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Versions: (draft-iyengar-quic-loss-recovery) 00 01 02 03 04 05 06 07 08 09 10 11 12

QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                           I. Swett, Ed.
Expires: November 23, 2018                                        Google
                                                            May 22, 2018


               QUIC Loss Detection and Congestion Control
                      draft-ietf-quic-recovery-12

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 [1].

   Working Group information can be found at https://github.com/quicwg
   [2]; source code and issues list for this draft can be found at
   https://github.com/quicwg/base-drafts/labels/-recovery [3].

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 November 23, 2018.

Copyright Notice

   Copyright (c) 2018 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  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Notational Conventions  . . . . . . . . . . . . . . . . .   4
   2.  Design of the QUIC Transmission Machinery . . . . . . . . . .   4
     2.1.  Relevant Differences Between QUIC and TCP . . . . . . . .   4
       2.1.1.  Monotonically Increasing Packet Numbers . . . . . . .   5
       2.1.2.  No Reneging . . . . . . . . . . . . . . . . . . . . .   5
       2.1.3.  More ACK Ranges . . . . . . . . . . . . . . . . . . .   5
       2.1.4.  Explicit Correction For Delayed ACKs  . . . . . . . .   5
   3.  Loss Detection  . . . . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Computing the RTT estimate  . . . . . . . . . . . . . . .   6
     3.2.  Ack-based Detection . . . . . . . . . . . . . . . . . . .   6
       3.2.1.  Fast Retransmit . . . . . . . . . . . . . . . . . . .   6
       3.2.2.  Early Retransmit  . . . . . . . . . . . . . . . . . .   7
     3.3.  Timer-based Detection . . . . . . . . . . . . . . . . . .   8
       3.3.1.  Handshake Timeout . . . . . . . . . . . . . . . . . .   8
       3.3.2.  Tail Loss Probe . . . . . . . . . . . . . . . . . . .   9
       3.3.3.  Retransmission Timeout  . . . . . . . . . . . . . . .  10
     3.4.  Generating Acknowledgements . . . . . . . . . . . . . . .  11
       3.4.1.  ACK Ranges  . . . . . . . . . . . . . . . . . . . . .  11
       3.4.2.  Receiver Tracking of ACK Frames . . . . . . . . . . .  12
     3.5.  Pseudocode  . . . . . . . . . . . . . . . . . . . . . . .  12
       3.5.1.  Constants of interest . . . . . . . . . . . . . . . .  12
       3.5.2.  Variables of interest . . . . . . . . . . . . . . . .  13
       3.5.3.  Initialization  . . . . . . . . . . . . . . . . . . .  14
       3.5.4.  On Sending a Packet . . . . . . . . . . . . . . . . .  15
       3.5.5.  On Ack Receipt  . . . . . . . . . . . . . . . . . . .  16
       3.5.6.  On Packet Acknowledgment  . . . . . . . . . . . . . .  17
       3.5.7.  Setting the Loss Detection Alarm  . . . . . . . . . .  18
       3.5.8.  On Alarm Firing . . . . . . . . . . . . . . . . . . .  20
       3.5.9.  Detecting Lost Packets  . . . . . . . . . . . . . . .  20
     3.6.  Discussion  . . . . . . . . . . . . . . . . . . . . . . .  21
   4.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .  22
     4.1.  Slow Start  . . . . . . . . . . . . . . . . . . . . . . .  22
     4.2.  Congestion Avoidance  . . . . . . . . . . . . . . . . . .  22
     4.3.  Recovery Period . . . . . . . . . . . . . . . . . . . . .  22
     4.4.  Tail Loss Probe . . . . . . . . . . . . . . . . . . . . .  23



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     4.5.  Retransmission Timeout  . . . . . . . . . . . . . . . . .  23
     4.6.  Pacing  . . . . . . . . . . . . . . . . . . . . . . . . .  23
     4.7.  Pseudocode  . . . . . . . . . . . . . . . . . . . . . . .  24
       4.7.1.  Constants of interest . . . . . . . . . . . . . . . .  24
       4.7.2.  Variables of interest . . . . . . . . . . . . . . . .  24
       4.7.3.  Initialization  . . . . . . . . . . . . . . . . . . .  24
       4.7.4.  On Packet Sent  . . . . . . . . . . . . . . . . . . .  25
       4.7.5.  On Packet Acknowledgement . . . . . . . . . . . . . .  25
       4.7.6.  On Packets Lost . . . . . . . . . . . . . . . . . . .  25
       4.7.7.  On Retransmission Timeout Verified  . . . . . . . . .  26
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  26
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  26
     6.1.  Normative References  . . . . . . . . . . . . . . . . . .  26
     6.2.  Informative References  . . . . . . . . . . . . . . . . .  26
     6.3.  URIs  . . . . . . . . . . . . . . . . . . . . . . . . . .  27
   Appendix A.  Acknowledgments  . . . . . . . . . . . . . . . . . .  28
   Appendix B.  Change Log . . . . . . . . . . . . . . . . . . . . .  28
     B.1.  Since draft-ietf-quic-recovery-10 . . . . . . . . . . . .  28
     B.2.  Since draft-ietf-quic-recovery-09 . . . . . . . . . . . .  28
     B.3.  Since draft-ietf-quic-recovery-08 . . . . . . . . . . . .  28
     B.4.  Since draft-ietf-quic-recovery-07 . . . . . . . . . . . .  28
     B.5.  Since draft-ietf-quic-recovery-06 . . . . . . . . . . . .  28
     B.6.  Since draft-ietf-quic-recovery-05 . . . . . . . . . . . .  29
     B.7.  Since draft-ietf-quic-recovery-04 . . . . . . . . . . . .  29
     B.8.  Since draft-ietf-quic-recovery-03 . . . . . . . . . . . .  29
     B.9.  Since draft-ietf-quic-recovery-02 . . . . . . . . . . . .  29
     B.10. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . .  29
     B.11. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . .  29
     B.12. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . .  29
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  30

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







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1.1.  Notational Conventions

   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.

2.  Design of the QUIC Transmission Machinery

   All transmissions in QUIC are sent with a packet-level header, which
   includes a packet sequence number (referred to below as a packet
   number).  These packet numbers never repeat in the lifetime of a
   connection, and are monotonically increasing, which prevents
   ambiguity.  This fundamental design decision obviates the need for
   disambiguating between transmissions and retransmissions and
   eliminates significant complexity from QUIC's interpretation of TCP
   loss detection mechanisms.

   Every packet may contain several frames.  We outline the frames that
   are important to the loss detection and congestion control machinery
   below.

   o  Retransmittable frames are those that count towards bytes in
      flight and need acknowledgement.  The most common are STREAM
      frames, which typically contain application data.

   o  Retransmittable packets are those that contain at least one
      retransmittable frame.

   o  Crypto handshake data is sent on stream 0, and uses the
      reliability machinery of QUIC underneath.

   o  ACK frames contain acknowledgment information.  ACK frames contain
      one or more ranges of acknowledged packets.

2.1.  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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2.1.1.  Monotonically Increasing Packet Numbers

   TCP conflates transmission sequence number at the sender with
   delivery sequence number at the receiver, which results in
   retransmissions of the same data carrying the same sequence number,
   and consequently to problems caused by "retransmission ambiguity".
   QUIC separates the two: QUIC uses a packet number for transmissions,
   and any data that is to be delivered to the receiving application(s)
   is sent in one or more streams, with delivery order determined by
   stream offsets encoded within STREAM frames.

   QUIC's packet number is strictly increasing, and directly encodes
   transmission order.  A higher QUIC packet number signifies that the
   packet was sent later, and a lower QUIC packet number signifies that
   the packet was sent earlier.  When a packet containing frames is
   deemed 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.

2.1.2.  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.

2.1.3.  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.

2.1.4.  Explicit Correction For Delayed ACKs

   QUIC ACKs explicitly encode the delay incurred at the receiver
   between when a packet is received and when the corresponding ACK is
   sent.  This allows the receiver of the ACK to adjust for receiver
   delays, specifically the delayed ack timer, when estimating the path
   RTT.  This mechanism also allows a receiver to measure and report the



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   delay from when a packet was received by the OS kernel, which is
   useful in receivers which may incur delays such as context-switch
   latency before a userspace QUIC receiver processes a received packet.

3.  Loss Detection

   QUIC senders use both ack information and timeouts to detect lost
   packets, and this section provides a description of these algorithms.
   Estimating the network round-trip time (RTT) is critical to these
   algorithms and is described first.

3.1.  Computing the RTT estimate

   RTT is calculated when an ACK frame arrives by computing the
   difference between the current time and the time the largest newly
   acked packet was sent.  If no packets are newly acknowledged, RTT
   cannot be calculated.  When RTT is calculated, the ack delay field
   from the ACK frame SHOULD be subtracted from the RTT as long as the
   result is larger than the Min RTT.  If the result is smaller than the
   min_rtt, the RTT should be used, but the ack delay field should be
   ignored.

   Like TCP, QUIC calculates both smoothed RTT and RTT variance similar
   to those specified in [RFC6298].

   Min RTT is the minimum RTT measured over the connection, prior to
   adjusting by ack delay.  Ignoring ack delay for min RTT prevents
   intentional or unintentional underestimation of min RTT, which in
   turn prevents underestimating smoothed RTT.

3.2.  Ack-based Detection

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

3.2.1.  Fast Retransmit

   An unacknowledged packet is marked as lost when an acknowledgment is
   received for a packet that was sent a threshold number of packets
   (kReorderingThreshold) after the unacknowledged packet.  Receipt of
   the ack indicates that a later packet was received, while
   kReorderingThreshold provides some tolerance for reordering of
   packets in the network.

   The RECOMMENDED initial value for kReorderingThreshold is 3.




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   We derive this default from recommendations for TCP loss recovery
   [RFC5681] [RFC6675].  It is possible for networks to exhibit higher
   degrees of reordering, causing a sender to detect spurious losses.
   Detecting spurious losses leads to unnecessary retransmissions and
   may result in degraded performance due to the actions of the
   congestion controller upon detecting loss.  Implementers MAY use
   algorithms developed for TCP, such as TCP-NCR [RFC4653], to improve
   QUIC's reordering resilience, though care should be taken to map TCP
   specifics to QUIC correctly.  Similarly, using time-based loss
   detection to deal with reordering, such as in PR-TCP, should be more
   readily usable in QUIC.  Making QUIC deal with such networks is
   important open research, and implementers are encouraged to explore
   this space.

3.2.2.  Early Retransmit

   Unacknowledged packets close to the tail may have fewer than
   kReorderingThreshold retransmittable packets sent after them.  Loss
   of such packets cannot be detected via Fast Retransmit.  To enable
   ack-based loss detection of such packets, receipt of an
   acknowledgment for the last outstanding retransmittable packet
   triggers the Early Retransmit process, as follows.

   If there are unacknowledged retransmittable packets still pending,
   they should be marked as lost.  To compensate for the reduced
   reordering resilience, the sender SHOULD set an alarm for a small
   period of time.  If the unacknowledged retransmittable packets are
   not acknowledged during this time, then these packets MUST be marked
   as lost.

   An endpoint SHOULD set the alarm such that a packet is marked as lost
   no earlier than 1.25 * max(SRTT, latest_RTT) since when it was sent.

   Using max(SRTT, latest_RTT) protects from the two following cases:

   o  the latest RTT sample is lower than the SRTT, perhaps due to
      reordering where packet whose ack triggered the Early Retransit
      process encountered a shorter path;

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

   The 1.25 multiplier increases reordering resilience.  Implementers
   MAY experiment with using other multipliers, bearing in mind that a
   lower multiplier reduces reordering resilience and increases spurious
   retransmissions, and a higher multipler increases loss recovery
   delay.



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   This mechanism is based on Early Retransmit for TCP [RFC5827].
   However, [RFC5827] does not include the alarm described above.  Early
   Retransmit is prone to spurious retransmissions due to its reduced
   reordering resilence without the alarm.  This observation led Linux
   TCP implementers to implement an alarm for TCP as well, and this
   document incorporates this advancement.

3.3.  Timer-based Detection

   Timer-based loss detection implements a handshake retransmission
   timer that is optimized for QUIC as well as the spirit of TCP's Tail
   Loss Probe and Retransmission Timeout mechanisms.

3.3.1.  Handshake Timeout

   Handshake packets, which contain STREAM frames for stream 0, are
   critical to QUIC transport and crypto negotiation, so a separate
   alarm is used for them.

   The initial handshake timeout SHOULD be set to twice the initial RTT.

   At the beginning, there are no prior RTT samples within a connection.
   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, or if the network changes, the
   initial RTT SHOULD be set to 100ms.

   When a handshake packet is sent, the sender SHOULD set an alarm for
   the handshake timeout period.

   When the alarm fires, the sender MUST retransmit all unacknowledged
   handshake data, by calling RetransmitAllUnackedHandshakeData().  On
   each consecutive firing of the handshake alarm, the sender SHOULD
   double the handshake timeout and set an alarm for this period.

   When an acknowledgement is received for a handshake packet, the new
   RTT is computed and the alarm SHOULD be set for twice the newly
   computed smoothed RTT.

   Handshake data may be cancelled by handshake state transitions.  In
   particular, all non-protected data SHOULD no longer be transmitted
   once packet protection is available.

   (TODO: Work this section some more.  Add text on client vs. server,
   and on stateless retry.)




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3.3.2.  Tail Loss Probe

   The algorithm described in this section is an adaptation of the Tail
   Loss Probe algorithm proposed for TCP [TLP].

   A packet sent at the tail is particularly vulnerable to slow loss
   detection, since acks of subsequent packets are needed to trigger
   ack-based detection.  To ameliorate this weakness of tail packets,
   the sender schedules an alarm when the last retransmittable packet
   before quiescence is transmitted.  When this alarm fires, a Tail Loss
   Probe (TLP) packet is sent to evoke an acknowledgement from the
   receiver.

   The alarm duration, or Probe Timeout (PTO), is set based on the
   following conditions:

   o  PTO SHOULD be scheduled for max(1.5*SRTT+MaxAckDelay,
      kMinTLPTimeout)

   o  If RTO (Section 3.3.3) is earlier, schedule a TLP alarm in its
      place.  That is, PTO SHOULD be scheduled for min(RTO, PTO).

   MaxAckDelay is the maximum ack delay supplied in an incoming ACK
   frame.  MaxAckDelay excludes ack delays that aren't included in an
   RTT sample because they're too large and excludes those which
   reference an ack-only packet.

   QUIC diverges from TCP by calculating MaxAckDelay dynamically,
   instead of assuming a constant delayed ack timeout for all
   connections.  QUIC includes this in all probe timeouts, because it
   assume the ack delay may come into play, regardless of the number of
   packets outstanding.  TCP's TLP assumes if at least 2 packets are
   outstanding, acks will not be delayed.

   A PTO value of at least 1.5*SRTT ensures that the ACK is overdue.
   The 1.5 is based on [TLP], but implementations MAY experiment with
   other constants.

   To reduce latency, it is RECOMMENDED that the sender set and allow
   the TLP alarm to fire twice before setting an RTO alarm.  In other
   words, when the TLP alarm fires the first time, a TLP packet is sent,
   and it is RECOMMENDED that the TLP alarm be scheduled for a second
   time.  When the TLP alarm fires the second time, a second TLP packet
   is sent, and an RTO alarm SHOULD be scheduled Section 3.3.3.

   A TLP packet SHOULD carry new data when possible.  If new data is
   unavailable or new data cannot be sent due to flow control, a TLP
   packet MAY retransmit unacknowledged data to potentially reduce



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   recovery time.  Since a TLP alarm is used to send a probe into the
   network prior to establishing any packet loss, prior unacknowledged
   packets SHOULD NOT be marked as lost when a TLP alarm fires.

   A sender may not know that a packet being sent is a tail packet.
   Consequently, a sender may have to arm or adjust the TLP alarm on
   every sent retransmittable packet.

3.3.3.  Retransmission Timeout

   A Retransmission Timeout (RTO) alarm is the final backstop for loss
   detection.  The algorithm used in QUIC is based on the RTO algorithm
   for TCP [RFC5681] and is additionally resilient to spurious RTO
   events [RFC5682].

   When the last TLP packet is sent, an alarm is scheduled for the RTO
   period.  When this alarm fires, the sender sends two packets, to
   evoke acknowledgements from the receiver, and restarts the RTO alarm.

   Similar to TCP [RFC6298], the RTO period is set based on the
   following conditions:

   o  When the final TLP packet is sent, the RTO period is set to
      max(SRTT + 4*RTTVAR + MaxAckDelay, kMinRTOTimeout)

   o  When an RTO alarm fires, the RTO period is doubled.

   The sender typically has incurred a high latency penalty by the time
   an RTO alarm fires, and this penalty increases exponentially in
   subsequent consecutive RTO events.  Sending a single packet on an RTO
   event therefore makes the connection very sensitive to single packet
   loss.  Sending two packets instead of one significantly increases
   resilience to packet drop in both directions, thus reducing the
   probability of consecutive RTO events.

   QUIC's RTO algorithm differs from TCP in that the firing of an RTO
   alarm is not considered a strong enough signal of packet loss, so
   does not result in an immediate change to congestion window or
   recovery state.  An RTO alarm fires only when there's a prolonged
   period of network silence, which could be caused by a change in the
   underlying network RTT.

   QUIC also diverges from TCP by including MaxAckDelay in the RTO
   period.  QUIC is able to explicitly model delay at the receiver via
   the ack delay field in the ACK frame.  Since QUIC corrects for this
   delay in its SRTT and RTTVAR computations, it is necessary to add
   this delay explicitly in the TLP and RTO computation.




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   When an acknowledgment is received for a packet sent on an RTO event,
   any unacknowledged packets with lower packet numbers than those
   acknowledged MUST be marked as lost.

   A packet sent when an RTO alarm fires MAY carry new data if available
   or unacknowledged data to potentially reduce recovery time.  Since
   this packet is sent as a probe into the network prior to establishing
   any packet loss, prior unacknowledged packets SHOULD NOT be marked as
   lost.

   A packet sent on an RTO alarm MUST NOT be blocked by the sender's
   congestion controller.  A sender MUST however count these bytes as
   additional bytes in flight, since this packet adds network load
   without establishing packet loss.

3.4.  Generating Acknowledgements

   QUIC SHOULD delay sending acknowledgements in response to packets,
   but MUST NOT excessively delay acknowledgements of packets containing
   non-ack frames.  Specifically, implementaions MUST attempt to enforce
   a maximum ack delay to avoid causing the peer spurious timeouts.  The
   default maximum ack delay in QUIC is 25ms.

   An acknowledgement MAY be sent for every second full-sized packet, as
   TCP does [RFC5681], or may be sent less frequently, as long as the
   delay does not exceed the maximum ack delay.  QUIC recovery
   algorithms do not assume the peer generates an acknowledgement
   immediately when receiving a second full-sized packet.

   Out-of-order packets SHOULD be acknowledged more quickly, in order to
   accelerate loss recovery.  The receiver SHOULD send an immediate ACK
   when it receives a new packet which is not one greater than the
   largest received packet number.

   As an optimization, a receiver MAY process multiple packets before
   sending any ACK frames in response.  In this case they can determine
   whether an immediate or delayed acknowledgement should be generated
   after processing incoming packets.

3.4.1.  ACK Ranges

   When an ACK frame is sent, one or more ranges of acknowledged packets
   are included.  Including older packets reduces the chance of spurious
   retransmits caused by losing previously sent ACK frames, at the cost
   of larger ACK frames.

   ACK frames SHOULD always acknowledge the most recently received
   packets, and the more out-of-order the packets are, the more



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   important it is to send an updated ACK frame quickly, to prevent the
   peer from declaring a packet as lost and spuriusly retransmitting the
   frames it contains.

   Below is one recommended approach for determining what packets to
   include in an ACK frame.

3.4.2.  Receiver Tracking of ACK Frames

   When a packet containing an ACK frame is sent, the largest
   acknowledged in that frame may be saved.  When a packet containing an
   ACK frame is acknowledged, the receiver can stop acknowledging
   packets less than or equal to the largest acknowledged in the sent
   ACK frame.

   In cases without ACK frame loss, this algorithm allows for a minimum
   of 1 RTT of reordering.  In cases with ACK frame loss, this approach
   does not guarantee that every acknowledgement is seen by the sender
   before it is no longer included in the ACK frame.  Packets could be
   received out of order and all subsequent ACK frames containing them
   could be lost.  In this case, the loss recovery algorithm may cause
   spurious retransmits, but the sender will continue making forward
   progress.

3.5.  Pseudocode

3.5.1.  Constants of interest

   Constants used in loss recovery are based on a combination of RFCs,
   papers, and common practice.  Some may need to be changed or
   negotiated in order to better suit a variety of environments.

   kMaxTLPs (default 2):  Maximum number of tail loss probes before an
      RTO fires.

   kReorderingThreshold (default 3):  Maximum reordering in packet
      number space before FACK style loss detection considers a packet
      lost.

   kTimeReorderingFraction (default 1/8):  Maximum reordering in time
      space before time based loss detection considers a packet lost.
      In fraction of an RTT.

   kUsingTimeLossDetection (default false):  Whether time based loss
      detection is in use.  If false, uses FACK style loss detection.

   kMinTLPTimeout (default 10ms):  Minimum time in the future a tail
      loss probe alarm may be set for.



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   kMinRTOTimeout (default 200ms):  Minimum time in the future an RTO
      alarm may be set for.

   kDelayedAckTimeout (default 25ms):  The length of the peer's delayed
      ack timer.

   kDefaultInitialRtt (default 100ms):  The default RTT used before an
      RTT sample is taken.

3.5.2.  Variables of interest

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

   loss_detection_alarm:  Multi-modal alarm used for loss detection.

   handshake_count:  The number of times all unacknowledged handshake
      data has been retransmitted without receiving an ack.

   tlp_count:  The number of times a tail loss probe has been sent
      without receiving an ack.

   rto_count:  The number of times an rto has been sent without
      receiving an ack.

   largest_sent_before_rto:  The last packet number sent prior to the
      first retransmission timeout.

   time_of_last_sent_retransmittable_packet:  The time the most recent
      retransmittable packet was sent.

   time_of_last_sent_handshake_packet:  The time the most recent packet
      containing handshake data was sent.

   largest_sent_packet:  The packet number of the most recently sent
      packet.

   largest_acked_packet:  The largest packet number acknowledged in an
      ACK frame.

   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 variance, computed as described in [RFC6298]




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

   max_ack_delay:  The maximum ack delay in an incoming ACK frame for
      this connection.  Excludes ack delays for ack only packets and
      those that create an RTT sample less than min_rtt.

   reordering_threshold:  The largest packet number gap between the
      largest acked retransmittable packet and an unacknowledged
      retransmittable packet before it is declared lost.

   time_reordering_fraction:  The reordering window as a fraction of
      max(smoothed_rtt, latest_rtt).

   loss_time:  The time at which the next packet will be considered lost
      based on early transmit or exceeding the reordering window in
      time.

   sent_packets:  An association of packet numbers to information about
      them, including a number field indicating the packet number, a
      time field indicating the time a packet was sent, a boolean
      indicating whether the packet is ack only, and a bytes field
      indicating the packet's size.  sent_packets is ordered by packet
      number, and packets remain in sent_packets until acknowledged or
      lost.

3.5.3.  Initialization

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






















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      loss_detection_alarm.reset()
      handshake_count = 0
      tlp_count = 0
      rto_count = 0
      if (kUsingTimeLossDetection)
        reordering_threshold = infinite
        time_reordering_fraction = kTimeReorderingFraction
      else:
        reordering_threshold = kReorderingThreshold
        time_reordering_fraction = infinite
      loss_time = 0
      smoothed_rtt = 0
      rttvar = 0
      min_rtt = infinite
      max_ack_delay = 0
      largest_sent_before_rto = 0
      time_of_last_sent_retransmittable_packet = 0
      time_of_last_sent_handshake_packet = 0
      largest_sent_packet = 0

3.5.4.  On Sending a Packet

   After any packet is sent, be it a new transmission or a rebundled
   transmission, the following OnPacketSent function is called.  The
   parameters to OnPacketSent are as follows:

   o  packet_number: The packet number of the sent packet.

   o  is_ack_only: A boolean that indicates whether a packet only
      contains an ACK frame.  If true, it is still expected an ack will
      be received for this packet, but it is not retransmittable.

   o  is_handshake_packet: A boolean that indicates whether a packet
      contains handshake data.

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

   Pseudocode for OnPacketSent follows:












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    OnPacketSent(packet_number, is_ack_only, is_handshake_packet,
                   sent_bytes):
      largest_sent_packet = packet_number
      sent_packets[packet_number].packet_number = packet_number
      sent_packets[packet_number].time = now
      sent_packets[packet_number].ack_only = is_ack_only
      if !is_ack_only:
        if is_handshake_packet:
          time_of_last_sent_handshake_packet = now
        time_of_last_sent_retransmittable_packet = now
        OnPacketSentCC(sent_bytes)
        sent_packets[packet_number].bytes = sent_bytes
        SetLossDetectionAlarm()

3.5.5.  On Ack Receipt

   When an ack is received, it may acknowledge 0 or more packets.

   Pseudocode for OnAckReceived and UpdateRtt follow:
































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     OnAckReceived(ack):
       largest_acked_packet = ack.largest_acked
       // If the largest acked is newly acked, update the RTT.
       if (sent_packets[ack.largest_acked]):
         latest_rtt = now - sent_packets[ack.largest_acked].time
         UpdateRtt(latest_rtt, ack.ack_delay)
       // Find all newly acked packets.
       for acked_packet in DetermineNewlyAckedPackets():
         OnPacketAcked(acked_packet.packet_number)

       DetectLostPackets(ack.largest_acked_packet)
       SetLossDetectionAlarm()


     UpdateRtt(latest_rtt, ack_delay):
       // min_rtt ignores ack delay.
       min_rtt = min(min_rtt, latest_rtt)
       // Adjust for ack delay if it's plausible.
       if (latest_rtt - min_rtt > ack_delay):
         latest_rtt -= ack_delay
         // Only save into max ack delay if it's used
         // for rtt calculation and is not ack only.
         if (!sent_packets[ack.largest_acked].ack_only)
           max_ack_delay = max(max_ack_delay, ack_delay)
       // Based on {{RFC6298}}.
       if (smoothed_rtt == 0):
         smoothed_rtt = latest_rtt
         rttvar = latest_rtt / 2
       else:
         rttvar_sample = abs(smoothed_rtt - latest_rtt)
         rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
         smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt

3.5.6.  On Packet Acknowledgment

   When a packet is acked 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 acked packets.

   OnPacketAcked takes one parameter, acked_packet, which is the struct
   of the newly acked packet.

   If this is the first acknowledgement following RTO, check if the
   smallest newly acknowledged packet is one sent by the RTO, and if so,
   inform congestion control of a verified RTO, similar to F-RTO
   [RFC5682]




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   Pseudocode for OnPacketAcked follows:

      OnPacketAcked(acked_packet):
        if (!acked_packet.is_ack_only):
          OnPacketAckedCC(acked_packet)
        // If a packet sent prior to RTO was acked, then the RTO
        // was spurious.  Otherwise, inform congestion control.
        if (rto_count > 0 &&
            acked_packet.packet_number > largest_sent_before_rto)
          OnRetransmissionTimeoutVerified()
        handshake_count = 0
        tlp_count = 0
        rto_count = 0
        sent_packets.remove(acked_packet.packet_number)

3.5.7.  Setting the Loss Detection Alarm

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

3.5.7.1.  Handshake Alarm

   When a connection has unacknowledged handshake data, the handshake
   alarm is set and when it expires, all unacknowledgedd handshake data
   is retransmitted.

   When stateless rejects are in use, the connection is considered
   immediately closed once a reject is sent, so no timer is set to
   retransmit the reject.

   Version negotiation packets are always stateless, and MUST be sent
   once per handshake packet that uses an unsupported QUIC version, and
   MAY be sent in response to 0RTT packets.

3.5.7.2.  Tail Loss Probe and Retransmission Alarm

   Tail loss probes [TLP] and retransmission timeouts [RFC6298] are an
   alarm based mechanism to recover from cases when there are
   outstanding retransmittable packets, but an acknowledgement has not
   been received in a timely manner.

   The TLP and RTO timers are armed when there is not unacknowledged
   handshake data.  The TLP alarm is set until the max number of TLP
   packets have been sent, and then the RTO timer is set.




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3.5.7.3.  Early Retransmit Alarm

   Early retransmit [RFC5827] is implemented with a 1/4 RTT timer.  It
   is part of QUIC's time based loss detection, but is always enabled,
   even when only packet reordering loss detection is enabled.

3.5.7.4.  Pseudocode

   Pseudocode for SetLossDetectionAlarm follows:

    SetLossDetectionAlarm():
       // Don't arm the alarm if there are no packets with
       // retransmittable data in flight.
       if (bytes_in_flight == 0):
         loss_detection_alarm.cancel()
         return

       if (handshake packets are outstanding):
         // Handshake retransmission alarm.
         if (smoothed_rtt == 0):
           alarm_duration = 2 * kDefaultInitialRtt
         else:
           alarm_duration = 2 * smoothed_rtt
         alarm_duration = max(alarm_duration + max_ack_delay,
                              kMinTLPTimeout)
         alarm_duration = alarm_duration * (2 ^ handshake_count)
         loss_detection_alarm.set(
           time_of_last_sent_handshake_packet + alarm_duration)
         return;
       else if (loss_time != 0):
         // Early retransmit timer or time loss detection.
         alarm_duration = loss_time -
           time_of_last_sent_retransmittable_packet
       else:
         // RTO or TLP alarm
         // Calculate RTO duration
         alarm_duration =
           smoothed_rtt + 4 * rttvar + max_ack_delay
         alarm_duration = max(alarm_duration, kMinRTOTimeout)
         alarm_duration = alarm_duration * (2 ^ rto_count)
         if (tlp_count < kMaxTLPs):
           // Tail Loss Probe
           tlp_alarm_duration = max(1.5 * smoothed_rtt
                              + max_ack_delay, kMinTLPTimeout)
           alarm_duration = min(tlp_alarm_duration, alarm_duration)

       loss_detection_alarm.set(
         time_of_last_sent_retransmittable_packet + alarm_duration)



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3.5.8.  On Alarm Firing

   QUIC uses one loss recovery alarm, which when set, can be in one of
   several modes.  When the alarm fires, the mode determines the action
   to be performed.

   Pseudocode for OnLossDetectionAlarm follows:

      OnLossDetectionAlarm():
        if (handshake packets are outstanding):
          // Handshake retransmission alarm.
          RetransmitAllUnackedHandshakeData()
          handshake_count++
        else if (loss_time != 0):
          // Early retransmit or Time Loss Detection
          DetectLostPackets(largest_acked_packet)
        else if (tlp_count < kMaxTLPs):
          // Tail Loss Probe.
          SendOnePacket()
          tlp_count++
        else:
          // RTO.
          if (rto_count == 0)
            largest_sent_before_rto = largest_sent_packet
          SendTwoPackets()
          rto_count++

        SetLossDetectionAlarm()

3.5.9.  Detecting Lost Packets

   Packets in QUIC are only considered lost once a larger packet number
   is acknowledged.  DetectLostPackets is called every time an ack is
   received.  If the loss detection alarm fires and the loss_time is
   set, the previous largest acked packet is supplied.

3.5.9.1.  Handshake Packets

   The receiver MUST close the connection with an error of type
   OPTIMISTIC_ACK when receiving an unprotected packet that acks
   protected packets.  The receiver MUST trust protected acks for
   unprotected packets, however.  Aside from this, loss detection for
   handshake packets when an ack is processed is identical to other
   packets.







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3.5.9.2.  Pseudocode

   DetectLostPackets takes one parameter, acked, which is the largest
   acked packet.

   Pseudocode for DetectLostPackets follows:

   DetectLostPackets(largest_acked):
     loss_time = 0
     lost_packets = {}
     delay_until_lost = infinite
     if (kUsingTimeLossDetection):
       delay_until_lost =
         (1 + time_reordering_fraction) *
             max(latest_rtt, smoothed_rtt)
     else if (largest_acked.packet_number == largest_sent_packet):
       // Early retransmit alarm.
       delay_until_lost = 5/4 * max(latest_rtt, smoothed_rtt)
     foreach (unacked < largest_acked.packet_number):
       time_since_sent = now() - unacked.time_sent
       delta = largest_acked.packet_number - unacked.packet_number
       if (time_since_sent > delay_until_lost ||
           delta > reordering_threshold):
         sent_packets.remove(unacked.packet_number)
         if (!unacked.is_ack_only):
           lost_packets.insert(unacked)
       else if (loss_time == 0 && delay_until_lost != infinite):
         loss_time = now() + delay_until_lost - time_since_sent

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

3.6.  Discussion

   The majority of constants were derived from best common practices
   among widely deployed TCP implementations on the internet.
   Exceptions follow.

   A shorter delayed ack time of 25ms was chosen because longer delayed
   acks can delay loss recovery and for the small number of connections
   where less than packet per 25ms is delivered, acking every packet is
   beneficial to congestion control and loss recovery.

   The default initial RTT of 100ms was chosen because it is slightly
   higher than both the median and mean min_rtt typically observed on
   the public internet.



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4.  Congestion Control

   QUIC's congestion control is based on TCP NewReno [RFC6582]
   congestion control to determine the congestion window.  QUIC
   congestion control is specified in bytes due to finer control and the
   ease of appropriate byte counting [RFC3465].

   QUIC hosts MUST NOT send packets if they would increase
   bytes_in_flight (defined in Section 4.7.2) beyond the available
   congestion window, unless the packet is a probe packet sent after the
   TLP or RTO alarm fires, as described in Section 3.3.2 and
   Section 3.3.3.

4.1.  Slow Start

   QUIC begins every connection in slow start and exits slow start upon
   loss.  QUIC re-enters slow start anytime the congestion window is
   less than sshthresh, which typically only occurs after an RTO.  While
   in slow start, QUIC increases the congestion window by the number of
   acknowledged bytes when each ack is processed.

4.2.  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 MSS of bytes 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.

4.3.  Recovery Period

   Recovery is a period of time beginning with detection of a lost
   packet.  Because QUIC retransmits stream data and control frames, not
   packets, it defines the end of recovery as a packet sent after the
   start of recovery being acknowledged.  This is slightly different
   from TCP's definition of recovery ending when the lost packet that
   started recovery is acknowledged.

   During recovery, the congestion window is not increased or decreased.
   As such, multiple lost packets only decrease the congestion window
   once as long as they're lost before exiting recovery.  This causes
   QUIC to decrease the congestion window multiple times if
   retransmisions are lost, but limits the reduction to once per round
   trip.






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4.4.  Tail Loss Probe

   A TLP packet MUST NOT be blocked by the sender's congestion
   controller.  The sender MUST however count these bytes as additional
   bytes-in-flight, since a TLP adds network load without establishing
   packet loss.

   Acknowledgement or loss of tail loss probes are treated like any
   other packet.

4.5.  Retransmission Timeout

   When retransmissions are sent due to a retransmission timeout alarm,
   no change is made to the congestion window until the next
   acknowledgement arrives.  The retransmission timeout is considered
   spurious when this acknowledgement acknowledges packets sent prior to
   the first retransmission timeout.  The retransmission timeout is
   considered valid when this acknowledgement acknowledges no packets
   sent prior to the first retransmission timeout.  In this case, the
   congestion window MUST be reduced to the minimum congestion window
   and slow start is re-entered.

4.6.  Pacing

   This document does not specify a pacer, but it is RECOMMENDED that a
   sender pace sending of all retransmittable packets based on input
   from the congestion controller.  For example, a pacer might
   distribute the congestion window over the SRTT 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.

   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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4.7.  Pseudocode

4.7.1.  Constants of interest

   Constants used in congestion control are based on a combination of
   RFCs, papers, and common practice.  Some may need to be changed or
   negotiated in order to better suit a variety of environments.

   kDefaultMss (default 1460 bytes):  The default max packet size used
      for calculating default and minimum congestion windows.

   kInitialWindow (default 10 * kDefaultMss):  Default limit on the
      amount of outstanding data in bytes.

   kMinimumWindow (default 2 * kDefaultMss):  Default minimum congestion
      window.

   kLossReductionFactor (default 0.5):  Reduction in congestion window
      when a new loss event is detected.

4.7.2.  Variables of interest

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

   bytes_in_flight:  The sum of the size in bytes of all sent packets
      that contain at least one retransmittable frame, and have not been
      acked or declared lost.  The size does not include IP or UDP
      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.

   end_of_recovery:  The largest packet number sent when QUIC detects a
      loss.  When a larger packet is acknowledged, QUIC exits 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.

4.7.3.  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
      end_of_recovery = 0
      ssthresh = infinite

4.7.4.  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

4.7.5.  On Packet Acknowledgement

   Invoked from loss detection's OnPacketAcked and is supplied with
   acked_packet from sent_packets.

      InRecovery(packet_number)
        return packet_number <= end_of_recovery

      OnPacketAckedCC(acked_packet):
        // Remove from bytes_in_flight.
        bytes_in_flight -= acked_packet.bytes
        if (InRecovery(acked_packet.packet_number)):
          // Do not increase congestion window in recovery period.
          return
        if (congestion_window < ssthresh):
          // Slow start.
          congestion_window += acked_packet.bytes
        else:
          // Congestion avoidance.
          congestion_window +=
            kDefaultMss * acked_packet.bytes / congestion_window

4.7.6.  On Packets Lost

   Invoked by loss detection from DetectLostPackets when new packets are
   detected lost.












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      OnPacketsLost(lost_packets):
        // Remove lost packets from bytes_in_flight.
        for (lost_packet : lost_packets):
          bytes_in_flight -= lost_packet.bytes
        largest_lost_packet = lost_packets.last()
        // Start a new recovery epoch if the lost packet is larger
        // than the end of the previous recovery epoch.
        if (!InRecovery(largest_lost_packet.packet_number)):
          end_of_recovery = largest_sent_packet
          congestion_window *= kLossReductionFactor
          congestion_window = max(congestion_window, kMinimumWindow)
          ssthresh = congestion_window

4.7.7.  On Retransmission Timeout Verified

   QUIC decreases the congestion window to the minimum value once the
   retransmission timeout has been verified.

      OnRetransmissionTimeoutVerified()
        congestion_window = kMinimumWindow

5.  IANA Considerations

   This document has no IANA actions.  Yet.

6.  References

6.1.  Normative References

   [QUIC-TRANSPORT]
              Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", draft-ietf-quic-
              transport-12 (work in progress), May 2018.

   [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>.

   [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>.

6.2.  Informative References

   [RFC3465]  Allman, M., "TCP Congestion Control with Appropriate Byte
              Counting (ABC)", RFC 3465, DOI 10.17487/RFC3465, February
              2003, <https://www.rfc-editor.org/info/rfc3465>.



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   [RFC4653]  Bhandarkar, S., Reddy, A., 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
              Based on Selective Acknowledgment (SACK) for TCP",
              RFC 6675, DOI 10.17487/RFC6675, August 2012,
              <https://www.rfc-editor.org/info/rfc6675>.

   [TLP]      Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
              "Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
              Tail Losses", draft-dukkipati-tcpm-tcp-loss-probe-01 (work
              in progress), February 2013.

6.3.  URIs

   [1] https://mailarchive.ietf.org/arch/search/?email_list=quic

   [2] https://github.com/quicwg




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   [3] https://github.com/quicwg/base-drafts/labels/-recovery

Appendix A.  Acknowledgments

Appendix B.  Change Log

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

B.1.  Since draft-ietf-quic-recovery-10

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

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

   o  Define time_of_last_sent_handshake_packet (#1171)

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

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

B.2.  Since draft-ietf-quic-recovery-09

   No significant changes.

B.3.  Since draft-ietf-quic-recovery-08

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

B.4.  Since draft-ietf-quic-recovery-07

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

   o  Ack Delay in SRTT computation (#961)

   o  Default RTT and Slow Start (#590)

   o  Many editorial fixes.

B.5.  Since draft-ietf-quic-recovery-06

   No significant changes.







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B.6.  Since draft-ietf-quic-recovery-05

   o  Add more congestion control text (#776)

B.7.  Since draft-ietf-quic-recovery-04

   No significant changes.

B.8.  Since draft-ietf-quic-recovery-03

   No significant changes.

B.9.  Since draft-ietf-quic-recovery-02

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

   o  Add congestion control (#545, #395)

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

   o  Simplify RTO calculations (#142, #417)

B.10.  Since draft-ietf-quic-recovery-01

   o  Overview added to loss detection

   o  Changes initial default RTT to 100ms

   o  Added time-based loss detection and fixes early retransmit

   o  Clarified loss recovery for handshake packets

   o  Fixed references and made TCP references informative

B.11.  Since draft-ietf-quic-recovery-00

   o  Improved description of constants and ACK behavior

B.12.  Since draft-iyengar-quic-loss-recovery-01

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

   o  Updated authors/editors list

   o  Added table of contents





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Authors' Addresses

   Jana Iyengar (editor)
   Fastly

   Email: jri.ietf@gmail.com


   Ian Swett (editor)
   Google

   Email: ianswett@google.com







































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