

Appendix D.  Contributors

   The IETF QUIC Working Group received an enormous amount of support
   from many people.  The following people provided substantive
   contributions to this document: Alessandro Ghedini, Benjamin
   Saunders, Gorry Fairhurst, 奥 一穂 (Kazuho Oku), Lars Eggert, Magnus
   Westerlund, Marten Seemann, Martin Duke, Martin Thomson, Nick Banks,
   Praveen Balasubramanian.

Acknowledgments

Authors' Addresses

   Jana Iyengar (editor)
   Fastly

   Email: jri.ietf@gmail.com


   Ian Swett (editor)
   Google

   Email: ianswett@google.com




















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