QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                           I. Swett, Ed.
Expires: April 6, 26, 2019                                           Google
                                                        October 03, 23, 2018

               QUIC Loss Detection and Congestion Control


   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 April 6, 26, 2019.

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   Copyright (c) 2018 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Conventions and Definitions . . . . . . . . . . . . . . . . .   4
   3.  Design of the QUIC Transmission Machinery . . . . . . . . . .   4
     3.1.  Relevant Differences Between QUIC and TCP . . . . . . . .   5
       3.1.1.  Separate Packet Number Spaces . . . . . . . . . . . .   5
       3.1.2.  Monotonically Increasing Packet Numbers . . . . . . .   5   6
       3.1.3.  No Reneging . . . . . . . . . . . . . . . . . . . . .   6
       3.1.4.  More ACK Ranges . . . . . . . . . . . . . . . . . . .   6
       3.1.5.  Explicit Correction For Delayed ACKs  . . . . . . . .   6
   4.  Loss Detection  . . . . . . . . . . . . . . . . . . . . . . .   7
     4.1.  Computing the RTT estimate  . . . . . . . . . . . . . . .   7
     4.2.  Ack-based Detection . . . . . . . . . . . . . . . . . . .   7
       4.2.1.  Fast Retransmit . . . . . . . . . . . . . . . . . . .   7
       4.2.2.  Early Retransmit  . . . . . . . . . . . . . . . . . .   8
     4.3.  Timer-based Detection . . . . . . . . . . . . . . . . . .   9
       4.3.1.  Crypto Handshake Retransmission Timeout . . . . . . . . . . . . . .   9
       4.3.2.  Tail Loss Probe . . . . . . . . . . . . . . . . . . .  10
       4.3.3.  Retransmission Timeout  . . . . . . . . . . . . . . .  11
     4.4.  Generating Acknowledgements . . . . . . . . . . . . . . .  12
       4.4.1.  Crypto Handshake Data . . . . . . . . . . . . . . . .  12  13
       4.4.2.  ACK Ranges  . . . . . . . . . . . . . . . . . . . . .  13
       4.4.3.  Receiver Tracking of ACK Frames . . . . . . . . . . .  13
     4.5.  Pseudocode  . . . . . . . . . . . . . . . . . . . . . . .  13  14
       4.5.1.  Constants of interest . . . . . . . . . . . . . . . .  13  14
       4.5.2.  Variables of interest . . . . . . . . . . . . . . . .  14
       4.5.3.  Initialization  . . . . . . . . . . . . . . . . . . .  15  16
       4.5.4.  On Sending a Packet . . . . . . . . . . . . . . . . .  16
       4.5.5.  On Receiving an Acknowledgment  . . . . . . . . . . .  17
       4.5.6.  On Packet Acknowledgment  . . . . . . . . . . . . . .  19
       4.5.7.  Setting the Loss Detection Timer  . . . . . . . . . .  19
       4.5.8.  On Timeout  . . . . . . . . . . . . . . . . . . . . .  21  20
       4.5.9.  Detecting Lost Packets  . . . . . . . . . . . . . . .  22  21
     4.6.  Discussion  . . . . . . . . . . . . . . . . . . . . . . .  23  22
   5.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .  23  22
     5.1.  Explicit Congestion Notification  . . . . . . . . . . . .  24  23
     5.2.  Slow Start  . . . . . . . . . . . . . . . . . . . . . . .  24  23
     5.3.  Congestion Avoidance  . . . . . . . . . . . . . . . . . .  24  23
     5.4.  Recovery Period . . . . . . . . . . . . . . . . . . . . .  24  23
     5.5.  Tail Loss Probe . . . . . . . . . . . . . . . . . . . . .  25  24
     5.6.  Retransmission Timeout  . . . . . . . . . . . . . . . . .  25  24
     5.7.  Pacing  . . . . . . . . . . . . . . . . . . . . . . . . .  25  24
     5.8.  Pseudocode  . . . . . . . . . . . . . . . . . . . . . . .  26  25
       5.8.1.  Constants of interest . . . . . . . . . . . . . . . .  26  25
       5.8.2.  Variables of interest . . . . . . . . . . . . . . . .  26  25
       5.8.3.  Initialization  . . . . . . . . . . . . . . . . . . .  27  26
       5.8.4.  On Packet Sent  . . . . . . . . . . . . . . . . . . .  27  26
       5.8.5.  On Packet Acknowledgement . . . . . . . . . . . . . .  27  26
       5.8.6.  On New Congestion Event . . . . . . . . . . . . . . .  28  27
       5.8.7.  Process ECN Information . . . . . . . . . . . . . . .  28  27
       5.8.8.  On Packets Lost . . . . . . . . . . . . . . . . . . .  28  27
       5.8.9.  On Retransmission Timeout Verified  . . . . . . . . .  29  28
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  29  28
     6.1.  Congestion Signals  . . . . . . . . . . . . . . . . . . .  29  28
     6.2.  Traffic Analysis  . . . . . . . . . . . . . . . . . . . .  29  28
     6.3.  Misreporting ECN Markings . . . . . . . . . . . . . . . .  29  28
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  30  29
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  30  29
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  30  29
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  30  29
     8.3.  URIs  . . . . . . . . . . . . . . . . . . . . . . . . . .  31  30
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  32  31
     A.1.  Since draft-ietf-quic-recovery-14 . . . . . . . . . . . .  32  31
     A.2.  Since draft-ietf-quic-recovery-13 . . . . . . . . . . . .  32  31
     A.3.  Since draft-ietf-quic-recovery-12 . . . . . . . . . . . .  32  31
     A.4.  Since draft-ietf-quic-recovery-11 . . . . . . . . . . . .  32  31
     A.5.  Since draft-ietf-quic-recovery-10 . . . . . . . . . . . .  32  31
     A.6.  Since draft-ietf-quic-recovery-09 . . . . . . . . . . . .  33  32
     A.7.  Since draft-ietf-quic-recovery-08 . . . . . . . . . . . .  33  32
     A.8.  Since draft-ietf-quic-recovery-07 . . . . . . . . . . . .  33  32
     A.9.  Since draft-ietf-quic-recovery-06 . . . . . . . . . . . .  33  32
     A.10. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . .  33  32
     A.11. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . .  33  32
     A.12. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . .  33  32
     A.13. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . .  33  32
     A.14. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . .  34  33
     A.15. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . .  34  33
     A.16. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . .  34  33
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . .  34  33
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  34  33

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.

2.  Conventions and Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "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-only:  Any packet containing only an ACK frame.

   In-flight:  Packets are considered in-flight when they have been sent
      and neither acknowledged nor declared lost, and they are not ACK-

   Retransmittable Frames:  All frames besides ACK or PADDING are
      considered retransmittable.

   Retransmittable Packets:  Packets that contain retransmittable frames
      elicit an ACK from the receiver and are called retransmittable

   Crypto Packets:  Packets containing CRYPTO data sent in Initial or
      Handshake packets.

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 monotonically increase
   within a space, preventing ambiguity.

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

   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:

   o  All packets are acknowledged, though packets that contain only ACK
      and PADDING frames are not acknowledged immediately.

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

   o  Packets that contain only ACK frames do not count toward
      congestion control limits and are not considered in-flight.  Note
      that this means PADDING frames cause packets to contribute toward
      bytes in flight without directly causing an acknowledgment to be

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.

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 RTT measurement are unified across packet
   number spaces.

3.1.2.  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 application data is sent in one or more streams, with
   delivery order determined by stream offsets encoded within STREAM

   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.

3.1.3.  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

3.1.4.  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

3.1.5.  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
   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.

4.  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.

4.1.  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

   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.

4.2.  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.

4.2.1.  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) and/or a threshold amount of time after the
   unacknowledged packet.  Receipt of the acknowledgement indicates that
   a later packet was received, while the reordering threshold provides
   some tolerance for reordering of packets in the network.

   The RECOMMENDED initial value for kReorderingThreshold is 3, based on
   TCP loss recovery [RFC5681] [RFC6675].  Some networks may exhibit
   higher degrees of reordering, causing a sender to detect spurious
   losses.  Spuriously declaring packets lost 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.

   QUIC implementations can use time-based loss detection to handle
   reordering based on time elapsed since the packet was sent.  This may
   be used either as a replacement for a packet reordering threshold or
   in addition to it.  The RECOMMENDED time threshold, expressed as a
   fraction of the round-trip time (kTimeReorderingFraction), is 1/8.

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

   An endpoint SHOULD set the timer such that a packet is marked as lost
   no earlier than 1.125 * 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 1.125 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 multiplier increases loss recovery

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

4.3.  Timer-based Detection

   Timer-based loss detection recovers from losses that cannot be
   handled by ack-based loss detection.  It uses a single timer which
   switches between a handshake crypto retransmission timer, a Tail Loss Probe
   timer and Retransmission Timeout mechanisms.

4.3.1.  Crypto Handshake Retransmission Timeout

   Data in CRYPTO frames is critical to QUIC transport and crypto
   negotiation, so a more aggressive timeout is used to retransmit it.
   Below, the term "handshake packet" is used to refer to packets
   containing CRYPTO frames, not packets with the specific long header
   packet type Handshake.

   The initial handshake crypto retransmission 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 CRYPTO frames an
   acknowledgement is received, a new RTT is computed and the timer
   SHOULD be set for twice the newly computed smoothed RTT.

   When crypto packets are sent, the sender SHOULD MUST set a timer for the
   crypto timeout period.  Upon timeout, the sender MUST retransmit all
   unacknowledged CRYPTO data by calling
   RetransmitAllUnackedHandshakeData(). if possible.

   Until the server has validated the client's address on the path, the
   number of bytes it can send is limited, as specified in
   [QUIC-TRANSPORT].  If not all unacknowledged CRYPTO data can be sent,
   then all unacknowledged CRYPTO data sent in Initial packets should be
   retransmitted.  If no bytes can be sent, then no alarm should be
   armed until bytes have been received from the client.

   Because the server could be blocked until more packets are received,
   the client MUST start the crypto retransmission timer even if there
   is no unacknowledged CRYPTO data.  If the timer expires and the
   client has no CRYPTO data to retransmit and does not have Handshake
   keys, it SHOULD send an Initial packet in a UDP datagram of at least
   1200 octets.  If the client has Handshake keys, it SHOULD send a
   Handshake packet.

   On each consecutive expiration of the handshake crypto timer without receiving
   an acknowledgement for a new packet, the sender SHOULD double the handshake
   crypto retransmission timeout and set a timer for this period.

   When CRYPTO frames crypto packets are outstanding, the TLP and RTO timers are not
   active unless the CRYPTO frames were sent at 1-RTT encryption.

   When an acknowledgement is received for a handshake packet, the new
   RTT is computed and the timer SHOULD be set for twice the newly
   computed smoothed RTT.
   active.  Retry and Version Negotiation

   A Retry or Version Negotiation packet causes a client to send another
   Initial packet, effectively restarting the connection process.

   Either packet indicates that the Initial was received but not
   processed.  Neither packet can be treated as an acknowledgment for
   the Initial, but they MAY be used to improve the RTT estimate.

4.3.2.  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 a timer when the last retransmittable packet
   before quiescence is transmitted.  Upon timeout, a Tail Loss Probe
   (TLP) packet is sent to evoke an acknowledgement from the receiver.

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

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

   o  If RTO (Section 4.3.3) is earlier, schedule a TLP in its place.
      That is, PTO SHOULD be scheduled for min(RTO, PTO).

   QUIC includes MaxAckDelay in all probe timeouts, because it assumes
   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 timer to fire twice before setting an RTO timer.  In other
   words, when the TLP timer expires the first time, a TLP packet is
   sent, and it is RECOMMENDED that the TLP timer be scheduled for a
   second time.  When the TLP timer expires the second time, a second
   TLP packet is sent, and an RTO timer SHOULD be scheduled
   Section 4.3.3.

   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
   recovery time.  Since a TLP timer 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 timer expires.

   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 timer on
   every sent retransmittable packet.

4.3.3.  Retransmission Timeout

   A Retransmission Timeout (RTO) timer 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, a timer is set for the RTO period.
   When this timer expires, the sender sends two packets, to evoke
   acknowledgements from the receiver, and restarts the RTO timer.

   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 timer expires, the RTO period is doubled.

   The sender typically has incurred a high latency penalty by the time
   an RTO timer expires, 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
   timer 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 timer expires 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.  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.

   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.  If an acknowledgement for a
   packet sent on an RTO is received at the same time packets sent prior
   to the first RTO are acknowledged, the RTO is considered spurious and
   standard loss detection rules apply.

   A packet sent when an RTO timer expires 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 timer 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.

4.4.  Generating Acknowledgements

   QUIC SHOULD delay sending acknowledgements in response to packets,
   but MUST NOT excessively delay acknowledgements of packets containing
   frames other than ACK.  Specifically, implementations MUST attempt to
   enforce a maximum ack delay to avoid causing the peer spurious
   timeouts.  The maximum ack delay is communicated in the
   "max_ack_delay" transport parameter and the default value is 25ms.

   An acknowledgement SHOULD be sent immediately upon receipt of a
   second packet but the delay SHOULD NOT exceed the maximum ack delay.
   QUIC recovery algorithms do not assume the peer generates an
   acknowledgement immediately when receiving a second full-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.

   Similarly, packets marked with the ECN Congestion Experienced (CE)
   codepoint in the IP header SHOULD be acknowledged immediately, to
   reduce the peer's response time to congestion events.

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

4.4.1.  Crypto Handshake Data

   In order to quickly complete the handshake and avoid spurious
   retransmissions due to handshake crypto retransmission timeouts, handshake crypto packets
   SHOULD use a very short ack delay, such as 1ms.  ACK frames MAY be
   sent immediately when the crypto stack indicates all data for that
   encryption level has been received.

4.4.2.  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
   important it is to send an updated ACK frame quickly, to prevent the
   peer from declaring a packet as lost and spuriously retransmitting
   the frames it contains.

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

4.4.3.  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

4.5.  Pseudocode

4.5.1.  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:  Maximum number of tail loss probes before an RTO expires.
      The RECOMMENDED value is 2.

   kReorderingThreshold:  Maximum reordering in packet number space
      before FACK style loss detection considers a packet lost.  The
      RECOMMENDED value is 3.

   kTimeReorderingFraction:  Maximum reordering in time space before
      time based loss detection considers a packet lost.  In fraction of
      an RTT.  The RECOMMENDED value is 1/8.

   kUsingTimeLossDetection:  Whether time based loss detection is in
      use.  If false, uses FACK style loss detection.  The RECOMMENDED
      value is false.

   kMinTLPTimeout:  Minimum time in the future a tail loss probe timer
      may be set for.  The RECOMMENDED value is 10ms.

   kMinRTOTimeout:  Minimum time in the future an RTO timer may be set
      for.  The RECOMMENDED value is 200ms.

   kDelayedAckTimeout:  The length of the peer's delayed ack timer.  The
      RECOMMENDED value is 25ms.

   kInitialRtt:  The RTT used before an RTT sample is taken.  The
      RECOMMENDED value is 100ms.

4.5.2.  Variables of interest

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

   loss_detection_timer:  Multi-modal timer used for loss detection.


   crypto_count:  The number of times all unacknowledged handshake CRYPTO 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_crypto_packet:  The time the most recent crypto
      containing a CRYPTO frame was sent.

   largest_sent_packet:  The packet number of the most recently sent

   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]

   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, in milliseconds.  The actual
      ack_delay in a received ACK frame may be larger due to late
      timers, reordering, or lost ACKs.

   reordering_threshold:  The largest packet number gap between the
      largest acknowledged 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

   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, a boolean indicating
      whether it counts towards bytes in flight, 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.  A sent_packets data structure is maintained per packet
      number space, and ACK processing only applies to a single space.

4.5.3.  Initialization

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

      crypto_count = 0
      tlp_count = 0
      rto_count = 0
      if (kUsingTimeLossDetection)
        reordering_threshold = infinite
        time_reordering_fraction = kTimeReorderingFraction
        reordering_threshold = kReorderingThreshold
        time_reordering_fraction = infinite
      loss_time = 0
      smoothed_rtt = 0
      rttvar = 0
      min_rtt = infinite
      largest_sent_before_rto = 0
      time_of_last_sent_retransmittable_packet = 0
      time_of_last_sent_crypto_packet = 0
      largest_sent_packet = 0

4.5.4.  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  ack_only: A boolean that indicates whether a packet contains only
      ACK or PADDING frame(s).  If true, it is still expected an ack
      will be received for this packet, but it is not retransmittable.

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

   o  is_handshake_packet:  is_crypto_packet: A boolean that indicates whether the packet
      contains cryptographic handshake messages critical to the
      completion of the QUIC handshake.  In this version of QUIC, this
      includes any packet with the long header that includes a CRYPTO

   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:

    OnPacketSent(packet_number, ack_only, in_flight,
                 is_crypto_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 = ack_only
      sent_packets[packet_number].in_flight = in_flight
      if !ack_only:
        if is_handshake_packet:
          time_of_last_sent_handshake_packet is_crypto_packet:
          time_of_last_sent_crypto_packet = now
        time_of_last_sent_retransmittable_packet = now
        sent_packets[packet_number].bytes = sent_bytes

4.5.5.  On Receiving an Acknowledgment

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

   Pseudocode for OnAckReceived and UpdateRtt follow:

       largest_acked_packet = ack.largest_acked
       // If the largest acknowledged 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 in this ACK frame
       newly_acked_packets = DetermineNewlyAckedPackets(ack)
       for acked_packet in newly_acked_packets:

       if !newly_acked_packets.empty():
         // Find the smallest newly acknowledged packet
         smallest_newly_acked =
         // If any packets sent prior to RTO were acked, then the
         // RTO was spurious. Otherwise, inform congestion control.
         if (rto_count > 0 &&
               smallest_newly_acked > largest_sent_before_rto):
         crypto_count = 0
         tlp_count = 0
         rto_count = 0


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

     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
       // Based on {{RFC6298}}.
       if (smoothed_rtt == 0):
         smoothed_rtt = latest_rtt
         rttvar = latest_rtt / 2
         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

4.5.6.  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

   Pseudocode for OnPacketAcked follows:

        if (!acked_packet.is_ack_only):

4.5.7.  Setting the Loss Detection Timer

   QUIC loss detection uses a single timer for all timer-based 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.  Handshake Timer

   When a connection has unacknowledged handshake data, the handshake
   timer 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 0-RTT packets.  Tail Loss Probe and Retransmission Timer

   Tail loss probes [TLP] and retransmission timeouts [RFC6298] are
   timer based mechanisms 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 no unacknowledged
   handshake data.  The TLP timer is set until the max number of TLP
   packets have been sent, and then the RTO timer is set.  Early Retransmit Timer

   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.  Pseudocode

   Pseudocode for SetLossDetectionTimer follows:

       // Don't arm timer if there are no retransmittable packets
       // in flight.
       if (bytes_in_flight == 0):

       if (handshake (crypto packets are outstanding):
         // Handshake Crypto retransmission timer.
         if (smoothed_rtt == 0):
           timeout = 2 * kInitialRtt
           timeout = 2 * smoothed_rtt
         timeout = max(timeout, kMinTLPTimeout)
         timeout = timeout * (2 ^ handshake_count) crypto_count)
           time_of_last_sent_crypto_packet + timeout)
       if (loss_time != 0):
         // Early retransmit timer or time loss detection.
         timeout = loss_time -
         // RTO or TLP timer
         // Calculate RTO duration
         timeout =
           smoothed_rtt + 4 * rttvar + max_ack_delay
         timeout = max(timeout, kMinRTOTimeout)
         timeout = timeout * (2 ^ rto_count)
         if (tlp_count < kMaxTLPs):
           // Tail Loss Probe
           tlp_timeout = max(1.5 * smoothed_rtt
                              + max_ack_delay, kMinTLPTimeout)
           timeout = min(tlp_timeout, timeout)

         time_of_last_sent_retransmittable_packet + timeout)

4.5.8.  On Timeout

   QUIC uses one loss recovery timer, which when set, can be in one of
   several modes.

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

   Pseudocode for OnLossDetectionTimeout follows:

        if (handshake (crypto packets are outstanding):
          // Handshake Crypto retransmission timeout.
        else if (loss_time != 0):
          // Early retransmit or Time Loss Detection
        else if (tlp_count < kMaxTLPs):
          // Tail Loss Probe.
          // RTO.
          if (rto_count == 0)
            largest_sent_before_rto = largest_sent_packet


4.5.9.  Detecting Lost Packets

   Packets in QUIC are only considered lost once a larger packet number
   in the same packet number space is acknowledged.  DetectLostPackets
   is called every time an ack is received and operates on the
   sent_packets for that packet number space.  If the loss detection
   timer expires and the loss_time is set, the previous largest acked
   packet is supplied.  Pseudocode

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

   Pseudocode for DetectLostPackets follows:

     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 timer.
       delay_until_lost = 9/8 * 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):
         if (!unacked.is_ack_only):
       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()):

4.6.  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.

5.  Congestion Control

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

   QUIC hosts MUST NOT send packets if they would increase
   bytes_in_flight (defined in Section 5.8.2) beyond the available
   congestion window, unless the packet is a probe packet sent after the
   TLP or RTO timer expires, as described in Section 4.3.2 and
   Section 4.3.3.

   Implementations MAY use other congestion control algorithms, and
   endpoints MAY use different algorithms from one another.  The signals
   QUIC provides for congestion control are generic and are designed to
   support different algorithms.

5.1.  Explicit Congestion Notification

   If a path has been verified to support ECN, QUIC treats a Congestion
   Experienced codepoint in the IP header as a signal of congestion.
   This document specifies an endpoint's response when its peer receives
   packets with the Congestion Experienced codepoint.  As discussed in
   [RFC8311], endpoints are permitted to experiment with other response

5.2.  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 anytime the congestion window is less than ssthresh, which
   typically only occurs after an RTO.  While in slow start, QUIC
   increases the congestion window by the number of bytes acknowledged
   when each ack is processed.

5.3.  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.

5.4.  Recovery Period

   Recovery is a period of time beginning with detection of a lost
   packet or an increase in the ECN-CE counter.  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, which ends when the lost packet that started recovery is

   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.

5.5.  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.

5.6.  Retransmission Timeout

   When retransmissions are sent due to a retransmission timeout timer,
   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.

5.7.  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 SRTT when used with a window-based
   controller, and a pacer might use the rate estimate of a rate-based

   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).

5.8.  Pseudocode

5.8.1.  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.

   kMaxDatagramSize:  The sender's maximum payload size.  Does not
      include UDP or IP overhead.  The max packet size is used for
      calculating initial and minimum congestion windows.  The
      RECOMMENDED value is 1200 bytes.

   kInitialWindow:  Default limit on the initial amount of outstanding
      data in bytes.  Taken from [RFC6928].  The RECOMMENDED value is
      the minimum of 10 * kMaxDatagramSize and max(2* kMaxDatagramSize,

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

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

5.8.2.  Variables of interest

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

   ecn_ce_counter:  The highest value reported for the ECN-CE counter by
      the peer in an ACK frame.  This variable 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 retransmittable 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

   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.

5.8.3.  Initialization

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

      congestion_window = kInitialWindow
      bytes_in_flight = 0
      end_of_recovery = 0
      ssthresh = infinite
      ecn_ce_counter = 0

5.8.4.  On Packet Sent

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

        bytes_in_flight += bytes_sent

5.8.5.  On Packet Acknowledgement

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

        return packet_number <= end_of_recovery

        // 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.
        if (congestion_window < ssthresh):
          // Slow start.
          congestion_window += acked_packet.bytes
          // Congestion avoidance.
          congestion_window += kMaxDatagramSize * acked_packet.bytes
              / congestion_window

5.8.6.  On New Congestion Event

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

        // Start a new congestion event if packet_number
        // is larger than the end of the previous recovery epoch.
        if (!InRecovery(packet_number)):
          end_of_recovery = largest_sent_packet
          congestion_window *= kLossReductionFactor
          congestion_window = max(congestion_window, kMinimumWindow)
          ssthresh = congestion_window

5.8.7.  Process ECN Information

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

        // If the ECN-CE counter reported by the peer has increased,
        // this could be a new congestion event.
        if (ack.ce_counter > ecn_ce_counter):
          ecn_ce_counter = ack.ce_counter
          // Start a new congestion event if the last acknowledged
          // packet is past the end of the previous recovery epoch.

5.8.8.  On Packets Lost

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

        // 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 congestion epoch if the last lost packet
        // is past the end of the previous recovery epoch.

5.8.9.  On Retransmission Timeout Verified

   QUIC decreases the congestion window to the minimum value once the
   retransmission timeout has been verified and removes any packets sent
   before the newly acknowledged RTO packet.

        congestion_window = kMinimumWindow
        // Declare all packets prior to packet_number lost.
        for (sent_packet: sent_packets):
          if (sent_packet.packet_number < packet_number):
            bytes_in_flight -= lost_packet.bytes sent_packet.bytes

6.  Security Considerations

6.1.  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.

6.2.  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.

6.3.  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 marked
   with ECN-CE is not reported as having been marked when the packet is
   acknowledged, the sender SHOULD then disable ECN for that path.

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

   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.

7.  IANA Considerations

   This document has no IANA actions.  Yet.

8.  References

8.1.  Normative References

              Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", draft-ietf-quic-
              transport-16 (work in progress), October 2018.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

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

   [RFC8311]  Black, D., "Relaxing Restrictions on Explicit Congestion
              Notification (ECN) Experimentation", RFC 8311,
              DOI 10.17487/RFC8311, January 2018,

8.2.  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>.

   [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,

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,

   [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,

   [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,

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,

   [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,

   [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,

   [RFC6928]  Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
              "Increasing TCP's Initial Window", RFC 6928,
              DOI 10.17487/RFC6928, April 2013,

   [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.

8.3.  URIs

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

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

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

Appendix A.  Change Log

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

A.1.  Since draft-ietf-quic-recovery-14

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

   o  Merge ACK and ACK_ECN (#1783)

A.2.  Since draft-ietf-quic-recovery-13

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

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

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

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

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

A.3.  Since draft-ietf-quic-recovery-12

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

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

A.4.  Since draft-ietf-quic-recovery-11

   No significant changes.

A.5.  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

A.6.  Since draft-ietf-quic-recovery-09

   No significant changes.

A.7.  Since draft-ietf-quic-recovery-08

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

A.8.  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.

A.9.  Since draft-ietf-quic-recovery-06

   No significant changes.

A.10.  Since draft-ietf-quic-recovery-05

   o  Add more congestion control text (#776)

A.11.  Since draft-ietf-quic-recovery-04

   No significant changes.

A.12.  Since draft-ietf-quic-recovery-03

   No significant changes.

A.13.  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

   o  Simplify RTO calculations (#142, #417)

A.14.  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

A.15.  Since draft-ietf-quic-recovery-00

   o  Improved description of constants and ACK behavior

A.16.  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


Authors' Addresses

   Jana Iyengar (editor)

   Email: jri.ietf@gmail.com

   Ian Swett (editor)

   Email: ianswett@google.com