draft-ietf-quic-recovery-29.txt   draft-ietf-quic-recovery-30.txt 
QUIC J. Iyengar, Ed. QUIC J. Iyengar, Ed.
Internet-Draft Fastly Internet-Draft Fastly
Intended status: Standards Track I. Swett, Ed. Intended status: Standards Track I. Swett, Ed.
Expires: 12 December 2020 Google Expires: March 14, 2021 Google
10 June 2020 September 10, 2020
QUIC Loss Detection and Congestion Control QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-29 draft-ietf-quic-recovery-30
Abstract Abstract
This document describes loss detection and congestion control This document describes loss detection and congestion control
mechanisms for QUIC. mechanisms for QUIC.
Note to Readers Note to Readers
Discussion of this draft takes place on the QUIC working group Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org (mailto:quic@ietf.org)), which is mailing list (quic@ietf.org (mailto:quic@ietf.org)), which is
skipping to change at page 1, line 43 skipping to change at page 1, line 43
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/. Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on 12 December 2020. This Internet-Draft will expire on March 14, 2021.
Copyright Notice Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/ Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document. license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights Please review these documents carefully, as they describe your rights
skipping to change at page 2, line 31 skipping to change at page 2, line 31
4.2. Monotonically Increasing Packet Numbers . . . . . . . . . 6 4.2. Monotonically Increasing Packet Numbers . . . . . . . . . 6
4.3. Clearer Loss Epoch . . . . . . . . . . . . . . . . . . . 6 4.3. Clearer Loss Epoch . . . . . . . . . . . . . . . . . . . 6
4.4. No Reneging . . . . . . . . . . . . . . . . . . . . . . . 7 4.4. No Reneging . . . . . . . . . . . . . . . . . . . . . . . 7
4.5. More ACK Ranges . . . . . . . . . . . . . . . . . . . . . 7 4.5. More ACK Ranges . . . . . . . . . . . . . . . . . . . . . 7
4.6. Explicit Correction For Delayed Acknowledgements . . . . 7 4.6. Explicit Correction For Delayed Acknowledgements . . . . 7
4.7. Probe Timeout Replaces RTO and TLP . . . . . . . . . . . 7 4.7. Probe Timeout Replaces RTO and TLP . . . . . . . . . . . 7
4.8. The Minimum Congestion Window is Two Packets . . . . . . 8 4.8. The Minimum Congestion Window is Two Packets . . . . . . 8
5. Estimating the Round-Trip Time . . . . . . . . . . . . . . . 8 5. Estimating the Round-Trip Time . . . . . . . . . . . . . . . 8
5.1. Generating RTT samples . . . . . . . . . . . . . . . . . 8 5.1. Generating RTT samples . . . . . . . . . . . . . . . . . 8
5.2. Estimating min_rtt . . . . . . . . . . . . . . . . . . . 9 5.2. Estimating min_rtt . . . . . . . . . . . . . . . . . . . 9
5.3. Estimating smoothed_rtt and rttvar . . . . . . . . . . . 9 5.3. Estimating smoothed_rtt and rttvar . . . . . . . . . . . 10
6. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 11 6. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 12
6.1. Acknowledgement-based Detection . . . . . . . . . . . . . 11 6.1. Acknowledgement-Based Detection . . . . . . . . . . . . . 12
6.1.1. Packet Threshold . . . . . . . . . . . . . . . . . . 11 6.1.1. Packet Threshold . . . . . . . . . . . . . . . . . . 13
6.1.2. Time Threshold . . . . . . . . . . . . . . . . . . . 12 6.1.2. Time Threshold . . . . . . . . . . . . . . . . . . . 13
6.2. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 13 6.2. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 14
6.2.1. Computing PTO . . . . . . . . . . . . . . . . . . . . 13 6.2.1. Computing PTO . . . . . . . . . . . . . . . . . . . . 14
6.2.2. Handshakes and New Paths . . . . . . . . . . . . . . 14 6.2.2. Handshakes and New Paths . . . . . . . . . . . . . . 16
6.2.3. Speeding Up Handshake Completion . . . . . . . . . . 15 6.2.3. Speeding Up Handshake Completion . . . . . . . . . . 17
6.2.4. Sending Probe Packets . . . . . . . . . . . . . . . . 16 6.2.4. Sending Probe Packets . . . . . . . . . . . . . . . . 17
6.3. Handling Retry Packets . . . . . . . . . . . . . . . . . 17 6.3. Handling Retry Packets . . . . . . . . . . . . . . . . . 18
6.4. Discarding Keys and Packet State . . . . . . . . . . . . 17 6.4. Discarding Keys and Packet State . . . . . . . . . . . . 19
7. Congestion Control . . . . . . . . . . . . . . . . . . . . . 18 7. Congestion Control . . . . . . . . . . . . . . . . . . . . . 19
7.1. Explicit Congestion Notification . . . . . . . . . . . . 19 7.1. Explicit Congestion Notification . . . . . . . . . . . . 20
7.2. Initial and Minimum Congestion Window . . . . . . . . . . 19 7.2. Initial and Minimum Congestion Window . . . . . . . . . . 20
7.3. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 19 7.3. Congestion Control States . . . . . . . . . . . . . . . . 20
7.4. Congestion Avoidance . . . . . . . . . . . . . . . . . . 20 7.3.1. Slow Start . . . . . . . . . . . . . . . . . . . . . 21
7.5. Recovery Period . . . . . . . . . . . . . . . . . . . . . 20 7.3.2. Recovery . . . . . . . . . . . . . . . . . . . . . . 21
7.6. Ignoring Loss of Undecryptable Packets . . . . . . . . . 20 7.3.3. Congestion Avoidance . . . . . . . . . . . . . . . . 22
7.7. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 21 7.4. Ignoring Loss of Undecryptable Packets . . . . . . . . . 22
7.8. Persistent Congestion . . . . . . . . . . . . . . . . . . 21 7.5. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 23
7.9. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 22 7.6. Persistent Congestion . . . . . . . . . . . . . . . . . . 23
7.10. Under-utilizing the Congestion Window . . . . . . . . . . 23 7.6.1. Duration . . . . . . . . . . . . . . . . . . . . . . 23
8. Security Considerations . . . . . . . . . . . . . . . . . . . 24 7.6.2. Establishing Persistent Congestion . . . . . . . . . 24
8.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 24 7.6.3. Example . . . . . . . . . . . . . . . . . . . . . . . 24
8.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 24 7.7. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 25
8.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 24 7.8. Under-utilizing the Congestion Window . . . . . . . . . . 27
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25 8. Security Considerations . . . . . . . . . . . . . . . . . . . 27
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 25 8.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 27
10.1. Normative References . . . . . . . . . . . . . . . . . . 25 8.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 27
10.2. Informative References . . . . . . . . . . . . . . . . . 25 8.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 27
Appendix A. Loss Recovery Pseudocode . . . . . . . . . . . . . . 27 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 28
A.1. Tracking Sent Packets . . . . . . . . . . . . . . . . . . 27 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 28
A.1.1. Sent Packet Fields . . . . . . . . . . . . . . . . . 27 10.1. Normative References . . . . . . . . . . . . . . . . . . 28
A.2. Constants of Interest . . . . . . . . . . . . . . . . . . 28 10.2. Informative References . . . . . . . . . . . . . . . . . 29
A.3. Variables of interest . . . . . . . . . . . . . . . . . . 28 Appendix A. Loss Recovery Pseudocode . . . . . . . . . . . . . . 30
A.4. Initialization . . . . . . . . . . . . . . . . . . . . . 29 A.1. Tracking Sent Packets . . . . . . . . . . . . . . . . . . 30
A.5. On Sending a Packet . . . . . . . . . . . . . . . . . . . 29 A.1.1. Sent Packet Fields . . . . . . . . . . . . . . . . . 31
A.6. On Receiving a Datagram . . . . . . . . . . . . . . . . . 30 A.2. Constants of Interest . . . . . . . . . . . . . . . . . . 31
A.7. On Receiving an Acknowledgment . . . . . . . . . . . . . 30 A.3. Variables of interest . . . . . . . . . . . . . . . . . . 32
A.8. Setting the Loss Detection Timer . . . . . . . . . . . . 32 A.4. Initialization . . . . . . . . . . . . . . . . . . . . . 33
A.9. On Timeout . . . . . . . . . . . . . . . . . . . . . . . 33 A.5. On Sending a Packet . . . . . . . . . . . . . . . . . . . 33
A.10. Detecting Lost Packets . . . . . . . . . . . . . . . . . 34 A.6. On Receiving a Datagram . . . . . . . . . . . . . . . . . 33
Appendix B. Congestion Control Pseudocode . . . . . . . . . . . 35 A.7. On Receiving an Acknowledgment . . . . . . . . . . . . . 34
B.1. Constants of interest . . . . . . . . . . . . . . . . . . 35 A.8. Setting the Loss Detection Timer . . . . . . . . . . . . 35
B.2. Variables of interest . . . . . . . . . . . . . . . . . . 36 A.9. On Timeout . . . . . . . . . . . . . . . . . . . . . . . 37
B.3. Initialization . . . . . . . . . . . . . . . . . . . . . 36 A.10. Detecting Lost Packets . . . . . . . . . . . . . . . . . 38
B.4. On Packet Sent . . . . . . . . . . . . . . . . . . . . . 37 Appendix B. Congestion Control Pseudocode . . . . . . . . . . . 39
B.5. On Packet Acknowledgement . . . . . . . . . . . . . . . . 37 B.1. Constants of interest . . . . . . . . . . . . . . . . . . 39
B.6. On New Congestion Event . . . . . . . . . . . . . . . . . 37 B.2. Variables of interest . . . . . . . . . . . . . . . . . . 40
B.7. Process ECN Information . . . . . . . . . . . . . . . . . 38 B.3. Initialization . . . . . . . . . . . . . . . . . . . . . 40
B.8. On Packets Lost . . . . . . . . . . . . . . . . . . . . . 38 B.4. On Packet Sent . . . . . . . . . . . . . . . . . . . . . 41
B.9. Upon dropping Initial or Handshake keys . . . . . . . . . 39 B.5. On Packet Acknowledgement . . . . . . . . . . . . . . . . 41
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 39 B.6. On New Congestion Event . . . . . . . . . . . . . . . . . 42
C.1. Since draft-ietf-quic-recovery-28 . . . . . . . . . . . . 39 B.7. Process ECN Information . . . . . . . . . . . . . . . . . 43
C.2. Since draft-ietf-quic-recovery-27 . . . . . . . . . . . . 39 B.8. On Packets Lost . . . . . . . . . . . . . . . . . . . . . 43
C.3. Since draft-ietf-quic-recovery-26 . . . . . . . . . . . . 40 B.9. Upon dropping Initial or Handshake keys . . . . . . . . . 43
C.4. Since draft-ietf-quic-recovery-25 . . . . . . . . . . . . 40 Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 44
C.5. Since draft-ietf-quic-recovery-24 . . . . . . . . . . . . 40 C.1. Since draft-ietf-quic-recovery-29 . . . . . . . . . . . . 44
C.6. Since draft-ietf-quic-recovery-23 . . . . . . . . . . . . 40 C.2. Since draft-ietf-quic-recovery-28 . . . . . . . . . . . . 44
C.7. Since draft-ietf-quic-recovery-22 . . . . . . . . . . . . 40 C.3. Since draft-ietf-quic-recovery-27 . . . . . . . . . . . . 44
C.8. Since draft-ietf-quic-recovery-21 . . . . . . . . . . . . 40 C.4. Since draft-ietf-quic-recovery-26 . . . . . . . . . . . . 45
C.9. Since draft-ietf-quic-recovery-20 . . . . . . . . . . . . 40 C.5. Since draft-ietf-quic-recovery-25 . . . . . . . . . . . . 45
C.10. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . . 41 C.6. Since draft-ietf-quic-recovery-24 . . . . . . . . . . . . 45
C.11. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . . 41 C.7. Since draft-ietf-quic-recovery-23 . . . . . . . . . . . . 45
C.12. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . . 42 C.8. Since draft-ietf-quic-recovery-22 . . . . . . . . . . . . 46
C.13. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . . 42 C.9. Since draft-ietf-quic-recovery-21 . . . . . . . . . . . . 46
C.14. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 43 C.10. Since draft-ietf-quic-recovery-20 . . . . . . . . . . . . 46
C.15. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 43 C.11. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . . 46
C.16. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 43 C.12. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . . 46
C.17. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 43 C.13. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . . 47
C.18. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 43 C.14. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . . 47
C.19. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 44 C.15. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 48
C.20. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 44 C.16. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 48
C.21. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 44 C.17. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 48
C.22. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 44 C.18. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 49
C.23. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 44 C.19. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 49
C.24. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 44 C.20. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 49
C.25. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 44 C.21. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 49
C.26. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 44 C.22. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 49
C.27. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 45 C.23. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 49
C.28. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 45 C.24. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 49
C.29. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 45 C.25. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 50
Appendix D. Contributors . . . . . . . . . . . . . . . . . . . . 45 C.26. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 50
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 45 C.27. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 50
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45 C.28. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 50
C.29. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 50
C.30. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 50
Appendix D. Contributors . . . . . . . . . . . . . . . . . . . . 51
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 51
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 51
1. Introduction 1. Introduction
QUIC is a new multiplexed and secure transport protocol atop UDP, QUIC is a new multiplexed and secure transport protocol atop UDP,
specified in [QUIC-TRANSPORT]. This document describes congestion specified in [QUIC-TRANSPORT]. This document describes congestion
control and loss recovery for QUIC. Mechanisms described in this control and loss recovery for QUIC. Mechanisms described in this
document follow the spirit of existing TCP congestion control and document follow the spirit of existing TCP congestion control and
loss recovery mechanisms, described in RFCs, various Internet-drafts, loss recovery mechanisms, described in RFCs, various Internet-drafts,
or academic papers, and also those prevalent in TCP implementations. or academic papers, and also those prevalent in TCP implementations.
2. Conventions and Definitions 2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in "OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here. capitals, as shown here.
Definitions of terms that are used in this document: Definitions of terms that are used in this document:
Ack-eliciting Frames: All frames other than ACK, PADDING, and Ack-eliciting frames: All frames other than ACK, PADDING, and
CONNECTION_CLOSE are considered ack-eliciting. CONNECTION_CLOSE are considered ack-eliciting.
Ack-eliciting Packets: Packets that contain ack-eliciting frames Ack-eliciting packets: Packets that contain ack-eliciting frames
elicit an ACK from the receiver within the maximum ack delay and elicit an ACK from the receiver within the maximum acknowledgement
are called ack-eliciting packets. delay and are called ack-eliciting packets.
In-flight: Packets are considered in-flight when they are ack- In-flight: Packets are considered in-flight when they are ack-
eliciting or contain a PADDING frame, and they have been sent but eliciting or contain a PADDING frame, and they have been sent but
are not acknowledged, declared lost, or abandoned along with old are not acknowledged, declared lost, or discarded along with old
keys. keys.
3. Design of the QUIC Transmission Machinery 3. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which All transmissions in QUIC are sent with a packet-level header, which
indicates the encryption level and includes a packet sequence number indicates the encryption level and includes a packet sequence number
(referred to below as a packet number). The encryption level (referred to below as a packet number). The encryption level
indicates the packet number space, as described in [QUIC-TRANSPORT]. indicates the packet number space, as described in [QUIC-TRANSPORT].
Packet numbers never repeat within a packet number space for the Packet numbers never repeat within a packet number space for the
lifetime of a connection. Packet numbers are sent in monotonically lifetime of a connection. Packet numbers are sent in monotonically
increasing order within a space, preventing ambiguity. increasing order within a space, preventing ambiguity.
This design obviates the need for disambiguating between This design obviates the need for disambiguating between
transmissions and retransmissions and eliminates significant transmissions and retransmissions; this eliminates significant
complexity from QUIC's interpretation of TCP loss detection complexity from QUIC's interpretation of TCP loss detection
mechanisms. mechanisms.
QUIC packets can contain multiple frames of different types. The QUIC packets can contain multiple frames of different types. The
recovery mechanisms ensure that data and frames that need reliable recovery mechanisms ensure that data and frames that need reliable
delivery are acknowledged or declared lost and sent in new packets as delivery are acknowledged or declared lost and sent in new packets as
necessary. The types of frames contained in a packet affect recovery necessary. The types of frames contained in a packet affect recovery
and congestion control logic: and congestion control logic:
* All packets are acknowledged, though packets that contain no ack- * All packets are acknowledged, though packets that contain no ack-
skipping to change at page 5, line 45 skipping to change at page 5, line 50
count toward congestion control limits and are considered in- count toward congestion control limits and are considered in-
flight. flight.
* PADDING frames cause packets to contribute toward bytes in flight * PADDING frames cause packets to contribute toward bytes in flight
without directly causing an acknowledgment to be sent. without directly causing an acknowledgment to be sent.
4. Relevant Differences Between QUIC and TCP 4. Relevant Differences Between QUIC and TCP
Readers familiar with TCP's loss detection and congestion control Readers familiar with TCP's loss detection and congestion control
will find algorithms here that parallel well-known TCP ones. will find algorithms here that parallel well-known TCP ones.
Protocol differences between QUIC and TCP however contribute to However, protocol differences between QUIC and TCP contribute to
algorithmic differences. We briefly describe these protocol algorithmic differences. These protocol differences are briefly
differences below. described below.
4.1. Separate Packet Number Spaces 4.1. Separate Packet Number Spaces
QUIC uses separate packet number spaces for each encryption level, QUIC uses separate packet number spaces for each encryption level,
except 0-RTT and all generations of 1-RTT keys use the same packet except 0-RTT and all generations of 1-RTT keys use the same packet
number space. Separate packet number spaces ensures acknowledgement number space. Separate packet number spaces ensures acknowledgement
of packets sent with one level of encryption will not cause spurious of packets sent with one level of encryption will not cause spurious
retransmission of packets sent with a different encryption level. retransmission of packets sent with a different encryption level.
Congestion control and round-trip time (RTT) measurement are unified Congestion control and round-trip time (RTT) measurement are unified
across packet number spaces. across packet number spaces.
skipping to change at page 6, line 29 skipping to change at page 6, line 29
carrying the same sequence number, and consequently leads to carrying the same sequence number, and consequently leads to
"retransmission ambiguity". QUIC separates the two. QUIC uses a "retransmission ambiguity". QUIC separates the two. QUIC uses a
packet number to indicate transmission order. Application data is packet number to indicate transmission order. Application data is
sent in one or more streams and delivery order is determined by sent in one or more streams and delivery order is determined by
stream offsets encoded within STREAM frames. stream offsets encoded within STREAM frames.
QUIC's packet number is strictly increasing within a packet number QUIC's packet number is strictly increasing within a packet number
space, and directly encodes transmission order. A higher packet space, and directly encodes transmission order. A higher packet
number signifies that the packet was sent later, and a lower packet number signifies that the packet was sent later, and a lower packet
number signifies that the packet was sent earlier. When a packet number signifies that the packet was sent earlier. When a packet
containing ack-eliciting frames is detected lost, QUIC rebundles containing ack-eliciting frames is detected lost, QUIC includes
necessary frames in a new packet with a new packet number, removing necessary frames in a new packet with a new packet number, removing
ambiguity about which packet is acknowledged when an ACK is received. ambiguity about which packet is acknowledged when an ACK is received.
Consequently, more accurate RTT measurements can be made, spurious Consequently, more accurate RTT measurements can be made, spurious
retransmissions are trivially detected, and mechanisms such as Fast retransmissions are trivially detected, and mechanisms such as Fast
Retransmit can be applied universally, based only on packet number. Retransmit can be applied universally, based only on packet number.
This design point significantly simplifies loss detection mechanisms This design point significantly simplifies loss detection mechanisms
for QUIC. Most TCP mechanisms implicitly attempt to infer for QUIC. Most TCP mechanisms implicitly attempt to infer
transmission ordering based on TCP sequence numbers - a non-trivial transmission ordering based on TCP sequence numbers - a non-trivial
task, especially when TCP timestamps are not available. task, especially when TCP timestamps are not available.
skipping to change at page 7, line 8 skipping to change at page 7, line 8
the gap in the sequence number space to be filled, and so if a the gap in the sequence number space to be filled, and so if a
segment is lost multiple times in a row, the loss epoch may not end segment is lost multiple times in a row, the loss epoch may not end
for several round trips. Because both should reduce their congestion for several round trips. Because both should reduce their congestion
windows only once per epoch, QUIC will do it once for every round windows only once per epoch, QUIC will do it once for every round
trip that experiences loss, while TCP may only do it once across trip that experiences loss, while TCP may only do it once across
multiple round trips. multiple round trips.
4.4. No Reneging 4.4. No Reneging
QUIC ACKs contain information that is similar to TCP SACK, but QUIC QUIC ACKs contain information that is similar to TCP SACK, but QUIC
does not allow any acked packet to be reneged, greatly simplifying does not allow any acknowledged packet to be reneged, greatly
implementations on both sides and reducing memory pressure on the simplifying implementations on both sides and reducing memory
sender. pressure on the sender.
4.5. More ACK Ranges 4.5. More ACK Ranges
QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In
high loss environments, this speeds recovery, reduces spurious high loss environments, this speeds recovery, reduces spurious
retransmits, and ensures forward progress without relying on retransmits, and ensures forward progress without relying on
timeouts. timeouts.
4.6. Explicit Correction For Delayed Acknowledgements 4.6. Explicit Correction For Delayed Acknowledgements
QUIC endpoints measure the delay incurred between when a packet is QUIC endpoints measure the delay incurred between when a packet is
received and when the corresponding acknowledgment is sent, allowing received and when the corresponding acknowledgment is sent, allowing
a peer to maintain a more accurate round-trip time estimate; see a peer to maintain a more accurate round-trip time estimate; see
Section 13.2 of [QUIC-TRANSPORT]. Section 13.2 of [QUIC-TRANSPORT].
4.7. Probe Timeout Replaces RTO and TLP 4.7. Probe Timeout Replaces RTO and TLP
QUIC uses a probe timeout (see Section 6.2), with a timer based on QUIC uses a probe timeout (PTO; see Section 6.2), with a timer based
TCP's RTO computation. QUIC's PTO includes the peer's maximum on TCP's RTO computation. QUIC's PTO includes the peer's maximum
expected acknowledgement delay instead of using a fixed minimum expected acknowledgement delay instead of using a fixed minimum
timeout. QUIC does not collapse the congestion window until timeout. QUIC does not collapse the congestion window until
persistent congestion (Section 7.8) is declared, unlike TCP, which persistent congestion (Section 7.6) is declared, unlike TCP, which
collapses the congestion window upon expiry of an RTO. Instead of collapses the congestion window upon expiry of an RTO. Instead of
collapsing the congestion window and declaring everything in-flight collapsing the congestion window and declaring everything in-flight
lost, QUIC allows probe packets to temporarily exceed the congestion lost, QUIC allows probe packets to temporarily exceed the congestion
window whenever the timer expires. window whenever the timer expires.
In doing this, QUIC avoids unnecessary congestion window reductions, In doing this, QUIC avoids unnecessary congestion window reductions,
obviating the need for correcting mechanisms such as F-RTO [RFC5682]. obviating the need for correcting mechanisms such as F-RTO
Since QUIC does not collapse the congestion window on a PTO ([RFC5682]). Since QUIC does not collapse the congestion window on a
expiration, a QUIC sender is not limited from sending more in-flight PTO expiration, a QUIC sender is not limited from sending more in-
packets after a PTO expiration if it still has available congestion flight packets after a PTO expiration if it still has available
window. This occurs when a sender is application-limited and the PTO congestion window. This occurs when a sender is application-limited
timer expires. This is more aggressive than TCP's RTO mechanism when and the PTO timer expires. This is more aggressive than TCP's RTO
application-limited, but identical when not application-limited. mechanism when application-limited, but identical when not
application-limited.
A single packet loss at the tail does not indicate persistent A single packet loss at the tail does not indicate persistent
congestion, so QUIC specifies a time-based definition to ensure one congestion, so QUIC specifies a time-based definition to ensure one
or more packets are sent prior to a dramatic decrease in congestion or more packets are sent prior to a dramatic decrease in congestion
window; see Section 7.8. window; see Section 7.6.
4.8. The Minimum Congestion Window is Two Packets 4.8. The Minimum Congestion Window is Two Packets
TCP uses a minimum congestion window of one packet. However, loss of TCP uses a minimum congestion window of one packet. However, loss of
that single packet means that the sender needs to waiting for a PTO that single packet means that the sender needs to waiting for a PTO
(Section 6.2) to recover, which can be much longer than a round-trip (Section 6.2) to recover, which can be much longer than a round-trip
time. Sending a single ack-eliciting packet also increases the time. Sending a single ack-eliciting packet also increases the
chances of incurring additional latency when a receiver delays its chances of incurring additional latency when a receiver delays its
acknowledgement. acknowledgement.
skipping to change at page 8, line 47 skipping to change at page 8, line 47
* the largest acknowledged packet number is newly acknowledged, and * the largest acknowledged packet number is newly acknowledged, and
* at least one of the newly acknowledged packets was ack-eliciting. * at least one of the newly acknowledged packets was ack-eliciting.
The RTT sample, latest_rtt, is generated as the time elapsed since The RTT sample, latest_rtt, is generated as the time elapsed since
the largest acknowledged packet was sent: the largest acknowledged packet was sent:
latest_rtt = ack_time - send_time_of_largest_acked latest_rtt = ack_time - send_time_of_largest_acked
An RTT sample is generated using only the largest acknowledged packet An RTT sample is generated using only the largest acknowledged packet
in the received ACK frame. This is because a peer reports ACK delays in the received ACK frame. This is because a peer reports
for only the largest acknowledged packet in an ACK frame. While the acknowledgment delays for only the largest acknowledged packet in an
reported ACK delay is not used by the RTT sample measurement, it is ACK frame. While the reported acknowledgment delay is not used by
used to adjust the RTT sample in subsequent computations of the RTT sample measurement, it is used to adjust the RTT sample in
smoothed_rtt and rttvar Section 5.3. subsequent computations of smoothed_rtt and rttvar (Section 5.3).
To avoid generating multiple RTT samples for a single packet, an ACK To avoid generating multiple RTT samples for a single packet, an ACK
frame SHOULD NOT be used to update RTT estimates if it does not newly frame SHOULD NOT be used to update RTT estimates if it does not newly
acknowledge the largest acknowledged packet. acknowledge the largest acknowledged packet.
An RTT sample MUST NOT be generated on receiving an ACK frame that An RTT sample MUST NOT be generated on receiving an ACK frame that
does not newly acknowledge at least one ack-eliciting packet. A peer does not newly acknowledge at least one ack-eliciting packet. A peer
usually does not send an ACK frame when only non-ack-eliciting usually does not send an ACK frame when only non-ack-eliciting
packets are received. Therefore an ACK frame that contains packets are received. Therefore an ACK frame that contains
acknowledgements for only non-ack-eliciting packets could include an acknowledgements for only non-ack-eliciting packets could include an
arbitrarily large Ack Delay value. Ignoring such ACK frames avoids arbitrarily large ACK Delay value. Ignoring such ACK frames avoids
complications in subsequent smoothed_rtt and rttvar computations. complications in subsequent smoothed_rtt and rttvar computations.
A sender might generate multiple RTT samples per RTT when multiple A sender might generate multiple RTT samples per RTT when multiple
ACK frames are received within an RTT. As suggested in [RFC6298], ACK frames are received within an RTT. As suggested in [RFC6298],
doing so might result in inadequate history in smoothed_rtt and doing so might result in inadequate history in smoothed_rtt and
rttvar. Ensuring that RTT estimates retain sufficient history is an rttvar. Ensuring that RTT estimates retain sufficient history is an
open research question. open research question.
5.2. Estimating min_rtt 5.2. Estimating min_rtt
min_rtt is the minimum RTT observed for a given network path. min_rtt is the sender's estimate of the minimum RTT observed for a
min_rtt is set to the latest_rtt on the first RTT sample, and to the given network path. In this document, min_rtt is used by loss
lesser of min_rtt and latest_rtt on subsequent samples. In this detection to reject implausibly small rtt samples.
document, min_rtt is used by loss detection to reject implausibly
small rtt samples. min_rtt MUST be set to the latest_rtt on the first RTT sample.
min_rtt MUST be set to the lesser of min_rtt and latest_rtt
(Section 5.1) on all other samples.
An endpoint uses only locally observed times in computing the min_rtt An endpoint uses only locally observed times in computing the min_rtt
and does not adjust for ACK delays reported by the peer. Doing so and does not adjust for acknowledgment delays reported by the peer.
allows the endpoint to set a lower bound for the smoothed_rtt based Doing so allows the endpoint to set a lower bound for the
entirely on what it observes (see Section 5.3), and limits potential smoothed_rtt based entirely on what it observes (see Section 5.3),
underestimation due to erroneously-reported delays by the peer. and limits potential underestimation due to erroneously-reported
delays by the peer.
The RTT for a network path may change over time. If a path's actual The RTT for a network path may change over time. If a path's actual
RTT decreases, the min_rtt will adapt immediately on the first low RTT decreases, the min_rtt will adapt immediately on the first low
sample. If the path's actual RTT increases, the min_rtt will not sample. If the path's actual RTT increases however, the min_rtt will
adapt to it, allowing future RTT samples that are smaller than the not adapt to it, allowing future RTT samples that are smaller than
new RTT be included in smoothed_rtt. the new RTT to be included in smoothed_rtt.
Endpoints SHOULD set the min_rtt to the newest RTT sample after
persistent congestion is established. This is to allow a connection
to reset its estimate of min_rtt and smoothed_rtt (Section 5.3) after
a disruptive network event, and because it is possible that an
increase in path delay resulted in persistent congestion being
incorrectly declared.
Endpoints MAY re-establish the min_rtt at other times in the
connection, such as when traffic volume is low and an acknowledgement
is received with a low acknowledgement delay. Implementations SHOULD
NOT refresh the min_rtt value too often, since the actual minimum RTT
of the path is not frequently observable.
5.3. Estimating smoothed_rtt and rttvar 5.3. Estimating smoothed_rtt and rttvar
smoothed_rtt is an exponentially-weighted moving average of an smoothed_rtt is an exponentially-weighted moving average of an
endpoint's RTT samples, and rttvar is the variation in the RTT endpoint's RTT samples, and rttvar is the variation in the RTT
samples, estimated using a mean variation. samples, estimated using a mean variation.
The calculation of smoothed_rtt uses path latency after adjusting RTT The calculation of smoothed_rtt uses RTT samples after adjusting them
samples for acknowledgement delays. These delays are computed using for acknowledgement delays. These delays are computed using the ACK
the ACK Delay field of the ACK frame as described in Section 19.3 of Delay field of the ACK frame as described in Section 19.3 of
[QUIC-TRANSPORT]. For packets sent in the ApplicationData packet [QUIC-TRANSPORT].
number space, a peer limits any delay in sending an acknowledgement
for an ack-eliciting packet to no greater than the value it
advertised in the max_ack_delay transport parameter. Consequently,
when a peer reports an Ack Delay that is greater than its
max_ack_delay, the delay is attributed to reasons out of the peer's
control, such as scheduler latency at the peer or loss of previous
ACK frames. Any delays beyond the peer's max_ack_delay are therefore
considered effectively part of path delay and incorporated into the
smoothed_rtt estimate.
When adjusting an RTT sample using peer-reported acknowledgement The peer might report acknowledgement delays that are larger than the
delays, an endpoint: peer's max_ack_delay during the handshake (Section 13.2.1 of
[QUIC-TRANSPORT]). To account for this, the endpoint SHOULD ignore
max_ack_delay until the handshake is confirmed (Section 4.1.2 of
[QUIC-TLS]). When they occur, these large acknowledgement delays are
likely to be non-repeating and limited to the handshake. The
endpoint can therefore use them without limiting them to the
max_ack_delay, avoiding unnecessary inflation of the RTT estimate.
* MUST ignore the Ack Delay field of the ACK frame for packets sent Note however that a large acknowledgement delay can result in a
in the Initial and Handshake packet number space. substantially inflated smoothed_rtt, if there is either an error in
the peer's reporting of the acknowledgement delay or in the
endpoint's min_rtt estimate. Therefore, prior to handshake
confirmation, an endpoint MAY ignore RTT samples if adjusting the RTT
sample for acknowledgement delay causes the sample to be less than
the min_rtt.
* MUST use the lesser of the value reported in Ack Delay field of After the handshake is confirmed, any acknowledgement delays reported
the ACK frame and the peer's max_ack_delay transport parameter. by the peer that are greater than the peer's max_ack_delay are
attributed to unintentional but potentially repeating delays, such as
scheduler latency at the peer or loss of previous acknowledgements.
Therefore, these extra delays are considered effectively part of path
delay and incorporated into the RTT estimate.
* MUST NOT apply the adjustment if the resulting RTT sample is Therefore, when adjusting an RTT sample using peer-reported
smaller than the min_rtt. This limits the underestimation that a acknowledgement delays, an endpoint:
misreporting peer can cause to the smoothed_rtt.
* MAY ignore the acknowledgement delay for Initial packets, since
these acknowledgements are not delayed by the peer (Section 13.2.1
of [QUIC-TRANSPORT]);
* SHOULD ignore the peer's max_ack_delay until the handshake is
confirmed;
* MUST use the lesser of the acknowledgement delay and the peer's
max_ack_delay after the handshake is confirmed; and
* MUST NOT subtract the acknowledgement delay from the RTT sample if
the resulting value is smaller than the min_rtt. This limits the
underestimation of the smoothed_rtt due to a misreporting peer.
Additionally, an endpoint might postpone the processing of
acknowledgements when the corresponding decryption keys are not
immediately available. For example, a client might receive an
acknowledgement for a 0-RTT packet that it cannot decrypt because
1-RTT packet protection keys are not yet available to it. In such
cases, an endpoint SHOULD subtract such local delays from its RTT
sample until the handshake is confirmed.
smoothed_rtt and rttvar are computed as follows, similar to smoothed_rtt and rttvar are computed as follows, similar to
[RFC6298]. [RFC6298].
When there are no samples for a network path, and on the first RTT When there are no samples for a network path, and on the first RTT
sample for the network path: sample for the network path:
smoothed_rtt = rtt_sample smoothed_rtt = rtt_sample
rttvar = rtt_sample / 2 rttvar = rtt_sample / 2
Before any RTT samples are available, the initial RTT is used as Before any RTT samples are available, the initial RTT is used as
rtt_sample. On the first RTT sample for the network path, that rtt_sample. On the first RTT sample for the network path, that
sample is used as rtt_sample. This ensures that the first sample is used as rtt_sample. This ensures that the first
measurement erases the history of any persisted or default values. measurement erases the history of any persisted or default values.
On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows: On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:
ack_delay = min(Ack Delay in ACK Frame, max_ack_delay) ack_delay = decoded acknowledgement delay from ACK frame
if (handshake confirmed):
ack_delay = min(ack_delay, max_ack_delay)
adjusted_rtt = latest_rtt adjusted_rtt = latest_rtt
if (min_rtt + ack_delay < latest_rtt): if (min_rtt + ack_delay < latest_rtt):
adjusted_rtt = latest_rtt - ack_delay adjusted_rtt = latest_rtt - ack_delay
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
rttvar_sample = abs(smoothed_rtt - adjusted_rtt) rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
6. Loss Detection 6. Loss Detection
QUIC senders use acknowledgements to detect lost packets, and a probe QUIC senders use acknowledgements to detect lost packets, and a probe
time out (see Section 6.2) to ensure acknowledgements are received. time out (see Section 6.2) to ensure acknowledgements are received.
This section provides a description of these algorithms. This section provides a description of these algorithms.
If a packet is lost, the QUIC transport needs to recover from that If a packet is lost, the QUIC transport needs to recover from that
loss, such as by retransmitting the data, sending an updated frame, loss, such as by retransmitting the data, sending an updated frame,
or abandoning the frame. For more information, see Section 13.3 of or discarding the frame. For more information, see Section 13.3 of
[QUIC-TRANSPORT]. [QUIC-TRANSPORT].
6.1. Acknowledgement-based Detection Loss detection is separate per packet number space, unlike RTT
measurement and congestion control, because RTT and congestion
control are properties of the path, whereas loss detection also
relies upon key availability.
6.1. Acknowledgement-Based Detection
Acknowledgement-based loss detection implements the spirit of TCP's Acknowledgement-based loss detection implements the spirit of TCP's
Fast Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK], Fast Retransmit ([RFC5681]), Early Retransmit ([RFC5827]), FACK
SACK loss recovery [RFC6675], and RACK [RACK]. This section provides ([FACK]), SACK loss recovery ([RFC6675]), and RACK ([RACK]). This
an overview of how these algorithms are implemented in QUIC. section provides an overview of how these algorithms are implemented
in QUIC.
A packet is declared lost if it meets all the following conditions: A packet is declared lost if it meets all the following conditions:
* The packet is unacknowledged, in-flight, and was sent prior to an * The packet is unacknowledged, in-flight, and was sent prior to an
acknowledged packet. acknowledged packet.
* Either its packet number is kPacketThreshold smaller than an * Either its packet number is kPacketThreshold smaller than an
acknowledged packet (Section 6.1.1), or it was sent long enough in acknowledged packet (Section 6.1.1), or it was sent long enough in
the past (Section 6.1.2). the past (Section 6.1.2).
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Implementations can detect spurious retransmissions and increase the Implementations can detect spurious retransmissions and increase the
reordering threshold in packets or time to reduce future spurious reordering threshold in packets or time to reduce future spurious
retransmissions and loss events. Implementations with adaptive time retransmissions and loss events. Implementations with adaptive time
thresholds MAY choose to start with smaller initial reordering thresholds MAY choose to start with smaller initial reordering
thresholds to minimize recovery latency. thresholds to minimize recovery latency.
6.1.1. Packet Threshold 6.1.1. Packet Threshold
The RECOMMENDED initial value for the packet reordering threshold The RECOMMENDED initial value for the packet reordering threshold
(kPacketThreshold) is 3, based on best practices for TCP loss (kPacketThreshold) is 3, based on best practices for TCP loss
detection [RFC5681] [RFC6675]. Implementations SHOULD NOT use a detection ([RFC5681], [RFC6675]). In order to remain similar to TCP,
packet threshold less than 3, to keep in line with TCP [RFC5681]. implementations SHOULD NOT use a packet threshold less than 3; see
[RFC5681].
Some networks may exhibit higher degrees of reordering, causing a Some networks may exhibit higher degrees of packet reordering,
sender to detect spurious losses. Algorithms that increase the causing a sender to detect spurious losses. Additionally, packet
reordering threshold after spuriously detecting losses, such as TCP- reordering could be more common with QUIC than TCP, because network
NCR [RFC4653], have proven to be useful in TCP and are expected to at elements that could observe and reorder out-of-order TCP packets
least as useful in QUIC. Re-ordering could be more common with QUIC cannot do that for QUIC, because packet numbers are encrypted.
than TCP, because network elements cannot observe and fix the order Algorithms that increase the reordering threshold after spuriously
of out-of-order packets. detecting losses, such as RACK [RACK], have proven to be useful in
TCP and are expected to be at least as useful in QUIC.
6.1.2. Time Threshold 6.1.2. Time Threshold
Once a later packet within the same packet number space has been Once a later packet within the same packet number space has been
acknowledged, an endpoint SHOULD declare an earlier packet lost if it acknowledged, an endpoint SHOULD declare an earlier packet lost if it
was sent a threshold amount of time in the past. To avoid declaring was sent a threshold amount of time in the past. To avoid declaring
packets as lost too early, this time threshold MUST be set to at packets as lost too early, this time threshold MUST be set to at
least the local timer granularity, as indicated by the kGranularity least the local timer granularity, as indicated by the kGranularity
constant. The time threshold is: constant. The time threshold is:
skipping to change at page 13, line 13 skipping to change at page 14, line 19
loss detection delay. loss detection delay.
6.2. Probe Timeout 6.2. Probe Timeout
A Probe Timeout (PTO) triggers sending one or two probe datagrams A Probe Timeout (PTO) triggers sending one or two probe datagrams
when ack-eliciting packets are not acknowledged within the expected when ack-eliciting packets are not acknowledged within the expected
period of time or the server may not have validated the client's period of time or the server may not have validated the client's
address. A PTO enables a connection to recover from loss of tail address. A PTO enables a connection to recover from loss of tail
packets or acknowledgements. packets or acknowledgements.
As with loss detection, the probe timeout is per packet number space.
That is, a PTO value is computed per packet number space.
A PTO timer expiration event does not indicate packet loss and MUST A PTO timer expiration event does not indicate packet loss and MUST
NOT cause prior unacknowledged packets to be marked as lost. When an NOT cause prior unacknowledged packets to be marked as lost. When an
acknowledgement is received that newly acknowledges packets, loss acknowledgement is received that newly acknowledges packets, loss
detection proceeds as dictated by packet and time threshold detection proceeds as dictated by packet and time threshold
mechanisms; see Section 6.1. mechanisms; see Section 6.1.
As with loss detection, the probe timeout is per packet number space.
The PTO algorithm used in QUIC implements the reliability functions The PTO algorithm used in QUIC implements the reliability functions
of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for
TCP [RFC5682]. The timeout computation is based on TCP's TCP [RFC5682]. The timeout computation is based on TCP's
retransmission timeout period [RFC6298]. retransmission timeout period [RFC6298].
6.2.1. Computing PTO 6.2.1. Computing PTO
When an ack-eliciting packet is transmitted, the sender schedules a When an ack-eliciting packet is transmitted, the sender schedules a
timer for the PTO period as follows: timer for the PTO period as follows:
PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay
The PTO period is the amount of time that a sender ought to wait for The PTO period is the amount of time that a sender ought to wait for
an acknowledgement of a sent packet. This time period includes the an acknowledgement of a sent packet. This time period includes the
estimated network roundtrip-time (smoothed_rtt), the variation in the estimated network roundtrip-time (smoothed_rtt), the variation in the
estimate (4*rttvar), and max_ack_delay, to account for the maximum estimate (4*rttvar), and max_ack_delay, to account for the maximum
time by which a receiver might delay sending an acknowledgement. time by which a receiver might delay sending an acknowledgement.
When the PTO is armed for Initial or Handshake packet number spaces,
the max_ack_delay is 0, as specified in 13.2.1 of [QUIC-TRANSPORT].
The PTO value MUST be set to at least kGranularity, to avoid the When the PTO is armed for Initial or Handshake packet number spaces,
timer expiring immediately. the max_ack_delay in the PTO period computation is set to 0, since
the peer is expected to not delay these packets intentionally; see
13.2.1 of [QUIC-TRANSPORT].
A sender recomputes and may need to reset its PTO timer every time an The PTO period MUST be at least kGranularity, to avoid the timer
ack-eliciting packet is sent or acknowledged, when the handshake is expiring immediately.
confirmed, or when Initial or Handshake keys are discarded. This
ensures the PTO is always set based on the latest RTT information and
for the last sent packet in the correct packet number space.
When ack-eliciting packets in multiple packet number spaces are in When ack-eliciting packets in multiple packet number spaces are in
flight, the timer MUST be set for the packet number space with the flight, the timer MUST be set to the earlier value of the Initial and
earliest timeout, with one exception. The ApplicationData packet Handshake packet number spaces.
number space (Section 4.1.1 of [QUIC-TLS]) MUST be ignored until the
handshake completes. Not arming the PTO for ApplicationData prevents An endpoint MUST NOT set its PTO timer for the application data
a client from retransmitting a 0-RTT packet on a PTO expiration packet number space until the handshake is confirmed. Doing so
before confirming that the server is able to decrypt 0-RTT packets, prevents the endpoint from retransmitting information in packets when
and prevents a server from sending a 1-RTT packet on a PTO expiration either the peer does not yet have the keys to process them or the
before it has the keys to process an acknowledgement. endpoint does not yet have the keys to process their
acknowledgements. For example, this can happen when a client sends
0-RTT packets to the server; it does so without knowing whether the
server will be able to decrypt them. Similarly, this can happen when
a server sends 1-RTT packets before confirming that the client has
verified the server's certificate and can therefore read these 1-RTT
packets.
A sender SHOULD restart its PTO timer every time an ack-eliciting
packet is sent or acknowledged, when the handshake is confirmed
(Section 4.1.2 of [QUIC-TLS]), or when Initial or Handshake keys are
discarded (Section 9 of [QUIC-TLS]). This ensures the PTO is always
set based on the latest estimate of the round-trip time and for the
correct packet across packet number spaces.
When a PTO timer expires, the PTO backoff MUST be increased, When a PTO timer expires, the PTO backoff MUST be increased,
resulting in the PTO period being set to twice its current value. resulting in the PTO period being set to twice its current value.
The PTO backoff factor is reset when an acknowledgement is received, The PTO backoff factor is reset when an acknowledgement is received,
except in the following case. A server might take longer to respond except in the following case. A server might take longer to respond
to packets during the handshake than otherwise. To protect such a to packets during the handshake than otherwise. To protect such a
server from repeated client probes, the PTO backoff is not reset at a server from repeated client probes, the PTO backoff is not reset at a
client that is not yet certain that the server has finished client that is not yet certain that the server has finished
validating the client's address. That is, a client does not reset validating the client's address. That is, a client does not reset
the PTO backoff factor on receiving acknowledgements until it the PTO backoff factor on receiving acknowledgements until the
receives a HANDSHAKE_DONE frame or an acknowledgement for one of its handshake is confirmed; see Section 4.1.2 of [QUIC-TLS].
Handshake or 1-RTT packets.
This exponential reduction in the sender's rate is important because This exponential reduction in the sender's rate is important because
consecutive PTOs might be caused by loss of packets or consecutive PTOs might be caused by loss of packets or
acknowledgements due to severe congestion. Even when there are ack- acknowledgements due to severe congestion. Even when there are ack-
eliciting packets in-flight in multiple packet number spaces, the eliciting packets in-flight in multiple packet number spaces, the
exponential increase in probe timeout occurs across all spaces to exponential increase in probe timeout occurs across all spaces to
prevent excess load on the network. For example, a timeout in the prevent excess load on the network. For example, a timeout in the
Initial packet number space doubles the length of the timeout in the Initial packet number space doubles the length of the timeout in the
Handshake packet number space. Handshake packet number space.
The life of a connection that is experiencing consecutive PTOs is The time length of a connection that is experiencing consecutive PTOs
limited by the endpoint's idle timeout. is limited by the endpoint's idle timeout.
The probe timer MUST NOT be set if the time threshold Section 6.1.2 The probe timer MUST NOT be set if the time threshold (Section 6.1.2)
loss detection timer is set. The time threshold loss detection timer loss detection timer is set. The time threshold loss detection timer
is expected to both expire earlier than the PTO and be less likely to is expected to both expire earlier than the PTO and be less likely to
spuriously retransmit data. spuriously retransmit data.
6.2.2. Handshakes and New Paths 6.2.2. Handshakes and New Paths
Resumed connections over the same network MAY use the previous Resumed connections over the same network MAY use the previous
connection's final smoothed RTT value as the resumed connection's connection's final smoothed RTT value as the resumed connection's
initial RTT. When no previous RTT is available, the initial RTT initial RTT. When no previous RTT is available, the initial RTT
SHOULD be set to 333ms, resulting in a 1 second initial timeout, as SHOULD be set to 333ms, resulting in a 1 second initial timeout, as
recommended in [RFC6298]. recommended in [RFC6298].
A connection MAY use the delay between sending a PATH_CHALLENGE and A connection MAY use the delay between sending a PATH_CHALLENGE and
receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in
Appendix A.2) for a new path, but the delay SHOULD NOT be considered Appendix A.2) for a new path, but the delay SHOULD NOT be considered
an RTT sample. an RTT sample.
Prior to handshake completion, when few to none RTT samples have been
generated, it is possible that the probe timer expiration is due to
an incorrect RTT estimate at the client. To allow the client to
improve its RTT estimate, the new packet that it sends MUST be ack-
eliciting.
Initial packets and Handshake packets could be never acknowledged, Initial packets and Handshake packets could be never acknowledged,
but they are removed from bytes in flight when the Initial and but they are removed from bytes in flight when the Initial and
Handshake keys are discarded, as described below in Section 6.4. Handshake keys are discarded, as described below in Section 6.4.
When Initial or Handshake keys are discarded, the PTO and loss When Initial or Handshake keys are discarded, the PTO and loss
detection timers MUST be reset, because discarding keys indicates detection timers MUST be reset, because discarding keys indicates
forward progress and the loss detection timer might have been set for forward progress and the loss detection timer might have been set for
a now discarded packet number space. a now discarded packet number space.
6.2.2.1. Before Address Validation 6.2.2.1. Before Address Validation
Until the server has validated the client's address on the path, the Until the server has validated the client's address on the path, the
amount of data it can send is limited to three times the amount of amount of data it can send is limited to three times the amount of
data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If
no additional data can be sent, the server's PTO timer MUST NOT be no additional data can be sent, the server's PTO timer MUST NOT be
armed until datagrams have been received from the client, because armed until datagrams have been received from the client, because
packets sent on PTO count against the anti-amplification limit. Note packets sent on PTO count against the anti-amplification limit. Note
that the server could fail to validate the client's address even if that the server could fail to validate the client's address even if
0-RTT is accepted. 0-RTT is accepted.
Since the server could be blocked until more packets are received Since the server could be blocked until more datagrams are received
from the client, it is the client's responsibility to send packets to from the client, it is the client's responsibility to send packets to
unblock the server until it is certain that the server has finished unblock the server until it is certain that the server has finished
its address validation (see Section 8 of [QUIC-TRANSPORT]). That is, its address validation (see Section 8 of [QUIC-TRANSPORT]). That is,
the client MUST set the probe timer if the client has not received an the client MUST set the probe timer if the client has not received an
acknowledgement for one of its Handshake or 1-RTT packets, and has acknowledgement for one of its Handshake packets and the handshake is
not received a HANDSHAKE_DONE frame. If Handshake keys are available not confirmed (see Section 4.1.2 of [QUIC-TLS]), even if there are no
to the client, it MUST send a Handshake packet, and otherwise it MUST packets in flight. When the PTO fires, the client MUST send a
send an Initial packet in a UDP datagram of at least 1200 bytes. Handshake packet if it has Handshake keys, otherwise it MUST send an
Initial packet in a UDP datagram of at least 1200 bytes.
A client could have received and acknowledged a Handshake packet,
causing it to discard state for the Initial packet number space, but
not sent any ack-eliciting Handshake packets. In this case, the PTO
is set from the current time.
6.2.3. Speeding Up Handshake Completion 6.2.3. Speeding Up Handshake Completion
When a server receives an Initial packet containing duplicate CRYPTO When a server receives an Initial packet containing duplicate CRYPTO
data, it can assume the client did not receive all of the server's data, it can assume the client did not receive all of the server's
CRYPTO data sent in Initial packets, or the client's estimated RTT is CRYPTO data sent in Initial packets, or the client's estimated RTT is
too small. When a client receives Handshake or 1-RTT packets prior too small. When a client receives Handshake or 1-RTT packets prior
to obtaining Handshake keys, it may assume some or all of the to obtaining Handshake keys, it may assume some or all of the
server's Initial packets were lost. server's Initial packets were lost.
To speed up handshake completion under these conditions, an endpoint To speed up handshake completion under these conditions, an endpoint
MAY send a packet containing unacknowledged CRYPTO data earlier than MAY send a packet containing unacknowledged CRYPTO data earlier than
the PTO expiry, subject to address validation limits; see Section 8.1 the PTO expiry, subject to the address validation limits in
of [QUIC-TRANSPORT]. Section 8.1 of [QUIC-TRANSPORT].
Peers can also use coalesced packets to ensure that each datagram Endpoints can also use coalesced packets to ensure that each datagram
elicits at least one acknowledgement. For example, clients can elicits at least one acknowledgement. For example, a client can
coalesce an Initial packet containing PING and PADDING frames with a coalesce an Initial packet containing PING and PADDING frames with a
0-RTT data packet and a server can coalesce an Initial packet 0-RTT data packet and a server can coalesce an Initial packet
containing a PING frame with one or more packets in its first flight. containing a PING frame with one or more packets in its first flight.
6.2.4. Sending Probe Packets 6.2.4. Sending Probe Packets
When a PTO timer expires, a sender MUST send at least one ack- When a PTO timer expires, a sender MUST send at least one ack-
eliciting packet in the packet number space as a probe, unless there eliciting packet in the packet number space as a probe. An endpoint
is no data available to send. An endpoint MAY send up to two full- MAY send up to two full-sized datagrams containing ack-eliciting
sized datagrams containing ack-eliciting packets, to avoid an packets, to avoid an expensive consecutive PTO expiration due to a
expensive consecutive PTO expiration due to a single lost datagram or single lost datagram or transmit data from multiple packet number
transmit data from multiple packet number spaces. All probe packets spaces. All probe packets sent on a PTO MUST be ack-eliciting.
sent on a PTO MUST be ack-eliciting.
In addition to sending data in the packet number space for which the In addition to sending data in the packet number space for which the
timer expired, the sender SHOULD send ack-eliciting packets from timer expired, the sender SHOULD send ack-eliciting packets from
other packet number spaces with in-flight data, coalescing packets if other packet number spaces with in-flight data, coalescing packets if
possible. This is particularly valuable when the server has both possible. This is particularly valuable when the server has both
Initial and Handshake data in-flight or the client has both Handshake Initial and Handshake data in-flight or the client has both Handshake
and ApplicationData in-flight, because the peer might only have and Application Data in-flight, because the peer might only have
receive keys for one of the two packet number spaces. receive keys for one of the two packet number spaces.
If the sender wants to elicit a faster acknowledgement on PTO, it can If the sender wants to elicit a faster acknowledgement on PTO, it can
skip a packet number to eliminate the ack delay. skip a packet number to eliminate the acknowledgment delay.
When the PTO timer expires, and there is new or previously sent When the PTO timer expires, an ack-eliciting packet MUST be sent. An
unacknowledged data, it MUST be sent. A probe packet SHOULD carry endpoint SHOULD include new data in this packet. Previously sent
new data when possible. A probe packet MAY carry retransmitted data MAY be sent if no new data can be sent. Implementations MAY use
unacknowledged data when new data is unavailable, when flow control alternative strategies for determining the content of probe packets,
does not permit new data to be sent, or to opportunistically reduce including sending new or retransmitted data based on the
loss recovery delay. Implementations MAY use alternative strategies application's priorities.
for determining the content of probe packets, including sending new
or retransmitted data based on the application's priorities.
It is possible the sender has no new or previously-sent data to send. It is possible the sender has no new or previously-sent data to send.
As an example, consider the following sequence of events: new As an example, consider the following sequence of events: new
application data is sent in a STREAM frame, deemed lost, then application data is sent in a STREAM frame, deemed lost, then
retransmitted in a new packet, and then the original transmission is retransmitted in a new packet, and then the original transmission is
acknowledged. When there is no data to send, the sender SHOULD send acknowledged. When there is no data to send, the sender SHOULD send
a PING or other ack-eliciting frame in a single packet, re-arming the a PING or other ack-eliciting frame in a single packet, re-arming the
PTO timer. PTO timer.
Alternatively, instead of sending an ack-eliciting packet, the sender Alternatively, instead of sending an ack-eliciting packet, the sender
skipping to change at page 17, line 44 skipping to change at page 19, line 7
connection state, in particular cryptographic handshake messages, is connection state, in particular cryptographic handshake messages, is
retained; see Section 17.2.5 of [QUIC-TRANSPORT]. retained; see Section 17.2.5 of [QUIC-TRANSPORT].
The client MAY compute an RTT estimate to the server as the time The client MAY compute an RTT estimate to the server as the time
period from when the first Initial was sent to when a Retry or a period from when the first Initial was sent to when a Retry or a
Version Negotiation packet is received. The client MAY use this Version Negotiation packet is received. The client MAY use this
value in place of its default for the initial RTT estimate. value in place of its default for the initial RTT estimate.
6.4. Discarding Keys and Packet State 6.4. Discarding Keys and Packet State
When packet protection keys are discarded (see Section 4.10 of When packet protection keys are discarded (see Section 4.8 of
[QUIC-TLS]), all packets that were sent with those keys can no longer [QUIC-TLS]), all packets that were sent with those keys can no longer
be acknowledged because their acknowledgements cannot be processed be acknowledged because their acknowledgements cannot be processed
anymore. The sender MUST discard all recovery state associated with anymore. The sender MUST discard all recovery state associated with
those packets and MUST remove them from the count of bytes in flight. those packets and MUST remove them from the count of bytes in flight.
Endpoints stop sending and receiving Initial packets once they start Endpoints stop sending and receiving Initial packets once they start
exchanging Handshake packets; see Section 17.2.2.1 of exchanging Handshake packets; see Section 17.2.2.1 of
[QUIC-TRANSPORT]. At this point, recovery state for all in-flight [QUIC-TRANSPORT]. At this point, recovery state for all in-flight
Initial packets is discarded. Initial packets is discarded.
When 0-RTT is rejected, recovery state for all in-flight 0-RTT When 0-RTT is rejected, recovery state for all in-flight 0-RTT
packets is discarded. packets is discarded.
If a server accepts 0-RTT, but does not buffer 0-RTT packets that If a server accepts 0-RTT, but does not buffer 0-RTT packets that
arrive before Initial packets, early 0-RTT packets will be declared arrive before Initial packets, early 0-RTT packets will be declared
lost, but that is expected to be infrequent. lost, but that is expected to be infrequent.
It is expected that keys are discarded after packets encrypted with It is expected that keys are discarded after packets encrypted with
them would be acknowledged or declared lost. Initial secrets however them would be acknowledged or declared lost. However, Initial
might be destroyed sooner, as soon as handshake keys are available; secrets are discarded as soon as handshake keys are proven to be
see Section 4.11.1 of [QUIC-TLS]. available to both client and server; see Section 4.9.1 of [QUIC-TLS].
7. Congestion Control 7. Congestion Control
This document specifies a congestion controller for QUIC similar to This document specifies a congestion controller for QUIC similar to
TCP NewReno [RFC6582]. TCP NewReno ([RFC6582]).
The signals QUIC provides for congestion control are generic and are The signals QUIC provides for congestion control are generic and are
designed to support different algorithms. Endpoints can unilaterally designed to support different algorithms. Endpoints can unilaterally
choose a different algorithm to use, such as Cubic [RFC8312]. choose a different algorithm to use, such as Cubic ([RFC8312]).
If an endpoint uses a different controller than that specified in If an endpoint uses a different controller than that specified in
this document, the chosen controller MUST conform to the congestion this document, the chosen controller MUST conform to the congestion
control guidelines specified in Section 3.1 of [RFC8085]. control guidelines specified in Section 3.1 of [RFC8085].
Similar to TCP, packets containing only ACK frames do not count Similar to TCP, packets containing only ACK frames do not count
towards bytes in flight and are not congestion controlled. Unlike towards bytes in flight and are not congestion controlled. Unlike
TCP, QUIC can detect the loss of these packets and MAY use that TCP, QUIC can detect the loss of these packets and MAY use that
information to adjust the congestion controller or the rate of ACK- information to adjust the congestion controller or the rate of ACK-
only packets being sent, but this document does not describe a only packets being sent, but this document does not describe a
mechanism for doing so. mechanism for doing so.
The algorithm in this document specifies and uses the controller's The algorithm in this document specifies and uses the controller's
congestion window in bytes. congestion window in bytes.
An endpoint MUST NOT send a packet if it would cause bytes_in_flight An endpoint MUST NOT send a packet if it would cause bytes_in_flight
(see Appendix B.2) to be larger than the congestion window, unless (see Appendix B.2) to be larger than the congestion window, unless
the packet is sent on a PTO timer expiration; see Section 6.2. the packet is sent on a PTO timer expiration (see Section 6.2) or
when entering recovery (see Section 7.3.2).
7.1. Explicit Congestion Notification 7.1. Explicit Congestion Notification
If a path has been verified to support ECN [RFC3168] [RFC8311], QUIC If a path has been verified to support ECN ([RFC3168], [RFC8311]),
treats a Congestion Experienced (CE) codepoint in the IP header as a QUIC treats a Congestion Experienced (CE) codepoint in the IP header
signal of congestion. This document specifies an endpoint's response as a signal of congestion. This document specifies an endpoint's
when its peer receives packets with the ECN-CE codepoint. response when its peer receives packets with the ECN-CE codepoint.
7.2. Initial and Minimum Congestion Window 7.2. Initial and Minimum Congestion Window
QUIC begins every connection in slow start with the congestion window QUIC begins every connection in slow start with the congestion window
set to an initial value. Endpoints SHOULD use an initial congestion set to an initial value. Endpoints SHOULD use an initial congestion
window of 10 times the maximum datagram size (max_datagram_size), window of 10 times the maximum datagram size (max_datagram_size),
limited to the larger of 14720 or twice the maximum datagram size. limited to the larger of 14720 or twice the maximum datagram size.
This follows the analysis and recommendations in [RFC6928], This follows the analysis and recommendations in [RFC6928],
increasing the byte limit to account for the smaller 8 byte overhead increasing the byte limit to account for the smaller 8 byte overhead
of UDP compared to the 20 byte overhead for TCP. of UDP compared to the 20 byte overhead for TCP.
If the maximum datagram size changes during the connection, the
initial congestion window SHOULD be recalculated with the new size.
If the maximum datagram size is decreased in order to complete the
handshake, the congestion window SHOULD be set to the new initial
congestion window.
Prior to validating the client's address, the server can be further Prior to validating the client's address, the server can be further
limited by the anti-amplification limit as specified in Section 8.1 limited by the anti-amplification limit as specified in Section 8.1
of [QUIC-TRANSPORT]. Though the anti-amplification limit can prevent of [QUIC-TRANSPORT]. Though the anti-amplification limit can prevent
the congestion window from being fully utilized and therefore slow the congestion window from being fully utilized and therefore slow
down the increase in congestion window, it does not directly affect down the increase in congestion window, it does not directly affect
the congestion window. the congestion window.
The minimum congestion window is the smallest value the congestion The minimum congestion window is the smallest value the congestion
window can decrease to as a response to loss, ECN-CE, or persistent window can decrease to as a response to loss, ECN-CE, or persistent
congestion. The RECOMMENDED value is 2 * max_datagram_size. congestion. The RECOMMENDED value is 2 * max_datagram_size.
7.3. Slow Start 7.3. Congestion Control States
While in slow start, QUIC increases the congestion window by the The NewReno congestion controller described in this document has
number of bytes acknowledged when each acknowledgment is processed, three distinct states, as shown in Figure 1.
resulting in exponential growth of the congestion window.
QUIC exits slow start upon loss or upon increase in the ECN-CE New Path or +------------+
counter. When slow start is exited, the congestion window halves and persistent congestion | Slow |
the slow start threshold is set to the new congestion window. QUIC (O)---------------------->| Start |
re-enters slow start any time the congestion window is less than the +------------+
slow start threshold, which only occurs after persistent congestion |
is declared. Loss or |
ECN-CE increase |
v
+------------+ Loss or +------------+
| Congestion | ECN-CE increase | Recovery |
| Avoidance |------------------>| Period |
+------------+ +------------+
^ |
| |
+----------------------------+
Acknowledgment of packet
sent during recovery
7.4. Congestion Avoidance Figure 1: Congestion Control States and Transitions
Slow start exits to congestion avoidance. Congestion avoidance uses These states and the transitions between them are described in
an Additive Increase Multiplicative Decrease (AIMD) approach that subsequent sections.
increases the congestion window by one maximum packet size per
congestion window acknowledged. When a loss or ECN-CE marking is
detected, NewReno halves the congestion window, sets the slow start
threshold to the new congestion window, and then enters the recovery
period.
7.5. Recovery Period 7.3.1. Slow Start
A recovery period is entered when loss or ECN-CE marking of a packet A NewReno sender is in slow start any time the congestion window is
is detected in congestion avoidance after the congestion window and below the slow start threshold. A sender begins in slow start
slow start threshold have been decreased. A recovery period ends because the slow start threshold is initialized to an infinite value.
While a sender is in slow start, the congestion window increases by
the number of bytes acknowledged when each acknowledgment is
processed. This results in exponential growth of the congestion
window.
The sender MUST exit slow start and enter a recovery period when a
packet is lost or when the ECN-CE count reported by its peer
increases.
A sender re-enters slow start any time the congestion window is less
than the slow start threshold, which only occurs after persistent
congestion is declared.
7.3.2. Recovery
A NewReno sender enters a recovery period when it detects the loss of
a packet or the ECN-CE count reported by its peer increases. A
sender that is already in a recovery period stays in it and does not
re-enter it.
On entering a recovery period, a sender MUST set the slow start
threshold to half the value of the congestion window when loss is
detected. The congestion window MUST be set to the reduced value of
the slow start threshold before exiting the recovery period.
Implementations MAY reduce the congestion window immediately upon
entering a recovery period or use other mechanisms, such as
Proportional Rate Reduction ([PRR]), to reduce the congestion window
more gradually. If the congestion window is reduced immediately, a
single packet can be sent prior to reduction. This speeds up loss
recovery if the data in the lost packet is retransmitted and is
similar to TCP as described in Section 5 of [RFC6675].
The recovery period aims to limit congestion window reduction to once
per round trip. Therefore during a recovery period, the congestion
window does not change in response to new losses or increases in the
ECN-CE count.
A recovery period ends and the sender enters congestion avoidance
when a packet sent during the recovery period is acknowledged. This when a packet sent during the recovery period is acknowledged. This
is slightly different from TCP's definition of recovery, which ends is slightly different from TCP's definition of recovery, which ends
when the lost packet that started recovery is acknowledged. when the lost segment that started recovery is acknowledged
([RFC5681]).
The recovery period aims to limit congestion window reduction to once 7.3.3. Congestion Avoidance
per round trip. Therefore during recovery, the congestion window
remains unchanged irrespective of new losses or increases in the ECN-
CE counter.
When entering recovery, a single packet MAY be sent even if bytes in A NewReno sender is in congestion avoidance any time the congestion
flight now exceeds the recently reduced congestion window. This window is at or above the slow start threshold and not in a recovery
speeds up loss recovery if the data in the lost packet is period.
retransmitted and is similar to TCP as described in Section 5 of
[RFC6675]. If further packets are lost while the sender is in
recovery, sending any packets in response MUST obey the congestion
window limit.
7.6. Ignoring Loss of Undecryptable Packets A sender in congestion avoidance uses an Additive Increase
Multiplicative Decrease (AIMD) approach that MUST limit the increase
to the congestion window to at most one maximum datagram size for
each congestion window that is acknowledged.
The sender exits congestion avoidance and enters a recovery period
when a packet is lost or when the ECN-CE count reported by its peer
increases.
7.4. Ignoring Loss of Undecryptable Packets
During the handshake, some packet protection keys might not be During the handshake, some packet protection keys might not be
available when a packet arrives and the receiver can choose to drop available when a packet arrives and the receiver can choose to drop
the packet. In particular, Handshake and 0-RTT packets cannot be the packet. In particular, Handshake and 0-RTT packets cannot be
processed until the Initial packets arrive and 1-RTT packets cannot processed until the Initial packets arrive and 1-RTT packets cannot
be processed until the handshake completes. Endpoints MAY ignore the be processed until the handshake completes. Endpoints MAY ignore the
loss of Handshake, 0-RTT, and 1-RTT packets that might have arrived loss of Handshake, 0-RTT, and 1-RTT packets that might have arrived
before the peer had packet protection keys to process those packets. before the peer had packet protection keys to process those packets.
Endpoints MUST NOT ignore the loss of packets that were sent after Endpoints MUST NOT ignore the loss of packets that were sent after
the earliest acknowledged packet in a given packet number space. the earliest acknowledged packet in a given packet number space.
7.7. Probe Timeout 7.5. Probe Timeout
Probe packets MUST NOT be blocked by the congestion controller. A Probe packets MUST NOT be blocked by the congestion controller. A
sender MUST however count these packets as being additionally in sender MUST however count these packets as being additionally in
flight, since these packets add network load without establishing flight, since these packets add network load without establishing
packet loss. Note that sending probe packets might cause the packet loss. Note that sending probe packets might cause the
sender's bytes in flight to exceed the congestion window until an sender's bytes in flight to exceed the congestion window until an
acknowledgement is received that establishes loss or delivery of acknowledgement is received that establishes loss or delivery of
packets. packets.
7.8. Persistent Congestion 7.6. Persistent Congestion
When an ACK frame is received that establishes loss of all in-flight When a sender establishes loss of all in-flight packets sent over a
packets sent over a long enough period of time, the network is long enough duration, the network is considered to be experiencing
considered to be experiencing persistent congestion. Commonly, this persistent congestion.
can be established by consecutive PTOs, but since the PTO timer is
reset when a new ack-eliciting packet is sent, an explicit duration
must be used to account for those cases where PTOs do not occur or
are substantially delayed. The rationale for this threshold is to
enable a sender to use initial PTOs for aggressive probing, as TCP
does with Tail Loss Probe (TLP) [RACK], before establishing
persistent congestion, as TCP does with a Retransmission Timeout
(RTO) [RFC5681]. The RECOMMENDED value for
kPersistentCongestionThreshold is 3, which is approximately
equivalent to two TLPs before an RTO in TCP.
This duration is computed as follows: 7.6.1. Duration
(smoothed_rtt + 4 * rttvar + max_ack_delay) * The persistent congestion duration is computed as follows:
(smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay) *
kPersistentCongestionThreshold kPersistentCongestionThreshold
For example, assume: Unlike the PTO computation in Section 6.2, this duration includes the
max_ack_delay irrespective of the packet number spaces in which
losses are established.
smoothed_rtt = 1 This duration allows a sender to send as many packets before
rttvar = 0 establishing persistent congestion, including some in response to PTO
max_ack_delay = 0 expiration, as TCP does with Tail Loss Probes ([RACK]) and a
Retransmission Timeout ([RFC5681]).
The RECOMMENDED value for kPersistentCongestionThreshold is 3, which
is approximately equivalent to two TLPs before an RTO in TCP.
This design does not use consecutive PTO events to establish
persistent congestion, since application patterns impact PTO
expirations. For example, a sender that sends small amounts of data
with silence periods between them restarts the PTO timer every time
it sends, potentially preventing the PTO timer from expiring for a
long period of time, even when no acknowledgments are being received.
The use of a duration enables a sender to establish persistent
congestion without depending on PTO expiration.
7.6.2. Establishing Persistent Congestion
A sender establishes persistent congestion on receiving an
acknowledgement if at least two ack-eliciting packets are declared
lost, and:
* all packets, across all packet number spaces, sent between these
two send times are declared lost;
* the duration between the send times of these two packets exceeds
the persistent congestion duration (Section 7.6.1); and
* a prior RTT sample existed when both packets were sent.
The persistent congestion period SHOULD NOT start until there is at
least one RTT sample. Before the first RTT sample, a sender arms its
PTO timer based on the initial RTT (Section 6.2.2), which could be
substantially larger than the actual RTT. Requiring a prior RTT
sample prevents a sender from establishing persistent congestion with
potentially too few probes.
Since network congestion is not affected by packet number spaces,
persistent congestion SHOULD consider packets sent across packet
number spaces. A sender that does not have state for all packet
number spaces or an implementation that cannot compare send times
across packet number spaces MAY use state for just the packet number
space that was acknowledged.
When persistent congestion is declared, the sender's congestion
window MUST be reduced to the minimum congestion window
(kMinimumWindow), similar to a TCP sender's response on an RTO
([RFC5681]).
7.6.3. Example
The following example illustrates how a sender might establish
persistent congestion. Assume:
smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay = 2
kPersistentCongestionThreshold = 3 kPersistentCongestionThreshold = 3
If an ack-eliciting packet is sent at time t = 0, the following Consider the following sequence of events:
scenario would illustrate persistent congestion:
+------+------------------------+ +========+============================+
| Time | Action | | Time | Action |
+======+========================+ +========+============================+
| t=0 | Send Pkt #1 (App Data) | | t=0 | Send packet #1 (app data) |
+------+------------------------+ +--------+----------------------------+
| t=1 | Send Pkt #2 (PTO 1) | | t=1 | Send packet #2 (app data) |
+------+------------------------+ +--------+----------------------------+
| t=3 | Send Pkt #3 (PTO 2) | | t=1.2 | Recv acknowledgement of #1 |
+------+------------------------+ +--------+----------------------------+
| t=7 | Send Pkt #4 (PTO 3) | | t=2 | Send packet #3 (app data) |
+------+------------------------+ +--------+----------------------------+
| t=8 | Recv ACK of Pkt #4 | | t=3 | Send packet #4 (app data) |
+------+------------------------+ +--------+----------------------------+
| t=4 | Send packet #5 (app data) |
+--------+----------------------------+
| t=5 | Send packet #6 (app data) |
+--------+----------------------------+
| t=6 | Send packet #7 (app data) |
+--------+----------------------------+
| t=8 | Send packet #8 (PTO 1) |
+--------+----------------------------+
| t=12 | Send packet #9 (PTO 2) |
+--------+----------------------------+
| t=12.2 | Recv acknowledgement of #9 |
+--------+----------------------------+
Table 1 Table 1
The first three packets are determined to be lost when the Packets 2 through 8 are declared lost when the acknowledgement for
acknowledgement of packet 4 is received at t = 8. The congestion packet 9 is received at t = 12.2.
period is calculated as the time between the oldest and newest lost
packets: (3 - 0) = 3. The duration for persistent congestion is
equal to: (1 * kPersistentCongestionThreshold) = 3. Because the
threshold was reached and because none of the packets between the
oldest and the newest packets are acknowledged, the network is
considered to have experienced persistent congestion.
When persistent congestion is established, the sender's congestion The congestion period is calculated as the time between the oldest
window MUST be reduced to the minimum congestion window and newest lost packets: 8 - 1 = 7. The persistent congestion
(kMinimumWindow). This response of collapsing the congestion window duration is: 2 * 3 = 6. Because the threshold was reached and
on persistent congestion is functionally similar to a sender's because none of the packets between the oldest and the newest lost
response on a Retransmission Timeout (RTO) in TCP [RFC5681] after packets were acknowledged, the network is considered to have
Tail Loss Probes (TLP) [RACK]. experienced persistent congestion.
7.9. Pacing While this example shows PTO expiration, they are not required for
persistent congestion to be established.
This document does not specify a pacer, but it is RECOMMENDED that a 7.7. Pacing
sender pace sending of all in-flight packets based on input from the
congestion controller. Sending multiple packets into the network A sender SHOULD pace sending of all in-flight packets based on input
without any delay between them creates a packet burst that might from the congestion controller.
cause short-term congestion and losses. Implementations MUST either
use pacing or another method to limit such bursts to the initial Sending multiple packets into the network without any delay between
congestion window; see Section 7.2. them creates a packet burst that might cause short-term congestion
and losses. Senders MUST either use pacing or limit such bursts.
Senders SHOULD limit bursts to the initial congestion window; see
Section 7.2. A sender with knowledge that the network path to the
receiver can absorb larger bursts MAY use a higher limit.
An implementation should take care to architect its congestion An implementation should take care to architect its congestion
controller to work well with a pacer. For instance, a pacer might controller to work well with a pacer. For instance, a pacer might
wrap the congestion controller and control the availability of the wrap the congestion controller and control the availability of the
congestion window, or a pacer might pace out packets handed to it by congestion window, or a pacer might pace out packets handed to it by
the congestion controller. the congestion controller.
Timely delivery of ACK frames is important for efficient loss Timely delivery of ACK frames is important for efficient loss
recovery. Packets containing only ACK frames SHOULD therefore not be recovery. Packets containing only ACK frames SHOULD therefore not be
paced, to avoid delaying their delivery to the peer. paced, to avoid delaying their delivery to the peer.
skipping to change at page 23, line 22 skipping to change at page 26, line 35
can be computed by averaging the congestion window over the round- can be computed by averaging the congestion window over the round-
trip time. Expressed as a rate in bytes: trip time. Expressed as a rate in bytes:
rate = N * congestion_window / smoothed_rtt rate = N * congestion_window / smoothed_rtt
Or, expressed as an inter-packet interval: Or, expressed as an inter-packet interval:
interval = smoothed_rtt * packet_size / congestion_window / N interval = smoothed_rtt * packet_size / congestion_window / N
Using a value for "N" that is small, but at least 1 (for example, Using a value for "N" that is small, but at least 1 (for example,
1.25) ensures that variations in round-trip time don't result in 1.25) ensures that variations in round-trip time do not result in
under-utilization of the congestion window. Values of 'N' larger under-utilization of the congestion window. Values of 'N' larger
than 1 ultimately result in sending packets as acknowledgments are than 1 ultimately result in sending packets as acknowledgments are
received rather than when timers fire, provided the congestion window received rather than when timers fire, provided the congestion window
is fully utilized and acknowledgments arrive at regular intervals. is fully utilized and acknowledgments arrive at regular intervals.
Practical considerations, such as packetization, scheduling delays, Practical considerations, such as packetization, scheduling delays,
and computational efficiency, can cause a sender to deviate from this and computational efficiency, can cause a sender to deviate from this
rate over time periods that are much shorter than a round-trip time. rate over time periods that are much shorter than a round-trip time.
One possible implementation strategy for pacing uses a leaky bucket One possible implementation strategy for pacing uses a leaky bucket
algorithm, where the capacity of the "bucket" is limited to the algorithm, where the capacity of the "bucket" is limited to the
maximum burst size and the rate the "bucket" fills is determined by maximum burst size and the rate the "bucket" fills is determined by
the above function. the above function.
7.10. Under-utilizing the Congestion Window 7.8. Under-utilizing the Congestion Window
When bytes in flight is smaller than the congestion window and When bytes in flight is smaller than the congestion window and
sending is not pacing limited, the congestion window is under- sending is not pacing limited, the congestion window is under-
utilized. When this occurs, the congestion window SHOULD NOT be utilized. When this occurs, the congestion window SHOULD NOT be
increased in either slow start or congestion avoidance. This can increased in either slow start or congestion avoidance. This can
happen due to insufficient application data or flow control limits. happen due to insufficient application data or flow control limits.
A sender MAY use the pipeACK method described in Section 4.3 of A sender MAY use the pipeACK method described in Section 4.3 of
[RFC7661] to determine if the congestion window is sufficiently [RFC7661] to determine if the congestion window is sufficiently
utilized. utilized.
A sender that paces packets (see Section 7.9) might delay sending A sender that paces packets (see Section 7.7) might delay sending
packets and not fully utilize the congestion window due to this packets and not fully utilize the congestion window due to this
delay. A sender SHOULD NOT consider itself application limited if it delay. A sender SHOULD NOT consider itself application limited if it
would have fully utilized the congestion window without pacing delay. would have fully utilized the congestion window without pacing delay.
A sender MAY implement alternative mechanisms to update its A sender MAY implement alternative mechanisms to update its
congestion window after periods of under-utilization, such as those congestion window after periods of under-utilization, such as those
proposed for TCP in [RFC7661]. proposed for TCP in [RFC7661].
8. Security Considerations 8. Security Considerations
skipping to change at page 24, line 23 skipping to change at page 27, line 40
Congestion control fundamentally involves the consumption of signals Congestion control fundamentally involves the consumption of signals
- both loss and ECN codepoints - from unauthenticated entities. On- - both loss and ECN codepoints - from unauthenticated entities. On-
path attackers can spoof or alter these signals. An attacker can path attackers can spoof or alter these signals. An attacker can
cause endpoints to reduce their sending rate by dropping packets, or cause endpoints to reduce their sending rate by dropping packets, or
alter send rate by changing ECN codepoints. alter send rate by changing ECN codepoints.
8.2. Traffic Analysis 8.2. Traffic Analysis
Packets that carry only ACK frames can be heuristically identified by Packets that carry only ACK frames can be heuristically identified by
observing packet size. Acknowledgement patterns may expose observing packet size. Acknowledgement patterns may expose
information about link characteristics or application behavior. information about link characteristics or application behavior. To
Endpoints can use PADDING frames or bundle acknowledgments with other reduce leaked information, endpoints can bundle acknowledgments with
frames to reduce leaked information. other frames, or they can use PADDING frames at a potential cost to
performance.
8.3. Misreporting ECN Markings 8.3. Misreporting ECN Markings
A receiver can misreport ECN markings to alter the congestion A receiver can misreport ECN markings to alter the congestion
response of a sender. Suppressing reports of ECN-CE markings could response of a sender. Suppressing reports of ECN-CE markings could
cause a sender to increase their send rate. This increase could cause a sender to increase their send rate. This increase could
result in congestion and loss. result in congestion and loss.
A sender MAY attempt to detect suppression of reports by marking A sender MAY attempt to detect suppression of reports by marking
occasional packets that they send with ECN-CE. If a packet sent with occasional packets that they send with ECN-CE. If a packet sent with
ECN-CE is not reported as having been CE marked when the packet is ECN-CE is not reported as having been CE marked when the packet is
acknowledged, then the sender SHOULD disable ECN for that path. acknowledged, then the sender SHOULD disable ECN for that path.
Reporting additional ECN-CE markings will cause a sender to reduce Reporting additional ECN-CE markings will cause a sender to reduce
their sending rate, which is similar in effect to advertising reduced their sending rate, which is similar in effect to advertising reduced
connection flow control limits and so no advantage is gained by doing connection flow control limits and so no advantage is gained by doing
so. so.
Endpoints choose the congestion controller that they use. Though Endpoints choose the congestion controller that they use. Congestion
congestion controllers generally treat reports of ECN-CE markings as controllers respond to reports of ECN-CE by reducing their rate, but
equivalent to loss [RFC8311], the exact response for each controller the response may vary. Markings can be treated as equivalent to loss
could be different. Failure to correctly respond to information ([RFC3168]), but other responses can be specified, such as
about ECN markings is therefore difficult to detect. ([RFC8511]) or ([RFC8311]).
9. IANA Considerations 9. IANA Considerations
This document has no IANA actions. This document has no IANA actions.
10. References 10. References
10.1. Normative References 10.1. Normative References
[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure [QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", Work in Progress, Internet-Draft, draft-ietf-quic- QUIC", Work in Progress, Internet-Draft, draft-ietf-quic-
tls-29, 10 June 2020, tls-30, September 10, 2020,
<https://tools.ietf.org/html/draft-ietf-quic-tls-29>. <https://tools.ietf.org/html/draft-ietf-quic-tls-30>.
[QUIC-TRANSPORT] [QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", Work in Progress, Multiplexed and Secure Transport", Work in Progress,
Internet-Draft, draft-ietf-quic-transport-29, 10 June Internet-Draft, draft-ietf-quic-transport-30, September
2020, <https://tools.ietf.org/html/draft-ietf-quic- 10, 2020, <https://tools.ietf.org/html/draft-ietf-quic-
transport-29>. transport-30>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage [RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>. March 2017, <https://www.rfc-editor.org/info/rfc8085>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>. May 2017, <https://www.rfc-editor.org/info/rfc8174>.
10.2. Informative References 10.2. Informative References
[FACK] Mathis, M. and J. Mahdavi, "Forward Acknowledgement: [FACK] Mathis, M. and J. Mahdavi, "Forward Acknowledgement:
Refining TCP Congestion Control", ACM SIGCOMM , August Refining TCP Congestion Control", ACM SIGCOMM , August
1996. 1996.
[RACK] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "RACK: [PRR] Mathis, M., Dukkipati, N., and Y. Cheng, "Proportional
a time-based fast loss detection algorithm for TCP", Work Rate Reduction for TCP", RFC 6937, DOI 10.17487/RFC6937,
in Progress, Internet-Draft, draft-ietf-tcpm-rack-08, 9 May 2013, <https://www.rfc-editor.org/info/rfc6937>.
March 2020, <http://www.ietf.org/internet-drafts/draft-
ietf-tcpm-rack-08.txt>. [RACK] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The
RACK-TLP loss detection algorithm for TCP", Work in
Progress, Internet-Draft, draft-ietf-tcpm-rack-10, August
22, 2020, <http://www.ietf.org/internet-drafts/draft-ietf-
tcpm-rack-10.txt>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001, RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>. <https://www.rfc-editor.org/info/rfc3168>.
[RFC4653] Bhandarkar, S., Reddy, A. L. N., Allman, M., and E. [RFC3465] Allman, M., "TCP Congestion Control with Appropriate Byte
Blanton, "Improving the Robustness of TCP to Non- Counting (ABC)", RFC 3465, DOI 10.17487/RFC3465, February
Congestion Events", RFC 4653, DOI 10.17487/RFC4653, August 2003, <https://www.rfc-editor.org/info/rfc3465>.
2006, <https://www.rfc-editor.org/info/rfc4653>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>. <https://www.rfc-editor.org/info/rfc5681>.
[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata, [RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
"Forward RTO-Recovery (F-RTO): An Algorithm for Detecting "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
Spurious Retransmission Timeouts with TCP", RFC 5682, Spurious Retransmission Timeouts with TCP", RFC 5682,
DOI 10.17487/RFC5682, September 2009, DOI 10.17487/RFC5682, September 2009,
<https://www.rfc-editor.org/info/rfc5682>. <https://www.rfc-editor.org/info/rfc5682>.
skipping to change at page 27, line 20 skipping to change at page 30, line 36
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion [RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion
Notification (ECN) Experimentation", RFC 8311, Notification (ECN) Experimentation", RFC 8311,
DOI 10.17487/RFC8311, January 2018, DOI 10.17487/RFC8311, January 2018,
<https://www.rfc-editor.org/info/rfc8311>. <https://www.rfc-editor.org/info/rfc8311>.
[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and [RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks", R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
RFC 8312, DOI 10.17487/RFC8312, February 2018, RFC 8312, DOI 10.17487/RFC8312, February 2018,
<https://www.rfc-editor.org/info/rfc8312>. <https://www.rfc-editor.org/info/rfc8312>.
[RFC8511] Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", RFC 8511,
DOI 10.17487/RFC8511, December 2018,
<https://www.rfc-editor.org/info/rfc8511>.
Appendix A. Loss Recovery Pseudocode Appendix A. Loss Recovery Pseudocode
We now describe an example implementation of the loss detection We now describe an example implementation of the loss detection
mechanisms described in Section 6. mechanisms described in Section 6.
A.1. Tracking Sent Packets A.1. Tracking Sent Packets
To correctly implement congestion control, a QUIC sender tracks every To correctly implement congestion control, a QUIC sender tracks every
ack-eliciting packet until the packet is acknowledged or lost. It is ack-eliciting packet until the packet is acknowledged or lost. It is
expected that implementations will be able to access this information expected that implementations will be able to access this information
by packet number and crypto context and store the per-packet fields by packet number and crypto context and store the per-packet fields
(Appendix A.1.1) for loss recovery and congestion control. (Appendix A.1.1) for loss recovery and congestion control.
After a packet is declared lost, the endpoint can track it for an After a packet is declared lost, the endpoint can still maintain
amount of time comparable to the maximum expected packet reordering, state for it for an amount of time to allow for packet reordering;
such as 1 RTT. This allows for detection of spurious see Section 13.3 of [QUIC-TRANSPORT]. This enables a sender to
retransmissions. detect spurious retransmissions.
Sent packets are tracked for each packet number space, and ACK Sent packets are tracked for each packet number space, and ACK
processing only applies to a single space. processing only applies to a single space.
A.1.1. Sent Packet Fields A.1.1. Sent Packet Fields
packet_number: The packet number of the sent packet. packet_number: The packet number of the sent packet.
ack_eliciting: A boolean that indicates whether a packet is ack- ack_eliciting: A boolean that indicates whether a packet is ack-
eliciting. If true, it is expected that an acknowledgement will eliciting. If true, it is expected that an acknowledgement will
be received, though the peer could delay sending the ACK frame be received, though the peer could delay sending the ACK frame
containing it by up to the MaxAckDelay. containing it by up to the max_ack_delay.
in_flight: A boolean that indicates whether the packet counts in_flight: A boolean that indicates whether the packet counts
towards bytes in flight. towards bytes in flight.
sent_bytes: The number of bytes sent in the packet, not including sent_bytes: The number of bytes sent in the packet, not including
UDP or IP overhead, but including QUIC framing overhead. UDP or IP overhead, but including QUIC framing overhead.
time_sent: The time the packet was sent. time_sent: The time the packet was sent.
A.2. Constants of Interest A.2. Constants of Interest
skipping to change at page 28, line 27 skipping to change at page 31, line 47
recommended in Section 6.1.1 is 3. recommended in Section 6.1.1 is 3.
kTimeThreshold: Maximum reordering in time before time threshold kTimeThreshold: Maximum reordering in time before time threshold
loss detection considers a packet lost. Specified as an RTT loss detection considers a packet lost. Specified as an RTT
multiplier. The value recommended in Section 6.1.2 is 9/8. multiplier. The value recommended in Section 6.1.2 is 9/8.
kGranularity: Timer granularity. This is a system-dependent value, kGranularity: Timer granularity. This is a system-dependent value,
and Section 6.1.2 recommends a value of 1ms. and Section 6.1.2 recommends a value of 1ms.
kInitialRtt: The RTT used before an RTT sample is taken. The value kInitialRtt: The RTT used before an RTT sample is taken. The value
recommended in Section 6.2.2 is 500ms. recommended in Section 6.2.2 is 333ms.
kPacketNumberSpace: An enum to enumerate the three packet number kPacketNumberSpace: An enum to enumerate the three packet number
spaces. spaces.
enum kPacketNumberSpace { enum kPacketNumberSpace {
Initial, Initial,
Handshake, Handshake,
ApplicationData, ApplicationData,
} }
A.3. Variables of interest A.3. Variables of interest
Variables required to implement the congestion control mechanisms are Variables required to implement the congestion control mechanisms are
described in this section. described in this section.
latest_rtt: The most recent RTT measurement made when receiving an latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet. ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as smoothed_rtt: The smoothed RTT of the connection, computed as
described in Section 5.3. described in Section 5.3.
rttvar: The RTT variation, computed as described in Section 5.3. rttvar: The RTT variation, computed as described in Section 5.3.
min_rtt: The minimum RTT seen in the connection, ignoring ack delay, min_rtt: The minimum RTT seen in the connection, ignoring
as described in Section 5.2. acknowledgment delay, as described in Section 5.2.
max_ack_delay: The maximum amount of time by which the receiver max_ack_delay: The maximum amount of time by which the receiver
intends to delay acknowledgments for packets in the intends to delay acknowledgments for packets in the Application
ApplicationData packet number space. The actual ack_delay in a Data packet number space, as defined by the eponymous transport
received ACK frame may be larger due to late timers, reordering, parameter (Section 18.2 of [QUIC-TRANSPORT]). Note that the
or lost ACK frames. actual ack_delay in a received ACK frame may be larger due to late
timers, reordering, or loss.
loss_detection_timer: Multi-modal timer used for loss detection. loss_detection_timer: Multi-modal timer used for loss detection.
pto_count: The number of times a PTO has been sent without receiving pto_count: The number of times a PTO has been sent without receiving
an ack. an ack.
time_of_last_ack_eliciting_packet[kPacketNumberSpace]: The time the time_of_last_ack_eliciting_packet[kPacketNumberSpace]: The time the
most recent ack-eliciting packet was sent. most recent ack-eliciting packet was sent.
largest_acked_packet[kPacketNumberSpace]: The largest packet number largest_acked_packet[kPacketNumberSpace]: The largest packet number
skipping to change at page 29, line 35 skipping to change at page 33, line 10
sent_packets[kPacketNumberSpace]: An association of packet numbers sent_packets[kPacketNumberSpace]: An association of packet numbers
in a packet number space to information about them. Described in in a packet number space to information about them. Described in
detail above in Appendix A.1. detail above in Appendix A.1.
A.4. Initialization A.4. Initialization
At the beginning of the connection, initialize the loss detection At the beginning of the connection, initialize the loss detection
variables as follows: variables as follows:
loss_detection_timer.reset() loss_detection_timer.reset()
pto_count = 0 pto_count = 0
latest_rtt = 0 latest_rtt = 0
smoothed_rtt = initial_rtt smoothed_rtt = kInitialRtt
rttvar = initial_rtt / 2 rttvar = kInitialRtt / 2
min_rtt = 0 min_rtt = 0
max_ack_delay = 0 for pn_space in [ Initial, Handshake, ApplicationData ]:
for pn_space in [ Initial, Handshake, ApplicationData ]: largest_acked_packet[pn_space] = infinite
largest_acked_packet[pn_space] = infinite time_of_last_ack_eliciting_packet[pn_space] = 0
time_of_last_ack_eliciting_packet[pn_space] = 0 loss_time[pn_space] = 0
loss_time[pn_space] = 0
A.5. On Sending a Packet A.5. On Sending a Packet
After a packet is sent, information about the packet is stored. The After a packet is sent, information about the packet is stored. The
parameters to OnPacketSent are described in detail above in parameters to OnPacketSent are described in detail above in
Appendix A.1.1. Appendix A.1.1.
Pseudocode for OnPacketSent follows: Pseudocode for OnPacketSent follows:
OnPacketSent(packet_number, pn_space, ack_eliciting, OnPacketSent(packet_number, pn_space, ack_eliciting,
in_flight, sent_bytes): in_flight, sent_bytes):
sent_packets[pn_space][packet_number].packet_number = sent_packets[pn_space][packet_number].packet_number =
packet_number packet_number
sent_packets[pn_space][packet_number].time_sent = now() sent_packets[pn_space][packet_number].time_sent = now()
sent_packets[pn_space][packet_number].ack_eliciting = sent_packets[pn_space][packet_number].ack_eliciting =
ack_eliciting ack_eliciting
sent_packets[pn_space][packet_number].in_flight = in_flight sent_packets[pn_space][packet_number].in_flight = in_flight
if (in_flight): if (in_flight):
if (ack_eliciting): if (ack_eliciting):
time_of_last_ack_eliciting_packet[pn_space] = now() time_of_last_ack_eliciting_packet[pn_space] = now()
OnPacketSentCC(sent_bytes) OnPacketSentCC(sent_bytes)
sent_packets[pn_space][packet_number].size = sent_bytes sent_packets[pn_space][packet_number].sent_bytes =
SetLossDetectionTimer() sent_bytes
SetLossDetectionTimer()
A.6. On Receiving a Datagram A.6. On Receiving a Datagram
When a server is blocked by anti-amplification limits, receiving a When a server is blocked by anti-amplification limits, receiving a
datagram unblocks it, even if none of the packets in the datagram are datagram unblocks it, even if none of the packets in the datagram are
successfully processed. In such a case, the PTO timer will need to successfully processed. In such a case, the PTO timer will need to
be re-armed. be re-armed.
Pseudocode for OnDatagramReceived follows: Pseudocode for OnDatagramReceived follows:
skipping to change at page 30, line 44 skipping to change at page 34, line 18
if (server was at anti-amplification limit): if (server was at anti-amplification limit):
SetLossDetectionTimer() SetLossDetectionTimer()
A.7. On Receiving an Acknowledgment A.7. On Receiving an Acknowledgment
When an ACK frame is received, it may newly acknowledge any number of When an ACK frame is received, it may newly acknowledge any number of
packets. packets.
Pseudocode for OnAckReceived and UpdateRtt follow: Pseudocode for OnAckReceived and UpdateRtt follow:
IncludesAckEliciting(packets):
for packet in packets:
if (packet.ack_eliciting):
return true
return false
OnAckReceived(ack, pn_space): OnAckReceived(ack, pn_space):
if (largest_acked_packet[pn_space] == infinite): if (largest_acked_packet[pn_space] == infinite):
largest_acked_packet[pn_space] = ack.largest_acked largest_acked_packet[pn_space] = ack.largest_acked
else: else:
largest_acked_packet[pn_space] = largest_acked_packet[pn_space] =
max(largest_acked_packet[pn_space], ack.largest_acked) max(largest_acked_packet[pn_space], ack.largest_acked)
// DetectNewlyAckedPackets finds packets that are newly // DetectAndRemoveAckedPackets finds packets that are newly
// acknowledged and removes them from sent_packets. // acknowledged and removes them from sent_packets.
newly_acked_packets = newly_acked_packets =
DetectAndRemoveAckedPackets(ack, pn_space) DetectAndRemoveAckedPackets(ack, pn_space)
// Nothing to do if there are no newly acked packets. // Nothing to do if there are no newly acked packets.
if (newly_acked_packets.empty()): if (newly_acked_packets.empty()):
return return
// If the largest acknowledged is newly acked and // Update the RTT if the largest acknowledged is newly acked
// at least one ack-eliciting was newly acked, update the RTT. // and at least one ack-eliciting was newly acked.
if (newly_acked_packets.largest().packet_number == if (newly_acked_packets.largest().packet_number ==
ack.largest_acked && ack.largest_acked &&
IncludesAckEliciting(newly_acked_packets)): IncludesAckEliciting(newly_acked_packets)):
latest_rtt = latest_rtt =
now - sent_packets[pn_space][ack.largest_acked].time_sent now() - newly_acked_packets.largest().time_sent
ack_delay = 0 UpdateRtt(ack.ack_delay)
if (pn_space == ApplicationData):
ack_delay = ack.ack_delay
UpdateRtt(ack_delay)
// Process ECN information if present. // Process ECN information if present.
if (ACK frame contains ECN information): if (ACK frame contains ECN information):
ProcessECN(ack, pn_space) ProcessECN(ack, pn_space)
lost_packets = DetectAndRemoveLostPackets(pn_space) lost_packets = DetectAndRemoveLostPackets(pn_space)
if (!lost_packets.empty()): if (!lost_packets.empty()):
OnPacketsLost(lost_packets) OnPacketsLost(lost_packets)
OnPacketsAcked(newly_acked_packets) OnPacketsAcked(newly_acked_packets)
skipping to change at page 31, line 45 skipping to change at page 35, line 21
pto_count = 0 pto_count = 0
SetLossDetectionTimer() SetLossDetectionTimer()
UpdateRtt(ack_delay): UpdateRtt(ack_delay):
if (is first RTT sample): if (is first RTT sample):
min_rtt = latest_rtt min_rtt = latest_rtt
smoothed_rtt = latest_rtt smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2 rttvar = latest_rtt / 2
return return
// min_rtt ignores ack delay. // min_rtt ignores acknowledgment delay.
min_rtt = min(min_rtt, latest_rtt) min_rtt = min(min_rtt, latest_rtt)
// Limit ack_delay by max_ack_delay // Limit ack_delay by max_ack_delay after handshake
ack_delay = min(ack_delay, max_ack_delay) // confirmation. Note that ack_delay is 0 for
// Adjust for ack delay if plausible. // acknowledgements of Initial and Handshake packets.
if (handshake confirmed):
ack_delay = min(ack_delay, max_ack_delay)
// Adjust for acknowledgment delay if plausible.
adjusted_rtt = latest_rtt adjusted_rtt = latest_rtt
if (latest_rtt > min_rtt + ack_delay): if (latest_rtt > min_rtt + ack_delay):
adjusted_rtt = latest_rtt - ack_delay adjusted_rtt = latest_rtt - ack_delay
rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt) rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt)
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
A.8. Setting the Loss Detection Timer A.8. Setting the Loss Detection Timer
QUIC loss detection uses a single timer for all timeout loss QUIC loss detection uses a single timer for all timeout loss
detection. The duration of the timer is based on the timer's mode, 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 which is set in the packet and timer events further below. The
skipping to change at page 32, line 49 skipping to change at page 36, line 30
if (has handshake keys): if (has handshake keys):
return (now() + duration), Handshake return (now() + duration), Handshake
else: else:
return (now() + duration), Initial return (now() + duration), Initial
pto_timeout = infinite pto_timeout = infinite
pto_space = Initial pto_space = Initial
for space in [ Initial, Handshake, ApplicationData ]: for space in [ Initial, Handshake, ApplicationData ]:
if (no in-flight packets in space): if (no in-flight packets in space):
continue; continue;
if (space == ApplicationData): if (space == ApplicationData):
// Skip ApplicationData until handshake complete. // Skip Application Data until handshake confirmed.
if (handshake is not complete): if (handshake is not confirmed):
return pto_timeout, pto_space return pto_timeout, pto_space
// Include max_ack_delay and backoff for ApplicationData. // Include max_ack_delay and backoff for Application Data.
duration += max_ack_delay * (2 ^ pto_count) duration += max_ack_delay * (2 ^ pto_count)
t = time_of_last_ack_eliciting_packet[space] + duration t = time_of_last_ack_eliciting_packet[space] + duration
if (t < pto_timeout): if (t < pto_timeout):
pto_timeout = t pto_timeout = t
pto_space = space pto_space = space
return pto_timeout, pto_space return pto_timeout, pto_space
PeerCompletedAddressValidation(): PeerCompletedAddressValidation():
# Assume clients validate the server's address implicitly. // Assume clients validate the server's address implicitly.
if (endpoint is server): if (endpoint is server):
return true return true
# Servers complete address validation when a // Servers complete address validation when a
# protected packet is received. // protected packet is received.
return has received Handshake ACK || return has received Handshake ACK ||
has received 1-RTT ACK || handshake confirmed
has received HANDSHAKE_DONE
SetLossDetectionTimer(): SetLossDetectionTimer():
earliest_loss_time, _ = GetLossTimeAndSpace() earliest_loss_time, _ = GetLossTimeAndSpace()
if (earliest_loss_time != 0): if (earliest_loss_time != 0):
// Time threshold loss detection. // Time threshold loss detection.
loss_detection_timer.update(earliest_loss_time) loss_detection_timer.update(earliest_loss_time)
return return
if (server is at anti-amplification limit): if (server is at anti-amplification limit):
// The server's timer is not set if nothing can be sent. // The server's timer is not set if nothing can be sent.
skipping to change at page 34, line 9 skipping to change at page 38, line 9
When the loss detection timer expires, the timer's mode determines When the loss detection timer expires, the timer's mode determines
the action to be performed. the action to be performed.
Pseudocode for OnLossDetectionTimeout follows: Pseudocode for OnLossDetectionTimeout follows:
OnLossDetectionTimeout(): OnLossDetectionTimeout():
earliest_loss_time, pn_space = GetLossTimeAndSpace() earliest_loss_time, pn_space = GetLossTimeAndSpace()
if (earliest_loss_time != 0): if (earliest_loss_time != 0):
// Time threshold loss Detection // Time threshold loss Detection
lost_packets = DetectLostPackets(pn_space) lost_packets = DetectAndRemoveLostPackets(pn_space)
assert(!lost_packets.empty()) assert(!lost_packets.empty())
OnPacketsLost(lost_packets) OnPacketsLost(lost_packets)
SetLossDetectionTimer() SetLossDetectionTimer()
return return
if (bytes_in_flight > 0): if (bytes_in_flight > 0):
// PTO. Send new data if available, else retransmit old data. // PTO. Send new data if available, else retransmit old data.
// If neither is available, send a single PING frame. // If neither is available, send a single PING frame.
_, pn_space = GetPtoTimeAndSpace() _, pn_space = GetPtoTimeAndSpace()
SendOneOrTwoAckElicitingPackets(pn_space) SendOneOrTwoAckElicitingPackets(pn_space)
else: else:
assert(endpoint is client without 1-RTT keys) assert(!PeerCompletedAddressValidation())
// Client sends an anti-deadlock packet: Initial is padded // Client sends an anti-deadlock packet: Initial is padded
// to earn more anti-amplification credit, // to earn more anti-amplification credit,
// a Handshake packet proves address ownership. // a Handshake packet proves address ownership.
if (has Handshake keys): if (has Handshake keys):
SendOneAckElicitingHandshakePacket() SendOneAckElicitingHandshakePacket()
else: else:
SendOneAckElicitingPaddedInitialPacket() SendOneAckElicitingPaddedInitialPacket()
pto_count++ pto_count++
SetLossDetectionTimer() SetLossDetectionTimer()
skipping to change at page 36, line 4 skipping to change at page 40, line 4
Constants used in congestion control are based on a combination of Constants used in congestion control are based on a combination of
RFCs, papers, and common practice. RFCs, papers, and common practice.
kInitialWindow: Default limit on the initial bytes in flight as kInitialWindow: Default limit on the initial bytes in flight as
described in Section 7.2. described in Section 7.2.
kMinimumWindow: Minimum congestion window in bytes as described in kMinimumWindow: Minimum congestion window in bytes as described in
Section 7.2. Section 7.2.
kLossReductionFactor: Reduction in congestion window when a new loss kLossReductionFactor: Reduction in congestion window when a new loss
event is detected. The Section 7 section recommends a value is event is detected. Section 7 recommends a value is 0.5.
0.5.
kPersistentCongestionThreshold: Period of time for persistent kPersistentCongestionThreshold: Period of time for persistent
congestion to be established, specified as a PTO multiplier. The congestion to be established, specified as a PTO multiplier.
Section 7.8 section recommends a value of 3. Section 7.6 recommends a value of 3.
B.2. Variables of interest B.2. Variables of interest
Variables required to implement the congestion control mechanisms are Variables required to implement the congestion control mechanisms are
described in this section. described in this section.
max_datagram_size: The sender's current maximum payload size. Does max_datagram_size: The sender's current maximum payload size. Does
not include UDP or IP overhead. The max datagram size is used for not include UDP or IP overhead. The max datagram size is used for
congestion window computations. An endpoint sets the value of congestion window computations. An endpoint sets the value of
this variable based on its PMTU (see Section 14.1 of this variable based on its PMTU (see Section 14.1 of
[QUIC-TRANSPORT]), with a minimum value of 1200 bytes. [QUIC-TRANSPORT]), with a minimum value of 1200 bytes.
ecn_ce_counters[kPacketNumberSpace]: The highest value reported for ecn_ce_counters[kPacketNumberSpace]: The highest value reported for
the ECN-CE counter in the packet number space by the peer in an the ECN-CE counter in the packet number space by the peer in an
ACK frame. This value is used to detect increases in the reported ACK frame. This value is used to detect increases in the reported
ECN-CE counter. ECN-CE counter.
bytes_in_flight: The sum of the size in bytes of all sent packets bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one ack-eliciting or PADDING frame, and have that contain at least one ack-eliciting or PADDING frame, and have
not been acked or declared lost. The size does not include IP or not been acknowledged or declared lost. The size does not include
UDP overhead, but does include the QUIC header and AEAD overhead. IP or UDP overhead, but does include the QUIC header and AEAD
Packets only containing ACK frames do not count towards overhead. Packets only containing ACK frames do not count towards
bytes_in_flight to ensure congestion control does not impede bytes_in_flight to ensure congestion control does not impede
congestion feedback. congestion feedback.
congestion_window: Maximum number of bytes-in-flight that may be congestion_window: Maximum number of bytes-in-flight that may be
sent. sent.
congestion_recovery_start_time: The time when QUIC first detects congestion_recovery_start_time: The time when QUIC first detects
congestion due to loss or ECN, causing it to enter congestion congestion due to loss or ECN, causing it to enter congestion
recovery. When a packet sent after this time is acknowledged, recovery. When a packet sent after this time is acknowledged,
QUIC exits congestion recovery. QUIC exits congestion recovery.
ssthresh: Slow start threshold in bytes. When the congestion window ssthresh: Slow start threshold in bytes. When the congestion window
is below ssthresh, the mode is slow start and the window grows by is below ssthresh, the mode is slow start and the window grows by
the number of bytes acknowledged. the number of bytes acknowledged.
first_rtt_sample: The time that the first RTT sample was obtained.
B.3. Initialization B.3. Initialization
At the beginning of the connection, initialize the congestion control At the beginning of the connection, initialize the congestion control
variables as follows: variables as follows:
congestion_window = kInitialWindow congestion_window = kInitialWindow
bytes_in_flight = 0 bytes_in_flight = 0
congestion_recovery_start_time = 0 congestion_recovery_start_time = 0
ssthresh = infinite ssthresh = infinite
for pn_space in [ Initial, Handshake, ApplicationData ]: first_rtt_sample = 0
ecn_ce_counters[pn_space] = 0 for pn_space in [ Initial, Handshake, ApplicationData ]:
ecn_ce_counters[pn_space] = 0
B.4. On Packet Sent B.4. On Packet Sent
Whenever a packet is sent, and it contains non-ACK frames, the packet Whenever a packet is sent, and it contains non-ACK frames, the packet
increases bytes_in_flight. increases bytes_in_flight.
OnPacketSentCC(bytes_sent): OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent bytes_in_flight += bytes_sent
B.5. On Packet Acknowledgement B.5. On Packet Acknowledgement
Invoked from loss detection's OnAckReceived and is supplied with the Invoked from loss detection's OnAckReceived and is supplied with the
newly acked_packets from sent_packets. newly acked_packets from sent_packets.
InCongestionRecovery(sent_time): In congestion avoidance, implementers that use an integer
return sent_time <= congestion_recovery_start_time representation for congestion_window should be careful with division,
and can use the alternative approach suggested in Section 2.1 of
[RFC3465].
OnPacketsAcked(acked_packets): InCongestionRecovery(sent_time):
for (packet in acked_packets): return sent_time <= congestion_recovery_start_time
// Remove from bytes_in_flight.
bytes_in_flight -= packet.size OnPacketsAcked(acked_packets):
if (InCongestionRecovery(packet.time_sent)): if (first_rtt_sample == 0):
// Do not increase congestion window in recovery period. first_rtt_sample = now()
return
if (IsAppOrFlowControlLimited()): for acked_packet in acked_packets:
// Do not increase congestion_window if application OnPacketAcked(acked_packet)
// limited or flow control limited.
return OnPacketAcked(acked_packet):
if (congestion_window < ssthresh): // Remove from bytes_in_flight.
// Slow start. bytes_in_flight -= acked_packet.sent_bytes
congestion_window += packet.size // Do not increase congestion_window if application
return // limited or flow control limited.
// Congestion avoidance. if (IsAppOrFlowControlLimited())
congestion_window += max_datagram_size * acked_packet.size return
/ congestion_window // Do not increase congestion window in recovery period.
if (InCongestionRecovery(acked_packet.time_sent)):
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packet.sent_bytes
else:
// Congestion avoidance.
congestion_window +=
max_datagram_size * acked_packet.sent_bytes
/ congestion_window
B.6. On New Congestion Event B.6. On New Congestion Event
Invoked from ProcessECN and OnPacketsLost when a new congestion event Invoked from ProcessECN and OnPacketsLost when a new congestion event
is detected. May start a new recovery period and reduces the is detected. If not already in recovery, this starts a recovery
congestion window. period and reduces the slow start threshold and congestion window
immediately.
CongestionEvent(sent_time): OnCongestionEvent(sent_time):
// Start a new congestion event if packet was sent after the // No reaction if already in a recovery period.
// start of the previous congestion recovery period. if (InCongestionRecovery(sent_time)):
if (!InCongestionRecovery(sent_time)): return
congestion_recovery_start_time = now()
congestion_window *= kLossReductionFactor // Enter recovery period.
congestion_window = max(congestion_window, kMinimumWindow) congestion_recovery_start_time = now()
ssthresh = congestion_window ssthresh = congestion_window * kLossReductionFactor
// A packet can be sent to speed up loss recovery. congestion_window = max(ssthresh, kMinimumWindow)
MaybeSendOnePacket() // A packet can be sent to speed up loss recovery.
MaybeSendOnePacket()
B.7. Process ECN Information B.7. Process ECN Information
Invoked when an ACK frame with an ECN section is received from the Invoked when an ACK frame with an ECN section is received from the
peer. peer.
ProcessECN(ack, pn_space): ProcessECN(ack, pn_space):
// If the ECN-CE counter reported by the peer has increased, // If the ECN-CE counter reported by the peer has increased,
// this could be a new congestion event. // this could be a new congestion event.
if (ack.ce_counter > ecn_ce_counters[pn_space]): if (ack.ce_counter > ecn_ce_counters[pn_space]):
ecn_ce_counters[pn_space] = ack.ce_counter ecn_ce_counters[pn_space] = ack.ce_counter
CongestionEvent(sent_packets[ack.largest_acked].time_sent) sent_time = sent_packets[ack.largest_acked].time_sent
OnCongestionEvent(sent_time)
B.8. On Packets Lost B.8. On Packets Lost
Invoked from DetectLostPackets when packets are deemed lost. Invoked when DetectAndRemoveLostPackets deems packets lost.
InPersistentCongestion(lost_packets):
pto = smoothed_rtt + max(4 * rttvar, kGranularity) +
max_ack_delay
congestion_period = pto * kPersistentCongestionThreshold
// Determine if all packets in the time period before the
// largest newly lost packet, including the edges, are
// marked lost
return AreAllPacketsLost(lost_packets, congestion_period)
OnPacketsLost(lost_packets): OnPacketsLost(lost_packets):
// Remove lost packets from bytes_in_flight. // Remove lost packets from bytes_in_flight.
for (lost_packet : lost_packets): for lost_packet in lost_packets:
bytes_in_flight -= lost_packet.size bytes_in_flight -= lost_packet.sent_bytes
CongestionEvent(lost_packets.largest().time_sent) OnCongestionEvent(lost_packets.largest().time_sent)
// Collapse congestion window if persistent congestion // Reset the congestion window if the loss of these
if (InPersistentCongestion(lost_packets)): // packets indicates persistent congestion.
congestion_window = kMinimumWindow // Only consider packets sent after getting an RTT sample.
assert(first_rtt_sample != 0)
pc_lost = {}
for lost in lost_packets:
if lost.time_sent > first_rtt_sample:
pc_lost.insert(lost)
if (InPersistentCongestion(pc_lost)):
congestion_window = kMinimumWindow
congestion_recovery_start_time = 0
B.9. Upon dropping Initial or Handshake keys B.9. Upon dropping Initial or Handshake keys
When Initial or Handshake keys are discarded, packets from the space When Initial or Handshake keys are discarded, packets from the space
are discarded and loss detection state is updated. are discarded and loss detection state is updated.
Pseudocode for OnPacketNumberSpaceDiscarded follows: Pseudocode for OnPacketNumberSpaceDiscarded follows:
OnPacketNumberSpaceDiscarded(pn_space): OnPacketNumberSpaceDiscarded(pn_space):
assert(pn_space != ApplicationData) assert(pn_space != ApplicationData)
skipping to change at page 39, line 32 skipping to change at page 44, line 25
pto_count = 0 pto_count = 0
SetLossDetectionTimer() SetLossDetectionTimer()
Appendix C. Change Log Appendix C. Change Log
*RFC Editor's Note:* Please remove this section prior to *RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document. publication of a final version of this document.
Issue and pull request numbers are listed with a leading octothorp. Issue and pull request numbers are listed with a leading octothorp.
C.1. Since draft-ietf-quic-recovery-28 C.1. Since draft-ietf-quic-recovery-29
* Allow caching of packets that can't be decrypted, by allowing the
reported acknowledgment delay to exceed max_ack_delay prior to
confirming the handshake (#3821, #3980, #4035, #3874)
* Persistent congestion cannot include packets sent before the first
RTT sample for the path (#3875, #3889)
* Recommend reset of min_rtt in persistent congestion (#3927, #3975)
* Persistent congestion is independent of packet number space
(#3939, #3961)
* Only limit bursts to the initial window without information about
the path (#3892, #3936)
* Add normative requirements for increasing and reducing the
congestion window (#3944, #3978, #3997, #3998)
C.2. Since draft-ietf-quic-recovery-28
* Refactored pseudocode to correct PTO calculation (#3564, #3674, * Refactored pseudocode to correct PTO calculation (#3564, #3674,
#3681) #3681)
C.2. Since draft-ietf-quic-recovery-27 C.3. Since draft-ietf-quic-recovery-27
* Added recommendations for speeding up handshake under some loss * Added recommendations for speeding up handshake under some loss
conditions (#3078, #3080) conditions (#3078, #3080)
* PTO count is reset when handshake progress is made (#3272, #3415) * PTO count is reset when handshake progress is made (#3272, #3415)
* PTO count is not reset by a client when the server might be * PTO count is not reset by a client when the server might be
awaiting address validation (#3546, #3551) awaiting address validation (#3546, #3551)
* Recommend repairing losses immediately after entering the recovery * Recommend repairing losses immediately after entering the recovery
period (#3335, #3443) period (#3335, #3443)
* Clarified what loss conditions can be ignored during the handshake * Clarified what loss conditions can be ignored during the handshake
(#3456, #3450) (#3456, #3450)
* Allow, but don't recommend, using RTT from previous connection to * Allow, but don't recommend, using RTT from previous connection to
seed RTT (#3464, #3496) seed RTT (#3464, #3496)
* Recommend use of adaptive loss detection thresholds (#3571, #3572) * Recommend use of adaptive loss detection thresholds (#3571, #3572)
C.3. Since draft-ietf-quic-recovery-26 C.4. Since draft-ietf-quic-recovery-26
No changes. No changes.
C.4. Since draft-ietf-quic-recovery-25 C.5. Since draft-ietf-quic-recovery-25
No significant changes. No significant changes.
C.5. Since draft-ietf-quic-recovery-24 C.6. Since draft-ietf-quic-recovery-24
* Require congestion control of some sort (#3247, #3244, #3248) * Require congestion control of some sort (#3247, #3244, #3248)
* Set a minimum reordering threshold (#3256, #3240) * Set a minimum reordering threshold (#3256, #3240)
* PTO is specific to a packet number space (#3067, #3074, #3066) * PTO is specific to a packet number space (#3067, #3074, #3066)
C.6. Since draft-ietf-quic-recovery-23 C.7. Since draft-ietf-quic-recovery-23
* Define under-utilizing the congestion window (#2630, #2686, #2675) * Define under-utilizing the congestion window (#2630, #2686, #2675)
* PTO MUST send data if possible (#3056, #3057) * PTO MUST send data if possible (#3056, #3057)
* Connection Close is not ack-eliciting (#3097, #3098) * Connection Close is not ack-eliciting (#3097, #3098)
* MUST limit bursts to the initial congestion window (#3160) * MUST limit bursts to the initial congestion window (#3160)
* Define the current max_datagram_size for congestion control * Define the current max_datagram_size for congestion control
(#3041, #3167) (#3041, #3167)
C.7. Since draft-ietf-quic-recovery-22 C.8. Since draft-ietf-quic-recovery-22
* PTO should always send an ack-eliciting packet (#2895) * PTO should always send an ack-eliciting packet (#2895)
* Unify the Handshake Timer with the PTO timer (#2648, #2658, #2886) * Unify the Handshake Timer with the PTO timer (#2648, #2658, #2886)
* Move ACK generation text to transport draft (#1860, #2916) * Move ACK generation text to transport draft (#1860, #2916)
C.8. Since draft-ietf-quic-recovery-21 C.9. Since draft-ietf-quic-recovery-21
* No changes * No changes
C.9. Since draft-ietf-quic-recovery-20 C.10. Since draft-ietf-quic-recovery-20
* Path validation can be used as initial RTT value (#2644, #2687) * Path validation can be used as initial RTT value (#2644, #2687)
* max_ack_delay transport parameter defaults to 0 (#2638, #2646) * max_ack_delay transport parameter defaults to 0 (#2638, #2646)
* Ack Delay only measures intentional delays induced by the * ACK delay only measures intentional delays induced by the
implementation (#2596, #2786) implementation (#2596, #2786)
C.10. Since draft-ietf-quic-recovery-19 C.11. Since draft-ietf-quic-recovery-19
* Change kPersistentThreshold from an exponent to a multiplier * Change kPersistentThreshold from an exponent to a multiplier
(#2557) (#2557)
* Send a PING if the PTO timer fires and there's nothing to send * Send a PING if the PTO timer fires and there's nothing to send
(#2624) (#2624)
* Set loss delay to at least kGranularity (#2617) * Set loss delay to at least kGranularity (#2617)
* Merge application limited and sending after idle sections. Always * Merge application limited and sending after idle sections. Always
skipping to change at page 41, line 36 skipping to change at page 46, line 51
packet is ack-eliciting but the largest_acked is not (#2592) packet is ack-eliciting but the largest_acked is not (#2592)
* Don't arm the handshake timer if there is no handshake data * Don't arm the handshake timer if there is no handshake data
(#2590) (#2590)
* Clarify that the time threshold loss alarm takes precedence over * Clarify that the time threshold loss alarm takes precedence over
the crypto handshake timer (#2590, #2620) the crypto handshake timer (#2590, #2620)
* Change initial RTT to 500ms to align with RFC6298 (#2184) * Change initial RTT to 500ms to align with RFC6298 (#2184)
C.11. Since draft-ietf-quic-recovery-18 C.12. Since draft-ietf-quic-recovery-18
* Change IW byte limit to 14720 from 14600 (#2494) * Change IW byte limit to 14720 from 14600 (#2494)
* Update PTO calculation to match RFC6298 (#2480, #2489, #2490) * Update PTO calculation to match RFC6298 (#2480, #2489, #2490)
* Improve loss detection's description of multiple packet number * Improve loss detection's description of multiple packet number
spaces and pseudocode (#2485, #2451, #2417) spaces and pseudocode (#2485, #2451, #2417)
* Declare persistent congestion even if non-probe packets are sent * Declare persistent congestion even if non-probe packets are sent
and don't make persistent congestion more aggressive than RTO and don't make persistent congestion more aggressive than RTO
verified was (#2365, #2244) verified was (#2365, #2244)
* Move pseudocode to the appendices (#2408) * Move pseudocode to the appendices (#2408)
* What to send on multiple PTOs (#2380) * What to send on multiple PTOs (#2380)
C.12. Since draft-ietf-quic-recovery-17 C.13. Since draft-ietf-quic-recovery-17
* After Probe Timeout discard in-flight packets or send another * After Probe Timeout discard in-flight packets or send another
(#2212, #1965) (#2212, #1965)
* Endpoints discard initial keys as soon as handshake keys are * Endpoints discard initial keys as soon as handshake keys are
available (#1951, #2045) available (#1951, #2045)
* 0-RTT state is discarded when 0-RTT is rejected (#2300) * 0-RTT state is discarded when 0-RTT is rejected (#2300)
* Loss detection timer is cancelled when ack-eliciting frames are in * Loss detection timer is cancelled when ack-eliciting frames are in
skipping to change at page 42, line 31 skipping to change at page 47, line 45
controller (#2138, 2187) controller (#2138, 2187)
* Process ECN counts before marking packets lost (#2142) * Process ECN counts before marking packets lost (#2142)
* Mark packets lost before resetting crypto_count and pto_count * Mark packets lost before resetting crypto_count and pto_count
(#2208, #2209) (#2208, #2209)
* Congestion and loss recovery state are discarded when keys are * Congestion and loss recovery state are discarded when keys are
discarded (#2327) discarded (#2327)
C.13. Since draft-ietf-quic-recovery-16 C.14. Since draft-ietf-quic-recovery-16
* Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP * Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP
and min crypto timeouts; eliminate timeout validation (#2114, and min crypto timeouts; eliminate timeout validation (#2114,
#2166, #2168, #1017) #2166, #2168, #1017)
* Redefine how congestion avoidance in terms of when the period * Redefine how congestion avoidance in terms of when the period
starts (#1928, #1930) starts (#1928, #1930)
* Document what needs to be tracked for packets that are in flight * Document what needs to be tracked for packets that are in flight
(#765, #1724, #1939) (#765, #1724, #1939)
skipping to change at page 43, line 4 skipping to change at page 48, line 21
* Integrate both time and packet thresholds into loss detection * Integrate both time and packet thresholds into loss detection
(#1969, #1212, #934, #1974) (#1969, #1212, #934, #1974)
* Reduce congestion window after idle, unless pacing is used (#2007, * Reduce congestion window after idle, unless pacing is used (#2007,
#2023) #2023)
* Disable RTT calculation for packets that don't elicit * Disable RTT calculation for packets that don't elicit
acknowledgment (#2060, #2078) acknowledgment (#2060, #2078)
* Limit ack_delay by max_ack_delay (#2060, #2099) * Limit ack_delay by max_ack_delay (#2060, #2099)
* Initial keys are discarded once Handshake keys are available * Initial keys are discarded once Handshake keys are available
(#1951, #2045) (#1951, #2045)
* Reorder ECN and loss detection in pseudocode (#2142) * Reorder ECN and loss detection in pseudocode (#2142)
* Only cancel loss detection timer if ack-eliciting packets are in * Only cancel loss detection timer if ack-eliciting packets are in
flight (#2093, #2117) flight (#2093, #2117)
C.14. Since draft-ietf-quic-recovery-14 C.15. Since draft-ietf-quic-recovery-14
* Used max_ack_delay from transport params (#1796, #1782) * Used max_ack_delay from transport params (#1796, #1782)
* Merge ACK and ACK_ECN (#1783) * Merge ACK and ACK_ECN (#1783)
C.15. Since draft-ietf-quic-recovery-13 C.16. Since draft-ietf-quic-recovery-13
* Corrected the lack of ssthresh reduction in CongestionEvent * Corrected the lack of ssthresh reduction in CongestionEvent
pseudocode (#1598) pseudocode (#1598)
* Considerations for ECN spoofing (#1426, #1626) * Considerations for ECN spoofing (#1426, #1626)
* Clarifications for PADDING and congestion control (#837, #838, * Clarifications for PADDING and congestion control (#837, #838,
#1517, #1531, #1540) #1517, #1531, #1540)
* Reduce early retransmission timer to RTT/8 (#945, #1581) * Reduce early retransmission timer to RTT/8 (#945, #1581)
* Packets are declared lost after an RTO is verified (#935, #1582) * Packets are declared lost after an RTO is verified (#935, #1582)
C.16. Since draft-ietf-quic-recovery-12 C.17. Since draft-ietf-quic-recovery-12
* Changes to manage separate packet number spaces and encryption * Changes to manage separate packet number spaces and encryption
levels (#1190, #1242, #1413, #1450) levels (#1190, #1242, #1413, #1450)
* Added ECN feedback mechanisms and handling; new ACK_ECN frame * Added ECN feedback mechanisms and handling; new ACK_ECN frame
(#804, #805, #1372) (#804, #805, #1372)
C.17. Since draft-ietf-quic-recovery-11 C.18. Since draft-ietf-quic-recovery-11
No significant changes. No significant changes.
C.18. Since draft-ietf-quic-recovery-10 C.19. Since draft-ietf-quic-recovery-10
* Improved text on ack generation (#1139, #1159) * Improved text on ack generation (#1139, #1159)
* Make references to TCP recovery mechanisms informational (#1195) * Make references to TCP recovery mechanisms informational (#1195)
* Define time_of_last_sent_handshake_packet (#1171) * Define time_of_last_sent_handshake_packet (#1171)
* Added signal from TLS the data it includes needs to be sent in a * Added signal from TLS the data it includes needs to be sent in a
Retry packet (#1061, #1199) Retry packet (#1061, #1199)
* Minimum RTT (min_rtt) is initialized with an infinite value * Minimum RTT (min_rtt) is initialized with an infinite value
(#1169) (#1169)
C.19. Since draft-ietf-quic-recovery-09 C.20. Since draft-ietf-quic-recovery-09
No significant changes. No significant changes.
C.20. Since draft-ietf-quic-recovery-08 C.21. Since draft-ietf-quic-recovery-08
* Clarified pacing and RTO (#967, #977) * Clarified pacing and RTO (#967, #977)
C.21. Since draft-ietf-quic-recovery-07 C.22. Since draft-ietf-quic-recovery-07
* Include Ack Delay in RTO(and TLP) computations (#981) * Include ACK delay in RTO(and TLP) computations (#981)
* Ack Delay in SRTT computation (#961) * ACK delay in SRTT computation (#961)
* Default RTT and Slow Start (#590) * Default RTT and Slow Start (#590)
* Many editorial fixes. * Many editorial fixes.
C.22. Since draft-ietf-quic-recovery-06 C.23. Since draft-ietf-quic-recovery-06
No significant changes. No significant changes.
C.23. Since draft-ietf-quic-recovery-05 C.24. Since draft-ietf-quic-recovery-05
* Add more congestion control text (#776) * Add more congestion control text (#776)
C.24. Since draft-ietf-quic-recovery-04 C.25. Since draft-ietf-quic-recovery-04
No significant changes. No significant changes.
C.25. Since draft-ietf-quic-recovery-03 C.26. Since draft-ietf-quic-recovery-03
No significant changes. No significant changes.
C.26. Since draft-ietf-quic-recovery-02 C.27. Since draft-ietf-quic-recovery-02
* Integrate F-RTO (#544, #409) * Integrate F-RTO (#544, #409)
* Add congestion control (#545, #395) * Add congestion control (#545, #395)
* Require connection abort if a skipped packet was acknowledged * Require connection abort if a skipped packet was acknowledged
(#415) (#415)
* Simplify RTO calculations (#142, #417) * Simplify RTO calculations (#142, #417)
C.27. Since draft-ietf-quic-recovery-01 C.28. Since draft-ietf-quic-recovery-01
* Overview added to loss detection * Overview added to loss detection
* Changes initial default RTT to 100ms * Changes initial default RTT to 100ms
* Added time-based loss detection and fixes early retransmit * Added time-based loss detection and fixes early retransmit
* Clarified loss recovery for handshake packets * Clarified loss recovery for handshake packets
* Fixed references and made TCP references informative * Fixed references and made TCP references informative
C.28. Since draft-ietf-quic-recovery-00 C.29. Since draft-ietf-quic-recovery-00
* Improved description of constants and ACK behavior * Improved description of constants and ACK behavior
C.29. Since draft-iyengar-quic-loss-recovery-01 C.30. Since draft-iyengar-quic-loss-recovery-01
* Adopted as base for draft-ietf-quic-recovery * Adopted as base for draft-ietf-quic-recovery
* Updated authors/editors list * Updated authors/editors list
* Added table of contents * Added table of contents
Appendix D. Contributors Appendix D. Contributors
The IETF QUIC Working Group received an enormous amount of support The IETF QUIC Working Group received an enormous amount of support
from many people. The following people provided substantive from many people. The following people provided substantive
contributions to this document: Alessandro Ghedini, Benjamin contributions to this document: Alessandro Ghedini, Benjamin
Saunders, Gorry Fairhurst, 奥 一穂 (Kazuho Oku), Lars Eggert, Magnus Saunders, Gorry Fairhurst, 奥 一穂 (Kazuho Oku), Lars Eggert, Magnus
Westerlund, Marten Seemann, Martin Duke, Martin Thomson, Nick Banks, Westerlund, Marten Seemann, Martin Duke, Martin Thomson, Nick Banks,
Praveen Balasubramaniam. Praveen Balasubramanian.
Acknowledgments Acknowledgments
Authors' Addresses Authors' Addresses
Jana Iyengar (editor) Jana Iyengar (editor)
Fastly Fastly
Email: jri.ietf@gmail.com Email: jri.ietf@gmail.com
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