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QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                           I. Swett, Ed.
Expires: 10 September 2020                                        Google
                                                            9 March 2020


               QUIC Loss Detection and Congestion Control
                      draft-ietf-quic-recovery-27

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), which is archived at
   https://mailarchive.ietf.org/arch/search/?email_list=quic
   (https://mailarchive.ietf.org/arch/search/?email_list=quic).

   Working Group information can be found at https://github.com/quicwg
   (https://github.com/quicwg); source code and issues list for this
   draft can be found at https://github.com/quicwg/base-drafts/labels/-
   recovery (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 10 September 2020.

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
   and restrictions with respect to this document.  Code Components
   extracted from this document must include Simplified BSD License text
   as described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Conventions and Definitions . . . . . . . . . . . . . . . . .   4
   3.  Design of the QUIC Transmission Machinery . . . . . . . . . .   5
     3.1.  Relevant Differences Between QUIC and TCP . . . . . . . .   5
       3.1.1.  Separate Packet Number Spaces . . . . . . . . . . . .   6
       3.1.2.  Monotonically Increasing Packet Numbers . . . . . . .   6
       3.1.3.  Clearer Loss Epoch  . . . . . . . . . . . . . . . . .   6
       3.1.4.  No Reneging . . . . . . . . . . . . . . . . . . . . .   7
       3.1.5.  More ACK Ranges . . . . . . . . . . . . . . . . . . .   7
       3.1.6.  Explicit Correction For Delayed Acknowledgements  . .   7
   4.  Estimating the Round-Trip Time  . . . . . . . . . . . . . . .   7
     4.1.  Generating RTT samples  . . . . . . . . . . . . . . . . .   7
     4.2.  Estimating min_rtt  . . . . . . . . . . . . . . . . . . .   8
     4.3.  Estimating smoothed_rtt and rttvar  . . . . . . . . . . .   9
   5.  Loss Detection  . . . . . . . . . . . . . . . . . . . . . . .  10
     5.1.  Acknowledgement-based Detection . . . . . . . . . . . . .  10
       5.1.1.  Packet Threshold  . . . . . . . . . . . . . . . . . .  11
       5.1.2.  Time Threshold  . . . . . . . . . . . . . . . . . . .  11
     5.2.  Probe Timeout . . . . . . . . . . . . . . . . . . . . . .  12
       5.2.1.  Computing PTO . . . . . . . . . . . . . . . . . . . .  12
     5.3.  Handshakes and New Paths  . . . . . . . . . . . . . . . .  13
       5.3.1.  Sending Probe Packets . . . . . . . . . . . . . . . .  14
       5.3.2.  Loss Detection  . . . . . . . . . . . . . . . . . . .  15
     5.4.  Handling Retry Packets  . . . . . . . . . . . . . . . . .  15
     5.5.  Discarding Keys and Packet State  . . . . . . . . . . . .  15
   6.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .  16
     6.1.  Explicit Congestion Notification  . . . . . . . . . . . .  16
     6.2.  Slow Start  . . . . . . . . . . . . . . . . . . . . . . .  17
     6.3.  Congestion Avoidance  . . . . . . . . . . . . . . . . . .  17
     6.4.  Recovery Period . . . . . . . . . . . . . . . . . . . . .  17
     6.5.  Ignoring Loss of Undecryptable Packets  . . . . . . . . .  17
     6.6.  Probe Timeout . . . . . . . . . . . . . . . . . . . . . .  18
     6.7.  Persistent Congestion . . . . . . . . . . . . . . . . . .  18
     6.8.  Pacing  . . . . . . . . . . . . . . . . . . . . . . . . .  19
     6.9.  Under-utilizing the Congestion Window . . . . . . . . . .  20
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  20
     7.1.  Congestion Signals  . . . . . . . . . . . . . . . . . . .  20



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     7.2.  Traffic Analysis  . . . . . . . . . . . . . . . . . . . .  20
     7.3.  Misreporting ECN Markings . . . . . . . . . . . . . . . .  20
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  21
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  21
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  21
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  21
   Appendix A.  Loss Recovery Pseudocode . . . . . . . . . . . . . .  23
     A.1.  Tracking Sent Packets . . . . . . . . . . . . . . . . . .  23
       A.1.1.  Sent Packet Fields  . . . . . . . . . . . . . . . . .  24
     A.2.  Constants of interest . . . . . . . . . . . . . . . . . .  24
     A.3.  Variables of interest . . . . . . . . . . . . . . . . . .  25
     A.4.  Initialization  . . . . . . . . . . . . . . . . . . . . .  25
     A.5.  On Sending a Packet . . . . . . . . . . . . . . . . . . .  26
     A.6.  On Receiving an Acknowledgment  . . . . . . . . . . . . .  26
     A.7.  On Packet Acknowledgment  . . . . . . . . . . . . . . . .  28
     A.8.  Setting the Loss Detection Timer  . . . . . . . . . . . .  28
     A.9.  On Timeout  . . . . . . . . . . . . . . . . . . . . . . .  30
     A.10. Detecting Lost Packets  . . . . . . . . . . . . . . . . .  30
   Appendix B.  Congestion Control Pseudocode  . . . . . . . . . . .  31
     B.1.  Constants of interest . . . . . . . . . . . . . . . . . .  31
     B.2.  Variables of interest . . . . . . . . . . . . . . . . . .  32
     B.3.  Initialization  . . . . . . . . . . . . . . . . . . . . .  33
     B.4.  On Packet Sent  . . . . . . . . . . . . . . . . . . . . .  33
     B.5.  On Packet Acknowledgement . . . . . . . . . . . . . . . .  33
     B.6.  On New Congestion Event . . . . . . . . . . . . . . . . .  34
     B.7.  Process ECN Information . . . . . . . . . . . . . . . . .  34
     B.8.  On Packets Lost . . . . . . . . . . . . . . . . . . . . .  35
   Appendix C.  Change Log . . . . . . . . . . . . . . . . . . . . .  35
     C.1.  Since draft-ietf-quic-recovery-26 . . . . . . . . . . . .  35
     C.2.  Since draft-ietf-quic-recovery-25 . . . . . . . . . . . .  35
     C.3.  Since draft-ietf-quic-recovery-24 . . . . . . . . . . . .  35
     C.4.  Since draft-ietf-quic-recovery-23 . . . . . . . . . . . .  36
     C.5.  Since draft-ietf-quic-recovery-22 . . . . . . . . . . . .  36
     C.6.  Since draft-ietf-quic-recovery-21 . . . . . . . . . . . .  36
     C.7.  Since draft-ietf-quic-recovery-20 . . . . . . . . . . . .  36
     C.8.  Since draft-ietf-quic-recovery-19 . . . . . . . . . . . .  36
     C.9.  Since draft-ietf-quic-recovery-18 . . . . . . . . . . . .  37
     C.10. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . .  37
     C.11. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . .  38
     C.12. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . .  38
     C.13. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . .  38
     C.14. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . .  39
     C.15. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . .  39
     C.16. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . .  39
     C.17. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . .  39
     C.18. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . .  39
     C.19. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . .  39
     C.20. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . .  40



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     C.21. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . .  40
     C.22. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . .  40
     C.23. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . .  40
     C.24. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . .  40
     C.25. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . .  40
     C.26. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . .  40
     C.27. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . .  41
   Appendix D.  Contributors . . . . . . . . . . . . . . . . . . . .  41
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . .  41
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  41

1.  Introduction

   QUIC is a new multiplexed and secure transport atop UDP.  QUIC builds
   on decades of transport and security experience, and implements
   mechanisms that make it attractive as a modern general-purpose
   transport.  The QUIC protocol is described in [QUIC-TRANSPORT].

   QUIC implements the spirit of existing TCP congestion control and
   loss recovery mechanisms, described in RFCs, various Internet-drafts,
   and also those prevalent in the Linux TCP implementation.  This
   document describes QUIC congestion control and loss recovery, and
   where applicable, attributes the TCP equivalent in RFCs, Internet-
   drafts, academic papers, and/or TCP implementations.

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 ack 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 abandoned 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 and eliminates significant
   complexity from QUIC's interpretation of TCP loss detection
   mechanisms.

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

   *  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.

3.1.  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.
   Protocol differences between QUIC and TCP however contribute to
   algorithmic differences.  We briefly describe these protocol
   differences below.








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3.1.1.  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.

3.1.2.  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 rebundles
   necessary frames in a new packet with a new packet number, removing
   ambiguity about which packet is acknowledged when an ACK is received.
   Consequently, more accurate RTT measurements can be made, spurious
   retransmissions are trivially detected, and mechanisms such as Fast
   Retransmit can be applied universally, based only on packet number.

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

3.1.3.  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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3.1.4.  No Reneging

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

3.1.5.  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.

3.1.6.  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.  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).

4.1.  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




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   An RTT sample is generated using only the largest acknowledged packet
   in the received ACK frame.  This is because a peer reports ACK delays
   for only the largest acknowledged packet in an ACK frame.  While the
   reported ACK delay is not used by the RTT sample measurement, it is
   used to adjust the RTT sample in subsequent computations of
   smoothed_rtt and rttvar Section 4.3.

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

   An RTT sample MUST NOT be generated on receiving an ACK frame that
   does not newly acknowledge at least one ack-eliciting packet.  A peer
   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.

4.2.  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 ACK delays reported by the peer.  Doing so
   allows the endpoint to set a lower bound for the smoothed_rtt based
   entirely on what it observes (see Section 4.3), and limits potential
   underestimation due to erroneously-reported delays by the peer.

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







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4.3.  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
   [QUIC-TRANSPORT].  For packets sent in the ApplicationData packet
   number space, a peer limits any delay in sending an acknowledgement
   for an ack-eliciting packet to no greater than the value it
   advertised in the max_ack_delay transport parameter.  Consequently,
   when a peer reports an Ack Delay that is greater than its
   max_ack_delay, the delay is attributed to reasons out of the peer's
   control, such as scheduler latency at the peer or loss of previous
   ACK frames.  Any delays beyond the peer's max_ack_delay are therefore
   considered effectively part of path delay and incorporated into the
   smoothed_rtt estimate.

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

   *  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.

   On the first RTT sample for a network path, the smoothed_rtt is set
   to the latest_rtt.

   smoothed_rtt and rttvar are computed as follows, similar to
   [RFC6298].  On the first RTT sample for a network path:

   smoothed_rtt = latest_rtt
   rttvar = latest_rtt / 2

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








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   ack_delay = min(Ack Delay in ACK Frame, max_ack_delay)
   adjusted_rtt = latest_rtt
   if (min_rtt + ack_delay < latest_rtt):
     adjusted_rtt = latest_rtt - ack_delay
   smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
   rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
   rttvar = 3/4 * rttvar + 1/4 * rttvar_sample

5.  Loss Detection

   QUIC senders use acknowledgements to detect lost packets, and a probe
   time out (see Section 5.2) to ensure acknowledgements are received.
   This section provides a description of these algorithms.

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

5.1.  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 5.1.1), or it was sent long enough in
      the past (Section 5.1.2).

   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 that detect spurious retransmissions and increase the
   reordering threshold in packets or time MAY choose to start with
   smaller initial reordering thresholds to minimize recovery latency.






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5.1.1.  Packet Threshold

   The RECOMMENDED initial value for the packet reordering threshold
   (kPacketThreshold) is 3, based on best practices for TCP loss
   detection [RFC5681] [RFC6675].  Implementations SHOULD NOT use a
   packet threshold less than 3, to keep in line with TCP [RFC5681].

   Some networks may exhibit higher degrees of reordering, causing a
   sender to detect spurious losses.  Implementers MAY use algorithms
   developed for TCP, such as TCP-NCR [RFC4653], to improve QUIC's
   reordering resilience.

5.1.2.  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 kGranularity.  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.

   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.







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5.2.  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 handshake has not been completed.  A PTO
   enables a connection to recover from loss of tail packets or
   acknowledgements.

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

5.2.1.  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

   kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in
   Appendix A.2 and Appendix A.3.

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

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

   A sender computes its PTO timer every time an ack-eliciting packet is
   sent.  When ack-eliciting packets are in-flight in multiple packet
   number spaces, the timer MUST be set for the packet number space with
   the earliest timeout, except for ApplicationData, which MUST be
   ignored until the handshake completes; see Section 4.1.1 of
   [QUIC-TLS].  Not arming the PTO for ApplicationData prioritizes
   completing the handshake and prevents the server from sending a 1-RTT
   packet on a PTO before before it has the keys to process a 1-RTT
   packet.

   When a PTO timer expires, the PTO period MUST be set to twice its
   current value.  This exponential reduction in the sender's rate is
   important because consecutive PTOs might be caused by loss of packets



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   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 5.1.2
   loss detection timer is set.  The time threshold loss detection timer
   is expected to both expire earlier than the PTO and be less likely to
   spuriously retransmit data.

5.3.  Handshakes and New Paths

   The initial probe timeout for a new connection or new path SHOULD be
   set to twice the initial RTT.  Resumed connections over the same
   network SHOULD use the previous connection's final smoothed RTT value
   as the resumed connection's initial RTT.  If no previous RTT is
   available, the initial RTT SHOULD be set to 500ms, resulting in a 1
   second initial timeout as recommended in [RFC6298].

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

   Until the server has validated the client's address on the path, the
   amount of data it can send is limited to three times the amount of
   data received, as specified in Section 8.1 of [QUIC-TRANSPORT].  If
   no data can be sent, then the PTO alarm MUST NOT be armed until
   datagrams have been received from the client.

   Since the server could be blocked until more packets are received
   from the client, it is the client's responsibility to send packets to
   unblock the server until it is certain that the server has finished
   its address validation (see Section 8 of [QUIC-TRANSPORT]).  That is,
   the client MUST set the probe timer if the client has not received an
   acknowledgement for one of its Handshake or 1-RTT packets.

   Prior to handshake completion, when few to none RTT samples have been
   generated, it is possible that the probe timer expiration is due to
   an incorrect RTT estimate at the client.  To allow the client to
   improve its RTT estimate, the new packet that it sends MUST be ack-
   eliciting.  If Handshake keys are available to the client, it MUST




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   send a Handshake packet, and otherwise it MUST send an Initial packet
   in a UDP datagram of at least 1200 bytes.

   Initial packets and Handshake packets could be never acknowledged,
   but they are removed from bytes in flight when the Initial and
   Handshake keys are discarded.

5.3.1.  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, unless there
   is no data available to send.  An endpoint MAY send up to two full-
   sized datagrams containing ack-eliciting packets, to avoid an
   expensive consecutive PTO expiration due to a single lost datagram or
   transmit data from multiple packet number spaces.

   In addition to sending data in the packet number space for which the
   timer expired, the sender SHOULD send ack-eliciting packets from
   other packet number spaces with in-flight data, coalescing packets if
   possible.

   When the PTO timer expires, and there is new or previously sent
   unacknowledged data, it MUST be sent.

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

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

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

   Probe packets sent on a PTO MUST be ack-eliciting.  A probe packet
   SHOULD carry new data when possible.  A probe packet MAY carry
   retransmitted unacknowledged data when new data is unavailable, when
   flow control does not permit new data to be sent, or to



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   opportunistically reduce loss recovery delay.  Implementations MAY
   use alternative strategies for determining the content of probe
   packets, including sending new or retransmitted data based on the
   application's priorities.

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

5.3.2.  Loss Detection

   Delivery or loss of packets in flight is established when an ACK
   frame is received that newly acknowledges one or more packets.

   A PTO timer expiration event does not indicate packet loss and MUST
   NOT cause prior unacknowledged packets to be marked as lost.  When an
   acknowledgement is received that newly acknowledges packets, loss
   detection proceeds as dictated by packet and time threshold
   mechanisms; see Section 5.1.

5.4.  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.

5.5.  Discarding Keys and Packet State

   When packet protection keys are discarded (see Section 4.10 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.



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

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

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

   It is expected that keys are discarded after packets encrypted with
   them would be acknowledged or declared lost.  Initial secrets however
   might be destroyed sooner, as soon as handshake keys are available
   (see Section 4.10.1 of [QUIC-TLS]).

6.  Congestion Control

   This document specifies a Reno congestion controller for QUIC
   [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].

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

   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 5.2).

6.1.  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 Congestion Experienced
   codepoint.







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6.2.  Slow Start

   QUIC begins every connection in slow start and exits slow start upon
   loss or upon increase in the ECN-CE counter.  QUIC re-enters slow
   start any time the congestion window is less than ssthresh, which
   only occurs after persistent congestion is declared.  While in slow
   start, QUIC increases the congestion window by the number of bytes
   acknowledged when each acknowledgment is processed.

6.3.  Congestion Avoidance

   Slow start exits to congestion avoidance.  Congestion avoidance in
   NewReno uses an additive increase multiplicative decrease (AIMD)
   approach that increases the congestion window by one maximum packet
   size per congestion window acknowledged.  When a loss is detected,
   NewReno halves the congestion window and sets the slow start
   threshold to the new congestion window.

6.4.  Recovery Period

   A recovery period is entered when loss or ECN-CE marking of a packet
   is detected.  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 limits congestion window reduction to once per
   round trip.  During recovery, the congestion window remains unchanged
   irrespective of new losses or increases in the ECN-CE counter.

6.5.  Ignoring Loss of Undecryptable Packets

   During the handshake, some packet protection keys might not be
   available when a packet arrives.  In particular, Handshake and 0-RTT
   packets cannot be processed until the Initial packets arrive, and
   1-RTT packets cannot be processed until the handshake completes.
   Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets
   that might arrive before the peer has packet protection keys to
   process those packets.












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6.6.  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.

6.7.  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.  This duration is computed as follows:

   (smoothed_rtt + 4 * rttvar + max_ack_delay) *
       kPersistentCongestionThreshold

   For example, assume:

   smoothed_rtt = 1 rttvar = 0 max_ack_delay = 0
   kPersistentCongestionThreshold = 3

   If an ack-eliciting packet is sent at time = 0, the following
   scenario would illustrate persistent congestion:

                     +-----+------------------------+
                     | t=0 | Send Pkt #1 (App Data) |
                     +=====+========================+
                     | t=1 | Send Pkt #2 (PTO 1)    |
                     +-----+------------------------+
                     | t=3 | Send Pkt #3 (PTO 2)    |
                     +-----+------------------------+
                     | t=7 | Send Pkt #4 (PTO 3)    |
                     +-----+------------------------+
                     | t=8 | Recv ACK of Pkt #4     |
                     +-----+------------------------+

                                 Table 1

   The first three packets are determined to be lost when the
   acknowlegement of packet 4 is received at t=8.  The congestion period
   is calculated as the time between the oldest and newest lost packets:



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   (3 - 0) = 3.  The duration for persistent congestion is equal to: (1
   * kPersistentCongestionThreshold) = 3.  Because the threshold was
   reached and because none of the packets between the oldest and the
   newest packets are acknowledged, the network is considered to have
   experienced persistent congestion.

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

6.8.  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.  For example, a pacer might distribute the
   congestion window over the smoothed RTT when used with a window-based
   controller, and a pacer might use the rate estimate of a rate-based
   controller.

   An implementation should take care to architect its congestion
   controller to work well with a pacer.  For instance, a pacer might
   wrap the congestion controller and control the availability of the
   congestion window, or a pacer might pace out packets handed to it by
   the congestion controller.  Timely delivery of ACK frames is
   important for efficient loss recovery.  Packets containing only ACK
   frames should therefore not be paced, to avoid delaying their
   delivery to the peer.

   Sending multiple packets into the network without any delay between
   them creates a packet burst that might cause short-term congestion
   and losses.  Implementations MUST either use pacing or limit such
   bursts to the initial congestion window, which is recommended to be
   the minimum of 10 * max_datagram_size and max(2* max_datagram_size,
   14720)), where max_datagram_size is the current maximum size of a
   datagram for the connection, not including UDP or IP overhead.

   As an example of a well-known and publicly available implementation
   of a flow pacer, implementers are referred to the Fair Queue packet
   scheduler (fq qdisc) in Linux (3.11 onwards).









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6.9.  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 credit.

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

   A sender that paces packets (see Section 6.8) might delay sending
   packets and not fully utilize the congestion window due to this
   delay.  A sender should not consider itself application limited if it
   would have fully utilized the congestion window without pacing delay.

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

7.  Security Considerations

7.1.  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.

7.2.  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.
   Endpoints can use PADDING frames or bundle acknowledgments with other
   frames to reduce leaked information.

7.3.  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



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   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.

8.  IANA Considerations

   This document has no IANA actions.  Yet.

9.  References

9.1.  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-27, 9 March 2020,
              <https://tools.ietf.org/html/draft-ietf-quic-tls-27>.

   [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-27, 9 March
              2020, <https://tools.ietf.org/html/draft-ietf-quic-
              transport-27>.

   [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>.

   [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>.

9.2.  Informative References



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   [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:
              a time-based fast loss detection algorithm for TCP", Work
              in Progress, Internet-Draft, draft-ietf-tcpm-rack-07, 17
              January 2020, <http://www.ietf.org/internet-drafts/draft-
              ietf-tcpm-rack-07.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>.

   [RFC4653]  Bhandarkar, S., Reddy, A. L. N., Allman, M., and E.
              Blanton, "Improving the Robustness of TCP to Non-
              Congestion Events", RFC 4653, DOI 10.17487/RFC4653, August
              2006, <https://www.rfc-editor.org/info/rfc4653>.

   [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>.

   [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



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              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 5.

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.

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







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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 MaxAckDelay.

   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 RECOMMENDED
      value is 3.

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

   kGranularity:  Timer granularity.  This is a system-dependent value.
      However, implementations SHOULD use a value no smaller than 1ms.

   kInitialRtt:  The RTT used before an RTT sample is taken.  The
      RECOMMENDED value is 500ms.

   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 [RFC6298]

   rttvar:  The RTT variation, computed as described in [RFC6298]

   min_rtt:  The minimum RTT seen in the connection, ignoring ack delay.

   max_ack_delay:  The maximum amount of time by which the receiver
      intends to delay acknowledgments for packets in the
      ApplicationData 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_sent_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 = 0
      rttvar = 0
      min_rtt = 0
      max_ack_delay = 0
      for pn_space in [ Initial, Handshake, ApplicationData ]:
        largest_acked_packet[pn_space] = infinite
        time_of_last_sent_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_sent_ack_eliciting_packet[pn_space] = now
        OnPacketSentCC(sent_bytes)
        sent_packets[pn_space][packet_number].size = sent_bytes
        SetLossDetectionTimer()

A.6.  On Receiving an Acknowledgment

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

   Pseudocode for OnAckReceived and UpdateRtt follow:

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



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     // Nothing to do if there are no newly acked packets.
     newly_acked_packets = DetermineNewlyAckedPackets(ack, pn_space)
     if (newly_acked_packets.empty()):
       return

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

     // Process ECN information if present.
     if (ACK frame contains ECN information):
         ProcessECN(ack, pn_space)

     for acked_packet in newly_acked_packets:
       OnPacketAcked(acked_packet.packet_number, pn_space)

     DetectLostPackets(pn_space)

     pto_count = 0

     SetLossDetectionTimer()


   UpdateRtt(ack_delay):
     // First RTT sample.
     if (smoothed_rtt == 0):
       min_rtt = latest_rtt
       smoothed_rtt = latest_rtt
       rttvar = latest_rtt / 2
       return

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

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



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     smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt

A.7.  On Packet Acknowledgment

   When a packet is acknowledged for the first time, the following
   OnPacketAcked function is called.  Note that a single ACK frame may
   newly acknowledge several packets.  OnPacketAcked must be called once
   for each of these newly acknowledged packets.

   OnPacketAcked takes two parameters: acked_packet, which is the struct
   detailed in Appendix A.1.1, and the packet number space that this ACK
   frame was sent for.

   Pseudocode for OnPacketAcked follows:

      OnPacketAcked(acked_packet, pn_space):
        if (acked_packet.in_flight):
          OnPacketAckedCC(acked_packet)
        sent_packets[pn_space].remove(acked_packet.packet_number)

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 SHOULD
   fire immediately.

   Pseudocode for SetLossDetectionTimer follows:


















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   GetEarliestTimeAndSpace(times):
     time = times[Initial]
     space = Initial
     for pn_space in [ Handshake, ApplicationData ]:
       if (times[pn_space] != 0 &&
           (time == 0 || times[pn_space] < time) &&
           # Skip ApplicationData until handshake completion.
           (pn_space != ApplicationData ||
             IsHandshakeComplete()):
         time = times[pn_space];
         space = pn_space
     return time, space

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

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

     if (no ack-eliciting packets in flight &&
         PeerNotAwaitingAddressValidation()):
       loss_detection_timer.cancel()
       return

     // Use a default timeout if there are no RTT measurements
     if (smoothed_rtt == 0):
       timeout = 2 * kInitialRtt
     else:
       // Calculate PTO duration
       timeout = smoothed_rtt + max(4 * rttvar, kGranularity) +
         max_ack_delay
     timeout = timeout * (2 ^ pto_count)

     sent_time, _ = GetEarliestTimeAndSpace(
       time_of_last_sent_ack_eliciting_packet)
     loss_detection_timer.update(sent_time + timeout)





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A.9.  On Timeout

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

   Pseudocode for OnLossDetectionTimeout follows:

   OnLossDetectionTimeout():
     earliest_loss_time, pn_space =
       GetEarliestTimeAndSpace(loss_time)
     if (earliest_loss_time != 0):
       // Time threshold loss Detection
       DetectLostPackets(pn_space)
       SetLossDetectionTimer()
       return

     if (endpoint is client without 1-RTT keys):
       // Client sends an anti-deadlock packet: Initial is padded
       // to earn more anti-amplification credit,
       // a Handshake packet proves address ownership.
       if (has Handshake keys):
         SendOneAckElicitingHandshakePacket()
       else:
         SendOneAckElicitingPaddedInitialPacket()
     else:
       // PTO. Send new data if available, else retransmit old data.
       // If neither is available, send a single PING frame.
       _, pn_space = GetEarliestTimeAndSpace(
         time_of_last_sent_ack_eliciting_packet)
       SendOneOrTwoAckElicitingPackets(pn_space)

     pto_count++
     SetLossDetectionTimer()

A.10.  Detecting Lost Packets

   DetectLostPackets is called every time an ACK is received and
   operates on the sent_packets for that packet number space.

   Pseudocode for DetectLostPackets follows:











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

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

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

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

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

     // Inform the congestion controller of lost packets and
     // let it decide whether to retransmit immediately.
     if (!lost_packets.empty()):
       OnPacketsLost(lost_packets)

Appendix B.  Congestion Control Pseudocode

   We now describe an example implementation of the congestion
   controller described in Section 6.

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 amount of data in
      flight, in bytes.  The RECOMMENDED value is the minimum of 10 *
      max_datagram_size and max(2 * max_datagram_size, 14720)).  This



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      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.

   kMinimumWindow:  Minimum congestion window in bytes.  The RECOMMENDED
      value is 2 * max_datagram_size.

   kLossReductionFactor:  Reduction in congestion window when a new loss
      event is detected.  The RECOMMENDED value is 0.5.

   kPersistentCongestionThreshold:  Period of time for persistent
      congestion to be established, specified as a PTO multiplier.  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) [RACK], before establishing persistent congestion, as TCP
      does with a Retransmission Timeout (RTO) [RFC5681].  The
      RECOMMENDED value for kPersistentCongestionThreshold is 3, which
      is approximately equivalent to having two TLPs before an RTO in
      TCP.

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 acked 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.





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   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:

      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 OnPacketAcked and is supplied with the
   acked_packet from sent_packets.


















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      InCongestionRecovery(sent_time):
        return sent_time <= congestion_recovery_start_time

      OnPacketAckedCC(acked_packet):
        // Remove from bytes_in_flight.
        bytes_in_flight -= acked_packet.size
        if (InCongestionRecovery(acked_packet.time_sent)):
          // Do not increase congestion window in recovery period.
          return
        if (IsAppOrFlowControlLimited()):
          // Do not increase congestion_window if application
          // limited or flow control limited.
          return
        if (congestion_window < ssthresh):
          // Slow start.
          congestion_window += acked_packet.size
        else:
          // Congestion avoidance.
          congestion_window += max_datagram_size * acked_packet.size
              / congestion_window

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.

      CongestionEvent(sent_time):
        // Start a new congestion event if packet was sent after the
        // start of the previous congestion recovery period.
        if (!InCongestionRecovery(sent_time)):
          congestion_recovery_start_time = Now()
          congestion_window *= kLossReductionFactor
          congestion_window = max(congestion_window, kMinimumWindow)
          ssthresh = congestion_window

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
          CongestionEvent(sent_packets[ack.largest_acked].time_sent)




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B.8.  On Packets Lost

   Invoked from DetectLostPackets when packets are deemed lost.

      InPersistentCongestion(largest_lost_packet):
        pto = smoothed_rtt + max(4 * rttvar, kGranularity) +
          max_ack_delay
        congestion_period = pto * kPersistentCongestionThreshold
        // Determine if all packets in the time period before the
        // newest lost packet, including the edges, are marked
        // lost
        return AreAllPacketsLost(largest_lost_packet,
                                 congestion_period)

      OnPacketsLost(lost_packets):
        // Remove lost packets from bytes_in_flight.
        for (lost_packet : lost_packets):
          bytes_in_flight -= lost_packet.size
        largest_lost_packet = lost_packets.last()
        CongestionEvent(largest_lost_packet.time_sent)

        // Collapse congestion window if persistent congestion
        if (InPersistentCongestion(largest_lost_packet)):
          congestion_window = kMinimumWindow

Appendix C.  Change Log

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

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

C.1.  Since draft-ietf-quic-recovery-26

   No changes.

C.2.  Since draft-ietf-quic-recovery-25

   No significant changes.

C.3.  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)




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C.4.  Since draft-ietf-quic-recovery-23

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

   *  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.5.  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.6.  Since draft-ietf-quic-recovery-21

   *  No changes

C.7.  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.8.  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)




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

   *  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.9.  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.10.  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)




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

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

C.11.  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.12.  Since draft-ietf-quic-recovery-14

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

   *  Merge ACK and ACK_ECN (#1783)

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

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




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

   *  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.14.  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.15.  Since draft-ietf-quic-recovery-11

   No significant changes.

C.16.  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.17.  Since draft-ietf-quic-recovery-09

   No significant changes.

C.18.  Since draft-ietf-quic-recovery-08

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

C.19.  Since draft-ietf-quic-recovery-07

   *  Include Ack Delay in RTO(and TLP) computations (#981)

   *  Ack Delay in SRTT computation (#961)



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   *  Default RTT and Slow Start (#590)

   *  Many editorial fixes.

C.20.  Since draft-ietf-quic-recovery-06

   No significant changes.

C.21.  Since draft-ietf-quic-recovery-05

   *  Add more congestion control text (#776)

C.22.  Since draft-ietf-quic-recovery-04

   No significant changes.

C.23.  Since draft-ietf-quic-recovery-03

   No significant changes.

C.24.  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.25.  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.26.  Since draft-ietf-quic-recovery-00

   *  Improved description of constants and ACK behavior





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C.27.  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 Balasubramaniam.

Acknowledgments

Authors' Addresses

   Jana Iyengar (editor)
   Fastly

   Email: jri.ietf@gmail.com


   Ian Swett (editor)
   Google

   Email: ianswett@google.com




















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