QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-29
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 9002.
|
|
|---|---|---|---|
| Authors | Jana Iyengar , Ian Swett | ||
| Last updated | 2020-06-09 | ||
| Replaces | draft-iyengar-quic-loss-recovery | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
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| Send notices to | (None) |
draft-ietf-quic-recovery-29
QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track I. Swett, Ed.
Expires: 12 December 2020 Google
10 June 2020
QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-29
Abstract
This document describes loss detection and congestion control
mechanisms for QUIC.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org (mailto:quic@ietf.org)), which is
archived at https://mailarchive.ietf.org/arch/
search/?email_list=quic.
Working Group information can be found at https://github.com/quicwg;
source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/-recovery.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 12 December 2020.
Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document. Code Components
extracted from this document must include Simplified BSD License text
as described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 4
3. Design of the QUIC Transmission Machinery . . . . . . . . . . 5
4. Relevant Differences Between QUIC and TCP . . . . . . . . . . 5
4.1. Separate Packet Number Spaces . . . . . . . . . . . . . . 6
4.2. Monotonically Increasing Packet Numbers . . . . . . . . . 6
4.3. Clearer Loss Epoch . . . . . . . . . . . . . . . . . . . 6
4.4. No Reneging . . . . . . . . . . . . . . . . . . . . . . . 7
4.5. More ACK Ranges . . . . . . . . . . . . . . . . . . . . . 7
4.6. Explicit Correction For Delayed Acknowledgements . . . . 7
4.7. Probe Timeout Replaces RTO and TLP . . . . . . . . . . . 7
4.8. The Minimum Congestion Window is Two Packets . . . . . . 8
5. Estimating the Round-Trip Time . . . . . . . . . . . . . . . 8
5.1. Generating RTT samples . . . . . . . . . . . . . . . . . 8
5.2. Estimating min_rtt . . . . . . . . . . . . . . . . . . . 9
5.3. Estimating smoothed_rtt and rttvar . . . . . . . . . . . 9
6. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 11
6.1. Acknowledgement-based Detection . . . . . . . . . . . . . 11
6.1.1. Packet Threshold . . . . . . . . . . . . . . . . . . 11
6.1.2. Time Threshold . . . . . . . . . . . . . . . . . . . 12
6.2. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 13
6.2.1. Computing PTO . . . . . . . . . . . . . . . . . . . . 13
6.2.2. Handshakes and New Paths . . . . . . . . . . . . . . 14
6.2.3. Speeding Up Handshake Completion . . . . . . . . . . 15
6.2.4. Sending Probe Packets . . . . . . . . . . . . . . . . 16
6.3. Handling Retry Packets . . . . . . . . . . . . . . . . . 17
6.4. Discarding Keys and Packet State . . . . . . . . . . . . 17
7. Congestion Control . . . . . . . . . . . . . . . . . . . . . 18
7.1. Explicit Congestion Notification . . . . . . . . . . . . 19
7.2. Initial and Minimum Congestion Window . . . . . . . . . . 19
7.3. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 19
7.4. Congestion Avoidance . . . . . . . . . . . . . . . . . . 20
7.5. Recovery Period . . . . . . . . . . . . . . . . . . . . . 20
7.6. Ignoring Loss of Undecryptable Packets . . . . . . . . . 20
7.7. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 21
7.8. Persistent Congestion . . . . . . . . . . . . . . . . . . 21
7.9. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 22
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7.10. Under-utilizing the Congestion Window . . . . . . . . . . 23
8. Security Considerations . . . . . . . . . . . . . . . . . . . 24
8.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 24
8.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 24
8.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 24
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 25
10.1. Normative References . . . . . . . . . . . . . . . . . . 25
10.2. Informative References . . . . . . . . . . . . . . . . . 25
Appendix A. Loss Recovery Pseudocode . . . . . . . . . . . . . . 27
A.1. Tracking Sent Packets . . . . . . . . . . . . . . . . . . 27
A.1.1. Sent Packet Fields . . . . . . . . . . . . . . . . . 27
A.2. Constants of Interest . . . . . . . . . . . . . . . . . . 28
A.3. Variables of interest . . . . . . . . . . . . . . . . . . 28
A.4. Initialization . . . . . . . . . . . . . . . . . . . . . 29
A.5. On Sending a Packet . . . . . . . . . . . . . . . . . . . 29
A.6. On Receiving a Datagram . . . . . . . . . . . . . . . . . 30
A.7. On Receiving an Acknowledgment . . . . . . . . . . . . . 30
A.8. Setting the Loss Detection Timer . . . . . . . . . . . . 32
A.9. On Timeout . . . . . . . . . . . . . . . . . . . . . . . 33
A.10. Detecting Lost Packets . . . . . . . . . . . . . . . . . 34
Appendix B. Congestion Control Pseudocode . . . . . . . . . . . 35
B.1. Constants of interest . . . . . . . . . . . . . . . . . . 35
B.2. Variables of interest . . . . . . . . . . . . . . . . . . 36
B.3. Initialization . . . . . . . . . . . . . . . . . . . . . 36
B.4. On Packet Sent . . . . . . . . . . . . . . . . . . . . . 37
B.5. On Packet Acknowledgement . . . . . . . . . . . . . . . . 37
B.6. On New Congestion Event . . . . . . . . . . . . . . . . . 37
B.7. Process ECN Information . . . . . . . . . . . . . . . . . 38
B.8. On Packets Lost . . . . . . . . . . . . . . . . . . . . . 38
B.9. Upon dropping Initial or Handshake keys . . . . . . . . . 39
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 39
C.1. Since draft-ietf-quic-recovery-28 . . . . . . . . . . . . 39
C.2. Since draft-ietf-quic-recovery-27 . . . . . . . . . . . . 39
C.3. Since draft-ietf-quic-recovery-26 . . . . . . . . . . . . 40
C.4. Since draft-ietf-quic-recovery-25 . . . . . . . . . . . . 40
C.5. Since draft-ietf-quic-recovery-24 . . . . . . . . . . . . 40
C.6. Since draft-ietf-quic-recovery-23 . . . . . . . . . . . . 40
C.7. Since draft-ietf-quic-recovery-22 . . . . . . . . . . . . 40
C.8. Since draft-ietf-quic-recovery-21 . . . . . . . . . . . . 40
C.9. Since draft-ietf-quic-recovery-20 . . . . . . . . . . . . 40
C.10. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . . 41
C.11. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . . 41
C.12. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . . 42
C.13. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . . 42
C.14. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 43
C.15. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 43
C.16. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 43
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C.17. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 43
C.18. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 43
C.19. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 44
C.20. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 44
C.21. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 44
C.22. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 44
C.23. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 44
C.24. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 44
C.25. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 44
C.26. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 44
C.27. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 45
C.28. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 45
C.29. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 45
Appendix D. Contributors . . . . . . . . . . . . . . . . . . . . 45
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 45
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45
1. Introduction
QUIC is a new multiplexed and secure transport protocol atop UDP,
specified in [QUIC-TRANSPORT]. This document describes congestion
control and loss recovery for QUIC. Mechanisms described in this
document follow the spirit of existing TCP congestion control and
loss recovery mechanisms, described in RFCs, various Internet-drafts,
or academic papers, and also those prevalent in TCP implementations.
2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
Definitions of terms that are used in this document:
Ack-eliciting Frames: All frames other than ACK, PADDING, and
CONNECTION_CLOSE are considered ack-eliciting.
Ack-eliciting Packets: Packets that contain ack-eliciting frames
elicit an ACK from the receiver within the maximum ack delay and
are called ack-eliciting packets.
In-flight: Packets are considered in-flight when they are ack-
eliciting or contain a PADDING frame, and they have been sent but
are not acknowledged, declared lost, or abandoned along with old
keys.
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3. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
indicates the encryption level and includes a packet sequence number
(referred to below as a packet number). The encryption level
indicates the packet number space, as described in [QUIC-TRANSPORT].
Packet numbers never repeat within a packet number space for the
lifetime of a connection. Packet numbers are sent in monotonically
increasing order within a space, preventing ambiguity.
This design obviates the need for disambiguating between
transmissions and retransmissions and eliminates significant
complexity from QUIC's interpretation of TCP loss detection
mechanisms.
QUIC packets can contain multiple frames of different types. The
recovery mechanisms ensure that data and frames that need reliable
delivery are acknowledged or declared lost and sent in new packets as
necessary. The types of frames contained in a packet affect recovery
and congestion control logic:
* All packets are acknowledged, though packets that contain no ack-
eliciting frames are only acknowledged along with ack-eliciting
packets.
* Long header packets that contain CRYPTO frames are critical to the
performance of the QUIC handshake and use shorter timers for
acknowledgement.
* Packets containing frames besides ACK or CONNECTION_CLOSE frames
count toward congestion control limits and are considered in-
flight.
* PADDING frames cause packets to contribute toward bytes in flight
without directly causing an acknowledgment to be sent.
4. Relevant Differences Between QUIC and TCP
Readers familiar with TCP's loss detection and congestion control
will find algorithms here that parallel well-known TCP ones.
Protocol differences between QUIC and TCP however contribute to
algorithmic differences. We briefly describe these protocol
differences below.
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4.1. Separate Packet Number Spaces
QUIC uses separate packet number spaces for each encryption level,
except 0-RTT and all generations of 1-RTT keys use the same packet
number space. Separate packet number spaces ensures acknowledgement
of packets sent with one level of encryption will not cause spurious
retransmission of packets sent with a different encryption level.
Congestion control and round-trip time (RTT) measurement are unified
across packet number spaces.
4.2. Monotonically Increasing Packet Numbers
TCP conflates transmission order at the sender with delivery order at
the receiver, which results in retransmissions of the same data
carrying the same sequence number, and consequently leads to
"retransmission ambiguity". QUIC separates the two. QUIC uses a
packet number to indicate transmission order. Application data is
sent in one or more streams and delivery order is determined by
stream offsets encoded within STREAM frames.
QUIC's packet number is strictly increasing within a packet number
space, and directly encodes transmission order. A higher packet
number signifies that the packet was sent later, and a lower packet
number signifies that the packet was sent earlier. When a packet
containing ack-eliciting frames is detected lost, QUIC rebundles
necessary frames in a new packet with a new packet number, removing
ambiguity about which packet is acknowledged when an ACK is received.
Consequently, more accurate RTT measurements can be made, spurious
retransmissions are trivially detected, and mechanisms such as Fast
Retransmit can be applied universally, based only on packet number.
This design point significantly simplifies loss detection mechanisms
for QUIC. Most TCP mechanisms implicitly attempt to infer
transmission ordering based on TCP sequence numbers - a non-trivial
task, especially when TCP timestamps are not available.
4.3. Clearer Loss Epoch
QUIC starts a loss epoch when a packet is lost and ends one when any
packet sent after the epoch starts is acknowledged. TCP waits for
the gap in the sequence number space to be filled, and so if a
segment is lost multiple times in a row, the loss epoch may not end
for several round trips. Because both should reduce their congestion
windows only once per epoch, QUIC will do it once for every round
trip that experiences loss, while TCP may only do it once across
multiple round trips.
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4.4. No Reneging
QUIC ACKs contain information that is similar to TCP SACK, but QUIC
does not allow any acked packet to be reneged, greatly simplifying
implementations on both sides and reducing memory pressure on the
sender.
4.5. More ACK Ranges
QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In
high loss environments, this speeds recovery, reduces spurious
retransmits, and ensures forward progress without relying on
timeouts.
4.6. Explicit Correction For Delayed Acknowledgements
QUIC endpoints measure the delay incurred between when a packet is
received and when the corresponding acknowledgment is sent, allowing
a peer to maintain a more accurate round-trip time estimate; see
Section 13.2 of [QUIC-TRANSPORT].
4.7. Probe Timeout Replaces RTO and TLP
QUIC uses a probe timeout (see Section 6.2), with a timer based on
TCP's RTO computation. QUIC's PTO includes the peer's maximum
expected acknowledgement delay instead of using a fixed minimum
timeout. QUIC does not collapse the congestion window until
persistent congestion (Section 7.8) is declared, unlike TCP, which
collapses the congestion window upon expiry of an RTO. Instead of
collapsing the congestion window and declaring everything in-flight
lost, QUIC allows probe packets to temporarily exceed the congestion
window whenever the timer expires.
In doing this, QUIC avoids unnecessary congestion window reductions,
obviating the need for correcting mechanisms such as F-RTO [RFC5682].
Since QUIC does not collapse the congestion window on a PTO
expiration, a QUIC sender is not limited from sending more in-flight
packets after a PTO expiration if it still has available congestion
window. This occurs when a sender is application-limited and the PTO
timer expires. This is more aggressive than TCP's RTO mechanism when
application-limited, but identical when not application-limited.
A single packet loss at the tail does not indicate persistent
congestion, so QUIC specifies a time-based definition to ensure one
or more packets are sent prior to a dramatic decrease in congestion
window; see Section 7.8.
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4.8. The Minimum Congestion Window is Two Packets
TCP uses a minimum congestion window of one packet. However, loss of
that single packet means that the sender needs to waiting for a PTO
(Section 6.2) to recover, which can be much longer than a round-trip
time. Sending a single ack-eliciting packet also increases the
chances of incurring additional latency when a receiver delays its
acknowledgement.
QUIC therefore recommends that the minimum congestion window be two
packets. While this increases network load, it is considered safe,
since the sender will still reduce its sending rate exponentially
under persistent congestion (Section 6.2).
5. Estimating the Round-Trip Time
At a high level, an endpoint measures the time from when a packet was
sent to when it is acknowledged as a round-trip time (RTT) sample.
The endpoint uses RTT samples and peer-reported host delays (see
Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical
description of the network path's RTT. An endpoint computes the
following three values for each path: the minimum value observed over
the lifetime of the path (min_rtt), an exponentially-weighted moving
average (smoothed_rtt), and the mean deviation (referred to as
"variation" in the rest of this document) in the observed RTT samples
(rttvar).
5.1. Generating RTT samples
An endpoint generates an RTT sample on receiving an ACK frame that
meets the following two conditions:
* the largest acknowledged packet number is newly acknowledged, and
* at least one of the newly acknowledged packets was ack-eliciting.
The RTT sample, latest_rtt, is generated as the time elapsed since
the largest acknowledged packet was sent:
latest_rtt = ack_time - send_time_of_largest_acked
An RTT sample is generated using only the largest acknowledged packet
in the received ACK frame. This is because a peer reports ACK delays
for only the largest acknowledged packet in an ACK frame. While the
reported ACK delay is not used by the RTT sample measurement, it is
used to adjust the RTT sample in subsequent computations of
smoothed_rtt and rttvar Section 5.3.
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To avoid generating multiple RTT samples for a single packet, an ACK
frame SHOULD NOT be used to update RTT estimates if it does not newly
acknowledge the largest acknowledged packet.
An RTT sample MUST NOT be generated on receiving an ACK frame that
does not newly acknowledge at least one ack-eliciting packet. A peer
usually does not send an ACK frame when only non-ack-eliciting
packets are received. Therefore an ACK frame that contains
acknowledgements for only non-ack-eliciting packets could include an
arbitrarily large Ack Delay value. Ignoring such ACK frames avoids
complications in subsequent smoothed_rtt and rttvar computations.
A sender might generate multiple RTT samples per RTT when multiple
ACK frames are received within an RTT. As suggested in [RFC6298],
doing so might result in inadequate history in smoothed_rtt and
rttvar. Ensuring that RTT estimates retain sufficient history is an
open research question.
5.2. Estimating min_rtt
min_rtt is the minimum RTT observed for a given network path.
min_rtt is set to the latest_rtt on the first RTT sample, and to the
lesser of min_rtt and latest_rtt on subsequent samples. In this
document, min_rtt is used by loss detection to reject implausibly
small rtt samples.
An endpoint uses only locally observed times in computing the min_rtt
and does not adjust for ACK delays reported by the peer. Doing so
allows the endpoint to set a lower bound for the smoothed_rtt based
entirely on what it observes (see Section 5.3), and limits potential
underestimation due to erroneously-reported delays by the peer.
The RTT for a network path may change over time. If a path's actual
RTT decreases, the min_rtt will adapt immediately on the first low
sample. If the path's actual RTT increases, the min_rtt will not
adapt to it, allowing future RTT samples that are smaller than the
new RTT be included in smoothed_rtt.
5.3. Estimating smoothed_rtt and rttvar
smoothed_rtt is an exponentially-weighted moving average of an
endpoint's RTT samples, and rttvar is the variation in the RTT
samples, estimated using a mean variation.
The calculation of smoothed_rtt uses path latency after adjusting RTT
samples for acknowledgement delays. These delays are computed using
the ACK Delay field of the ACK frame as described in Section 19.3 of
[QUIC-TRANSPORT]. For packets sent in the ApplicationData packet
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number space, a peer limits any delay in sending an acknowledgement
for an ack-eliciting packet to no greater than the value it
advertised in the max_ack_delay transport parameter. Consequently,
when a peer reports an Ack Delay that is greater than its
max_ack_delay, the delay is attributed to reasons out of the peer's
control, such as scheduler latency at the peer or loss of previous
ACK frames. Any delays beyond the peer's max_ack_delay are therefore
considered effectively part of path delay and incorporated into the
smoothed_rtt estimate.
When adjusting an RTT sample using peer-reported acknowledgement
delays, an endpoint:
* MUST ignore the Ack Delay field of the ACK frame for packets sent
in the Initial and Handshake packet number space.
* MUST use the lesser of the value reported in Ack Delay field of
the ACK frame and the peer's max_ack_delay transport parameter.
* MUST NOT apply the adjustment if the resulting RTT sample is
smaller than the min_rtt. This limits the underestimation that a
misreporting peer can cause to the smoothed_rtt.
smoothed_rtt and rttvar are computed as follows, similar to
[RFC6298].
When there are no samples for a network path, and on the first RTT
sample for the network path:
smoothed_rtt = rtt_sample
rttvar = rtt_sample / 2
Before any RTT samples are available, the initial RTT is used as
rtt_sample. On the first RTT sample for the network path, that
sample is used as rtt_sample. This ensures that the first
measurement erases the history of any persisted or default values.
On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:
ack_delay = min(Ack Delay in ACK Frame, max_ack_delay)
adjusted_rtt = latest_rtt
if (min_rtt + ack_delay < latest_rtt):
adjusted_rtt = latest_rtt - ack_delay
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
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6. Loss Detection
QUIC senders use acknowledgements to detect lost packets, and a probe
time out (see Section 6.2) to ensure acknowledgements are received.
This section provides a description of these algorithms.
If a packet is lost, the QUIC transport needs to recover from that
loss, such as by retransmitting the data, sending an updated frame,
or abandoning the frame. For more information, see Section 13.3 of
[QUIC-TRANSPORT].
6.1. Acknowledgement-based Detection
Acknowledgement-based loss detection implements the spirit of TCP's
Fast Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK],
SACK loss recovery [RFC6675], and RACK [RACK]. This section provides
an overview of how these algorithms are implemented in QUIC.
A packet is declared lost if it meets all the following conditions:
* The packet is unacknowledged, in-flight, and was sent prior to an
acknowledged packet.
* Either its packet number is kPacketThreshold smaller than an
acknowledged packet (Section 6.1.1), or it was sent long enough in
the past (Section 6.1.2).
The acknowledgement indicates that a packet sent later was delivered,
and the packet and time thresholds provide some tolerance for packet
reordering.
Spuriously declaring packets as lost leads to unnecessary
retransmissions and may result in degraded performance due to the
actions of the congestion controller upon detecting loss.
Implementations can detect spurious retransmissions and increase the
reordering threshold in packets or time to reduce future spurious
retransmissions and loss events. Implementations with adaptive time
thresholds MAY choose to start with smaller initial reordering
thresholds to minimize recovery latency.
6.1.1. Packet Threshold
The RECOMMENDED initial value for the packet reordering threshold
(kPacketThreshold) is 3, based on best practices for TCP loss
detection [RFC5681] [RFC6675]. Implementations SHOULD NOT use a
packet threshold less than 3, to keep in line with TCP [RFC5681].
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Some networks may exhibit higher degrees of reordering, causing a
sender to detect spurious losses. Algorithms that increase the
reordering threshold after spuriously detecting losses, such as TCP-
NCR [RFC4653], have proven to be useful in TCP and are expected to at
least as useful in QUIC. Re-ordering could be more common with QUIC
than TCP, because network elements cannot observe and fix the order
of out-of-order packets.
6.1.2. Time Threshold
Once a later packet within the same packet number space has been
acknowledged, an endpoint SHOULD declare an earlier packet lost if it
was sent a threshold amount of time in the past. To avoid declaring
packets as lost too early, this time threshold MUST be set to at
least the local timer granularity, as indicated by the kGranularity
constant. The time threshold is:
max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity)
If packets sent prior to the largest acknowledged packet cannot yet
be declared lost, then a timer SHOULD be set for the remaining time.
Using max(smoothed_rtt, latest_rtt) protects from the two following
cases:
* the latest RTT sample is lower than the smoothed RTT, perhaps due
to reordering where the acknowledgement encountered a shorter
path;
* the latest RTT sample is higher than the smoothed RTT, perhaps due
to a sustained increase in the actual RTT, but the smoothed RTT
has not yet caught up.
The RECOMMENDED time threshold (kTimeThreshold), expressed as a
round-trip time multiplier, is 9/8. The RECOMMENDED value of the
timer granularity (kGranularity) is 1ms.
Implementations MAY experiment with absolute thresholds, thresholds
from previous connections, adaptive thresholds, or including RTT
variation. Smaller thresholds reduce reordering resilience and
increase spurious retransmissions, and larger thresholds increase
loss detection delay.
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6.2. Probe Timeout
A Probe Timeout (PTO) triggers sending one or two probe datagrams
when ack-eliciting packets are not acknowledged within the expected
period of time or the server may not have validated the client's
address. A PTO enables a connection to recover from loss of tail
packets or acknowledgements.
A PTO timer expiration event does not indicate packet loss and MUST
NOT cause prior unacknowledged packets to be marked as lost. When an
acknowledgement is received that newly acknowledges packets, loss
detection proceeds as dictated by packet and time threshold
mechanisms; see Section 6.1.
As with loss detection, the probe timeout is per packet number space.
The PTO algorithm used in QUIC implements the reliability functions
of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for
TCP [RFC5682]. The timeout computation is based on TCP's
retransmission timeout period [RFC6298].
6.2.1. Computing PTO
When an ack-eliciting packet is transmitted, the sender schedules a
timer for the PTO period as follows:
PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay
The PTO period is the amount of time that a sender ought to wait for
an acknowledgement of a sent packet. This time period includes the
estimated network roundtrip-time (smoothed_rtt), the variation in the
estimate (4*rttvar), and max_ack_delay, to account for the maximum
time by which a receiver might delay sending an acknowledgement.
When the PTO is armed for Initial or Handshake packet number spaces,
the max_ack_delay is 0, as specified in 13.2.1 of [QUIC-TRANSPORT].
The PTO value MUST be set to at least kGranularity, to avoid the
timer expiring immediately.
A sender recomputes and may need to reset its PTO timer every time an
ack-eliciting packet is sent or acknowledged, when the handshake is
confirmed, or when Initial or Handshake keys are discarded. This
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
flight, the timer MUST be set for the packet number space with the
earliest timeout, with one exception. The ApplicationData packet
number space (Section 4.1.1 of [QUIC-TLS]) MUST be ignored until the
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handshake completes. Not arming the PTO for ApplicationData prevents
a client from retransmitting a 0-RTT packet on a PTO expiration
before confirming that the server is able to decrypt 0-RTT packets,
and prevents a server from sending a 1-RTT packet on a PTO expiration
before it has the keys to process an acknowledgement.
When a PTO timer expires, the PTO backoff MUST be increased,
resulting in the PTO period being set to twice its current value.
The PTO backoff factor is reset when an acknowledgement is received,
except in the following case. A server might take longer to respond
to packets during the handshake than otherwise. To protect such a
server from repeated client probes, the PTO backoff is not reset at a
client that is not yet certain that the server has finished
validating the client's address. That is, a client does not reset
the PTO backoff factor on receiving acknowledgements until it
receives a HANDSHAKE_DONE frame or an acknowledgement for one of its
Handshake or 1-RTT packets.
This exponential reduction in the sender's rate is important because
consecutive PTOs might be caused by loss of packets or
acknowledgements due to severe congestion. Even when there are ack-
eliciting packets in-flight in multiple packet number spaces, the
exponential increase in probe timeout occurs across all spaces to
prevent excess load on the network. For example, a timeout in the
Initial packet number space doubles the length of the timeout in the
Handshake packet number space.
The life of a connection that is experiencing consecutive PTOs is
limited by the endpoint's idle timeout.
The probe timer MUST NOT be set if the time threshold Section 6.1.2
loss detection timer is set. The time threshold loss detection timer
is expected to both expire earlier than the PTO and be less likely to
spuriously retransmit data.
6.2.2. Handshakes and New Paths
Resumed connections over the same network MAY use the previous
connection's final smoothed RTT value as the resumed connection's
initial RTT. When no previous RTT is available, the initial RTT
SHOULD be set to 333ms, resulting in a 1 second initial timeout, as
recommended in [RFC6298].
A connection MAY use the delay between sending a PATH_CHALLENGE and
receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in
Appendix A.2) for a new path, but the delay SHOULD NOT be considered
an RTT sample.
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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,
but they are removed from bytes in flight when the Initial and
Handshake keys are discarded, as described below in Section 6.4.
When Initial or Handshake keys are discarded, the PTO and loss
detection timers MUST be reset, because discarding keys indicates
forward progress and the loss detection timer might have been set for
a now discarded packet number space.
6.2.2.1. Before Address Validation
Until the server has validated the client's address on the path, the
amount of data it can send is limited to three times the amount of
data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If
no additional data can be sent, the server's PTO timer MUST NOT be
armed until datagrams have been received from the client, because
packets sent on PTO count against the anti-amplification limit. Note
that the server could fail to validate the client's address even if
0-RTT is accepted.
Since the server could be blocked until more packets are received
from the client, it is the client's responsibility to send packets to
unblock the server until it is certain that the server has finished
its address validation (see Section 8 of [QUIC-TRANSPORT]). That is,
the client MUST set the probe timer if the client has not received an
acknowledgement for one of its Handshake or 1-RTT packets, and has
not received a HANDSHAKE_DONE frame. If Handshake keys are available
to the client, it MUST send a Handshake packet, and 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
When a server receives an Initial packet containing duplicate CRYPTO
data, it can assume the client did not receive all of the server's
CRYPTO data sent in Initial packets, or the client's estimated RTT is
too small. When a client receives Handshake or 1-RTT packets prior
to obtaining Handshake keys, it may assume some or all of the
server's Initial packets were lost.
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To speed up handshake completion under these conditions, an endpoint
MAY send a packet containing unacknowledged CRYPTO data earlier than
the PTO expiry, subject to address validation limits; see Section 8.1
of [QUIC-TRANSPORT].
Peers can also use coalesced packets to ensure that each datagram
elicits at least one acknowledgement. For example, clients can
coalesce an Initial packet containing PING and PADDING frames with a
0-RTT data packet and a server can coalesce an Initial packet
containing a PING frame with one or more packets in its first flight.
6.2.4. Sending Probe Packets
When a PTO timer expires, a sender MUST send at least one ack-
eliciting packet in the packet number space as a probe, unless there
is no data available to send. An endpoint MAY send up to two full-
sized datagrams containing ack-eliciting packets, to avoid an
expensive consecutive PTO expiration due to a single lost datagram or
transmit data from multiple packet number spaces. All probe packets
sent on a PTO MUST be ack-eliciting.
In addition to sending data in the packet number space for which the
timer expired, the sender SHOULD send ack-eliciting packets from
other packet number spaces with in-flight data, coalescing packets if
possible. This is particularly valuable when the server has both
Initial and Handshake data in-flight or the client has both Handshake
and ApplicationData in-flight, because the peer might only have
receive keys for one of the two packet number spaces.
If the sender wants to elicit a faster acknowledgement on PTO, it can
skip a packet number to eliminate the ack delay.
When the PTO timer expires, and there is new or previously sent
unacknowledged data, it MUST be sent. A probe packet SHOULD carry
new data when possible. A probe packet MAY carry retransmitted
unacknowledged data when new data is unavailable, when flow control
does not permit new data to be sent, or to opportunistically reduce
loss recovery delay. Implementations MAY use alternative strategies
for determining the content of probe packets, including sending new
or retransmitted data based on the application's priorities.
It is possible the sender has no new or previously-sent data to send.
As an example, consider the following sequence of events: new
application data is sent in a STREAM frame, deemed lost, then
retransmitted in a new packet, and then the original transmission is
acknowledged. When there is no data to send, the sender SHOULD send
a PING or other ack-eliciting frame in a single packet, re-arming the
PTO timer.
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Alternatively, instead of sending an ack-eliciting packet, the sender
MAY mark any packets still in flight as lost. Doing so avoids
sending an additional packet, but increases the risk that loss is
declared too aggressively, resulting in an unnecessary rate reduction
by the congestion controller.
Consecutive PTO periods increase exponentially, and as a result,
connection recovery latency increases exponentially as packets
continue to be dropped in the network. Sending two packets on PTO
expiration increases resilience to packet drops, thus reducing the
probability of consecutive PTO events.
When the PTO timer expires multiple times and new data cannot be
sent, implementations must choose between sending the same payload
every time or sending different payloads. Sending the same payload
may be simpler and ensures the highest priority frames arrive first.
Sending different payloads each time reduces the chances of spurious
retransmission.
6.3. Handling Retry Packets
A Retry packet causes a client to send another Initial packet,
effectively restarting the connection process. A Retry packet
indicates that the Initial was received, but not processed. A Retry
packet cannot be treated as an acknowledgment, because it does not
indicate that a packet was processed or specify the packet number.
Clients that receive a Retry packet reset congestion control and loss
recovery state, including resetting any pending timers. Other
connection state, in particular cryptographic handshake messages, is
retained; see Section 17.2.5 of [QUIC-TRANSPORT].
The client MAY compute an RTT estimate to the server as the time
period from when the first Initial was sent to when a Retry or a
Version Negotiation packet is received. The client MAY use this
value in place of its default for the initial RTT estimate.
6.4. Discarding Keys and Packet State
When packet protection keys are discarded (see Section 4.10 of
[QUIC-TLS]), all packets that were sent with those keys can no longer
be acknowledged because their acknowledgements cannot be processed
anymore. The sender MUST discard all recovery state associated with
those packets and MUST remove them from the count of bytes in flight.
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Endpoints stop sending and receiving Initial packets once they start
exchanging Handshake packets; see Section 17.2.2.1 of
[QUIC-TRANSPORT]. At this point, recovery state for all in-flight
Initial packets is discarded.
When 0-RTT is rejected, recovery state for all in-flight 0-RTT
packets is discarded.
If a server accepts 0-RTT, but does not buffer 0-RTT packets that
arrive before Initial packets, early 0-RTT packets will be declared
lost, but that is expected to be infrequent.
It is expected that keys are discarded after packets encrypted with
them would be acknowledged or declared lost. Initial secrets however
might be destroyed sooner, as soon as handshake keys are available;
see Section 4.11.1 of [QUIC-TLS].
7. Congestion Control
This document specifies a congestion controller for QUIC similar to
TCP NewReno [RFC6582].
The signals QUIC provides for congestion control are generic and are
designed to support different algorithms. Endpoints can unilaterally
choose a different algorithm to use, such as Cubic [RFC8312].
If an endpoint uses a different controller than that specified in
this document, the chosen controller MUST conform to the congestion
control guidelines specified in Section 3.1 of [RFC8085].
Similar to TCP, packets containing only ACK frames do not count
towards bytes in flight and are not congestion controlled. Unlike
TCP, QUIC can detect the loss of these packets and MAY use that
information to adjust the congestion controller or the rate of ACK-
only packets being sent, but this document does not describe a
mechanism for doing so.
The algorithm in this document specifies and uses the controller's
congestion window in bytes.
An endpoint MUST NOT send a packet if it would cause bytes_in_flight
(see Appendix B.2) to be larger than the congestion window, unless
the packet is sent on a PTO timer expiration; see Section 6.2.
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7.1. Explicit Congestion Notification
If a path has been verified to support ECN [RFC3168] [RFC8311], QUIC
treats a Congestion Experienced (CE) codepoint in the IP header as a
signal of congestion. This document specifies an endpoint's response
when its peer receives packets with the ECN-CE codepoint.
7.2. Initial and Minimum Congestion Window
QUIC begins every connection in slow start with the congestion window
set to an initial value. Endpoints SHOULD use an initial congestion
window of 10 times the maximum datagram size (max_datagram_size),
limited to the larger of 14720 or twice the maximum datagram size.
This follows the analysis and recommendations in [RFC6928],
increasing the byte limit to account for the smaller 8 byte overhead
of UDP compared to the 20 byte overhead for TCP.
Prior to validating the client's address, the server can be further
limited by the anti-amplification limit as specified in Section 8.1
of [QUIC-TRANSPORT]. Though the anti-amplification limit can prevent
the congestion window from being fully utilized and therefore slow
down the increase in congestion window, it does not directly affect
the congestion window.
The minimum congestion window is the smallest value the congestion
window can decrease to as a response to loss, ECN-CE, or persistent
congestion. The RECOMMENDED value is 2 * max_datagram_size.
7.3. Slow Start
While in slow start, QUIC increases the congestion window by the
number of bytes acknowledged when each acknowledgment is processed,
resulting in exponential growth of the congestion window.
QUIC exits slow start upon loss or upon increase in the ECN-CE
counter. When slow start is exited, the congestion window halves and
the slow start threshold is set to the new congestion window. QUIC
re-enters slow start any time the congestion window is less than the
slow start threshold, which only occurs after persistent congestion
is declared.
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7.4. Congestion Avoidance
Slow start exits to congestion avoidance. Congestion avoidance uses
an Additive Increase Multiplicative Decrease (AIMD) approach that
increases the congestion window by one maximum packet size per
congestion window acknowledged. When 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
A recovery period is entered when loss or ECN-CE marking of a packet
is detected in congestion avoidance after the congestion window and
slow start threshold have been decreased. A recovery period ends
when a packet sent during the recovery period is acknowledged. This
is slightly different from TCP's definition of recovery, which ends
when the lost packet that started recovery is acknowledged.
The recovery period aims to limit congestion window reduction to once
per round trip. Therefore during recovery, the congestion window
remains unchanged irrespective of new losses or increases in the ECN-
CE counter.
When entering recovery, a single packet MAY be sent even if bytes in
flight now exceeds the recently reduced congestion window. This
speeds up loss recovery if the data in the lost packet is
retransmitted and is similar to TCP as described in Section 5 of
[RFC6675]. If further packets are lost while the sender is in
recovery, sending any packets in response MUST obey the congestion
window limit.
7.6. Ignoring Loss of Undecryptable Packets
During the handshake, some packet protection keys might not be
available when a packet arrives and the receiver can choose to drop
the packet. In particular, Handshake and 0-RTT packets cannot be
processed until the Initial packets arrive and 1-RTT packets cannot
be processed until the handshake completes. Endpoints MAY ignore the
loss of Handshake, 0-RTT, and 1-RTT packets that might have arrived
before the peer had packet protection keys to process those packets.
Endpoints MUST NOT ignore the loss of packets that were sent after
the earliest acknowledged packet in a given packet number space.
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7.7. Probe Timeout
Probe packets MUST NOT be blocked by the congestion controller. A
sender MUST however count these packets as being additionally in
flight, since these packets add network load without establishing
packet loss. Note that sending probe packets might cause the
sender's bytes in flight to exceed the congestion window until an
acknowledgement is received that establishes loss or delivery of
packets.
7.8. Persistent Congestion
When an ACK frame is received that establishes loss of all in-flight
packets sent over a long enough period of time, the network is
considered to be experiencing persistent congestion. Commonly, this
can be established by consecutive PTOs, but since the PTO timer is
reset when a new ack-eliciting packet is sent, an explicit duration
must be used to account for those cases where PTOs do not occur or
are substantially delayed. The rationale for this threshold is to
enable a sender to use initial PTOs for aggressive probing, as TCP
does with Tail Loss Probe (TLP) [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:
(smoothed_rtt + 4 * rttvar + max_ack_delay) *
kPersistentCongestionThreshold
For example, assume:
smoothed_rtt = 1
rttvar = 0
max_ack_delay = 0
kPersistentCongestionThreshold = 3
If an ack-eliciting packet is sent at time t = 0, the following
scenario would illustrate persistent congestion:
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+------+------------------------+
| Time | Action |
+======+========================+
| t=0 | Send Pkt #1 (App Data) |
+------+------------------------+
| t=1 | Send Pkt #2 (PTO 1) |
+------+------------------------+
| t=3 | Send Pkt #3 (PTO 2) |
+------+------------------------+
| t=7 | Send Pkt #4 (PTO 3) |
+------+------------------------+
| t=8 | Recv ACK of Pkt #4 |
+------+------------------------+
Table 1
The first three packets are determined to be lost when the
acknowledgement of packet 4 is received at t = 8. The congestion
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
window MUST be reduced to the minimum congestion window
(kMinimumWindow). This response of collapsing the congestion window
on persistent congestion is functionally similar to a sender's
response on a Retransmission Timeout (RTO) in TCP [RFC5681] after
Tail Loss Probes (TLP) [RACK].
7.9. Pacing
This document does not specify a pacer, but it is RECOMMENDED that a
sender pace sending of all in-flight packets based on input from the
congestion controller. Sending multiple packets into the network
without any delay between them creates a packet burst that might
cause short-term congestion and losses. Implementations MUST either
use pacing or another method to limit such bursts to the initial
congestion window; see Section 7.2.
An implementation should take care to architect its congestion
controller to work well with a pacer. For instance, a pacer might
wrap the congestion controller and control the availability of the
congestion window, or a pacer might pace out packets handed to it by
the congestion controller.
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Timely delivery of ACK frames is important for efficient loss
recovery. Packets containing only ACK frames SHOULD therefore not be
paced, to avoid delaying their delivery to the peer.
Endpoints can implement pacing as they choose. A perfectly paced
sender spreads packets exactly evenly over time. For a window-based
congestion controller, such as the one in this document, that rate
can be computed by averaging the congestion window over the round-
trip time. Expressed as a rate in bytes:
rate = N * congestion_window / smoothed_rtt
Or, expressed as an inter-packet interval:
interval = smoothed_rtt * packet_size / congestion_window / N
Using a value for "N" that is small, but at least 1 (for example,
1.25) ensures that variations in round-trip time don't result in
under-utilization of the congestion window. Values of 'N' larger
than 1 ultimately result in sending packets as acknowledgments are
received rather than when timers fire, provided the congestion window
is fully utilized and acknowledgments arrive at regular intervals.
Practical considerations, such as packetization, scheduling delays,
and computational efficiency, can cause a sender to deviate from this
rate over time periods that are much shorter than a round-trip time.
One possible implementation strategy for pacing uses a leaky bucket
algorithm, where the capacity of the "bucket" is limited to the
maximum burst size and the rate the "bucket" fills is determined by
the above function.
7.10. Under-utilizing the Congestion Window
When bytes in flight is smaller than the congestion window and
sending is not pacing limited, the congestion window is under-
utilized. When this occurs, the congestion window SHOULD NOT be
increased in either slow start or congestion avoidance. This can
happen due to insufficient application data or flow control limits.
A sender MAY use the pipeACK method described in Section 4.3 of
[RFC7661] to determine if the congestion window is sufficiently
utilized.
A sender that paces packets (see Section 7.9) might delay sending
packets and not fully utilize the congestion window due to this
delay. A sender SHOULD NOT consider itself application limited if it
would have fully utilized the congestion window without pacing delay.
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A sender MAY implement alternative mechanisms to update its
congestion window after periods of under-utilization, such as those
proposed for TCP in [RFC7661].
8. Security Considerations
8.1. Congestion Signals
Congestion control fundamentally involves the consumption of signals
- both loss and ECN codepoints - from unauthenticated entities. On-
path attackers can spoof or alter these signals. An attacker can
cause endpoints to reduce their sending rate by dropping packets, or
alter send rate by changing ECN codepoints.
8.2. Traffic Analysis
Packets that carry only ACK frames can be heuristically identified by
observing packet size. Acknowledgement patterns may expose
information about link characteristics or application behavior.
Endpoints can use PADDING frames or bundle acknowledgments with other
frames to reduce leaked information.
8.3. Misreporting ECN Markings
A receiver can misreport ECN markings to alter the congestion
response of a sender. Suppressing reports of ECN-CE markings could
cause a sender to increase their send rate. This increase could
result in congestion and loss.
A sender MAY attempt to detect suppression of reports by marking
occasional packets that they send with ECN-CE. If a packet sent with
ECN-CE is not reported as having been CE marked when the packet is
acknowledged, then the sender SHOULD disable ECN for that path.
Reporting additional ECN-CE markings will cause a sender to reduce
their sending rate, which is similar in effect to advertising reduced
connection flow control limits and so no advantage is gained by doing
so.
Endpoints choose the congestion controller that they use. Though
congestion controllers generally treat reports of ECN-CE markings as
equivalent to loss [RFC8311], the exact response for each controller
could be different. Failure to correctly respond to information
about ECN markings is therefore difficult to detect.
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9. IANA Considerations
This document has no IANA actions.
10. References
10.1. Normative References
[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", Work in Progress, Internet-Draft, draft-ietf-quic-
tls-29, 10 June 2020,
<https://tools.ietf.org/html/draft-ietf-quic-tls-29>.
[QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", Work in Progress,
Internet-Draft, draft-ietf-quic-transport-29, 10 June
2020, <https://tools.ietf.org/html/draft-ietf-quic-
transport-29>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
10.2. Informative References
[FACK] Mathis, M. and J. Mahdavi, "Forward Acknowledgement:
Refining TCP Congestion Control", ACM SIGCOMM , August
1996.
[RACK] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "RACK:
a time-based fast loss detection algorithm for TCP", Work
in Progress, Internet-Draft, draft-ietf-tcpm-rack-08, 9
March 2020, <http://www.ietf.org/internet-drafts/draft-
ietf-tcpm-rack-08.txt>.
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[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>.
[RFC4653] Bhandarkar, S., Reddy, A. L. N., Allman, M., and E.
Blanton, "Improving the Robustness of TCP to Non-
Congestion Events", RFC 4653, DOI 10.17487/RFC4653, August
2006, <https://www.rfc-editor.org/info/rfc4653>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>.
[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
"Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
Spurious Retransmission Timeouts with TCP", RFC 5682,
DOI 10.17487/RFC5682, September 2009,
<https://www.rfc-editor.org/info/rfc5682>.
[RFC5827] Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
P. Hurtig, "Early Retransmit for TCP and Stream Control
Transmission Protocol (SCTP)", RFC 5827,
DOI 10.17487/RFC5827, May 2010,
<https://www.rfc-editor.org/info/rfc5827>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<https://www.rfc-editor.org/info/rfc6298>.
[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012,
<https://www.rfc-editor.org/info/rfc6582>.
[RFC6675] Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
and Y. Nishida, "A Conservative Loss Recovery Algorithm
Based on Selective Acknowledgment (SACK) for TCP",
RFC 6675, DOI 10.17487/RFC6675, August 2012,
<https://www.rfc-editor.org/info/rfc6675>.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
"Increasing TCP's Initial Window", RFC 6928,
DOI 10.17487/RFC6928, April 2013,
<https://www.rfc-editor.org/info/rfc6928>.
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[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<https://www.rfc-editor.org/info/rfc7661>.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion
Notification (ECN) Experimentation", RFC 8311,
DOI 10.17487/RFC8311, January 2018,
<https://www.rfc-editor.org/info/rfc8311>.
[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
RFC 8312, DOI 10.17487/RFC8312, February 2018,
<https://www.rfc-editor.org/info/rfc8312>.
Appendix A. Loss Recovery Pseudocode
We now describe an example implementation of the loss detection
mechanisms described in Section 6.
A.1. Tracking Sent Packets
To correctly implement congestion control, a QUIC sender tracks every
ack-eliciting packet until the packet is acknowledged or lost. It is
expected that implementations will be able to access this information
by packet number and crypto context and store the per-packet fields
(Appendix A.1.1) for loss recovery and congestion control.
After a packet is declared lost, the endpoint can track it for an
amount of time comparable to the maximum expected packet reordering,
such as 1 RTT. This allows for detection of spurious
retransmissions.
Sent packets are tracked for each packet number space, and ACK
processing only applies to a single space.
A.1.1. Sent Packet Fields
packet_number: The packet number of the sent packet.
ack_eliciting: A boolean that indicates whether a packet is ack-
eliciting. If true, it is expected that an acknowledgement will
be received, though the peer could delay sending the ACK frame
containing it by up to the MaxAckDelay.
in_flight: A boolean that indicates whether the packet counts
towards bytes in flight.
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sent_bytes: The number of bytes sent in the packet, not including
UDP or IP overhead, but including QUIC framing overhead.
time_sent: The time the packet was sent.
A.2. Constants of Interest
Constants used in loss recovery are based on a combination of RFCs,
papers, and common practice.
kPacketThreshold: Maximum reordering in packets before packet
threshold loss detection considers a packet lost. The value
recommended in Section 6.1.1 is 3.
kTimeThreshold: Maximum reordering in time before time threshold
loss detection considers a packet lost. Specified as an RTT
multiplier. The value recommended in Section 6.1.2 is 9/8.
kGranularity: Timer granularity. This is a system-dependent value,
and Section 6.1.2 recommends a value of 1ms.
kInitialRtt: The RTT used before an RTT sample is taken. The value
recommended in Section 6.2.2 is 500ms.
kPacketNumberSpace: An enum to enumerate the three packet number
spaces.
enum kPacketNumberSpace {
Initial,
Handshake,
ApplicationData,
}
A.3. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as
described in Section 5.3.
rttvar: The RTT variation, computed as described in Section 5.3.
min_rtt: The minimum RTT seen in the connection, ignoring ack delay,
as described in Section 5.2.
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max_ack_delay: The maximum amount of time by which the receiver
intends to delay acknowledgments for packets in the
ApplicationData packet number space. The actual ack_delay in a
received ACK frame may be larger due to late timers, reordering,
or lost ACK frames.
loss_detection_timer: Multi-modal timer used for loss detection.
pto_count: The number of times a PTO has been sent without receiving
an ack.
time_of_last_ack_eliciting_packet[kPacketNumberSpace]: The time the
most recent ack-eliciting packet was sent.
largest_acked_packet[kPacketNumberSpace]: The largest packet number
acknowledged in the packet number space so far.
loss_time[kPacketNumberSpace]: The time at which the next packet in
that packet number space will be considered lost based on
exceeding the reordering window in time.
sent_packets[kPacketNumberSpace]: An association of packet numbers
in a packet number space to information about them. Described in
detail above in Appendix A.1.
A.4. Initialization
At the beginning of the connection, initialize the loss detection
variables as follows:
loss_detection_timer.reset()
pto_count = 0
latest_rtt = 0
smoothed_rtt = initial_rtt
rttvar = initial_rtt / 2
min_rtt = 0
max_ack_delay = 0
for pn_space in [ Initial, Handshake, ApplicationData ]:
largest_acked_packet[pn_space] = infinite
time_of_last_ack_eliciting_packet[pn_space] = 0
loss_time[pn_space] = 0
A.5. On Sending a Packet
After a packet is sent, information about the packet is stored. The
parameters to OnPacketSent are described in detail above in
Appendix A.1.1.
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Pseudocode for OnPacketSent follows:
OnPacketSent(packet_number, pn_space, ack_eliciting,
in_flight, sent_bytes):
sent_packets[pn_space][packet_number].packet_number =
packet_number
sent_packets[pn_space][packet_number].time_sent = now()
sent_packets[pn_space][packet_number].ack_eliciting =
ack_eliciting
sent_packets[pn_space][packet_number].in_flight = in_flight
if (in_flight):
if (ack_eliciting):
time_of_last_ack_eliciting_packet[pn_space] = now()
OnPacketSentCC(sent_bytes)
sent_packets[pn_space][packet_number].size = sent_bytes
SetLossDetectionTimer()
A.6. On Receiving a Datagram
When a server is blocked by anti-amplification limits, receiving a
datagram unblocks it, even if none of the packets in the datagram are
successfully processed. In such a case, the PTO timer will need to
be re-armed.
Pseudocode for OnDatagramReceived follows:
OnDatagramReceived(datagram):
// If this datagram unblocks the server, arm the
// PTO timer to avoid deadlock.
if (server was at anti-amplification limit):
SetLossDetectionTimer()
A.7. On Receiving an Acknowledgment
When an ACK frame is received, it may newly acknowledge any number of
packets.
Pseudocode for OnAckReceived and UpdateRtt follow:
OnAckReceived(ack, pn_space):
if (largest_acked_packet[pn_space] == infinite):
largest_acked_packet[pn_space] = ack.largest_acked
else:
largest_acked_packet[pn_space] =
max(largest_acked_packet[pn_space], ack.largest_acked)
// DetectNewlyAckedPackets finds packets that are newly
// acknowledged and removes them from sent_packets.
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newly_acked_packets =
DetectAndRemoveAckedPackets(ack, pn_space)
// Nothing to do if there are no newly acked packets.
if (newly_acked_packets.empty()):
return
// If the largest acknowledged is newly acked and
// at least one ack-eliciting was newly acked, update the RTT.
if (newly_acked_packets.largest().packet_number ==
ack.largest_acked &&
IncludesAckEliciting(newly_acked_packets)):
latest_rtt =
now - sent_packets[pn_space][ack.largest_acked].time_sent
ack_delay = 0
if (pn_space == ApplicationData):
ack_delay = ack.ack_delay
UpdateRtt(ack_delay)
// Process ECN information if present.
if (ACK frame contains ECN information):
ProcessECN(ack, pn_space)
lost_packets = DetectAndRemoveLostPackets(pn_space)
if (!lost_packets.empty()):
OnPacketsLost(lost_packets)
OnPacketsAcked(newly_acked_packets)
// Reset pto_count unless the client is unsure if
// the server has validated the client's address.
if (PeerCompletedAddressValidation()):
pto_count = 0
SetLossDetectionTimer()
UpdateRtt(ack_delay):
if (is first RTT sample):
min_rtt = latest_rtt
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
return
// min_rtt ignores ack delay.
min_rtt = min(min_rtt, latest_rtt)
// Limit ack_delay by max_ack_delay
ack_delay = min(ack_delay, max_ack_delay)
// Adjust for ack delay if plausible.
adjusted_rtt = latest_rtt
if (latest_rtt > min_rtt + ack_delay):
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adjusted_rtt = latest_rtt - ack_delay
rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt)
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
A.8. Setting the Loss Detection Timer
QUIC loss detection uses a single timer for all timeout loss
detection. The duration of the timer is based on the timer's mode,
which is set in the packet and timer events further below. The
function SetLossDetectionTimer defined below shows how the single
timer is set.
This algorithm may result in the timer being set in the past,
particularly if timers wake up late. Timers set in the past fire
immediately.
Pseudocode for SetLossDetectionTimer follows:
GetLossTimeAndSpace():
time = loss_time[Initial]
space = Initial
for pn_space in [ Handshake, ApplicationData ]:
if (time == 0 || loss_time[pn_space] < time):
time = loss_time[pn_space];
space = pn_space
return time, space
GetPtoTimeAndSpace():
duration = (smoothed_rtt + max(4 * rttvar, kGranularity))
* (2 ^ pto_count)
// Arm PTO from now when there are no inflight packets.
if (no in-flight packets):
assert(!PeerCompletedAddressValidation())
if (has handshake keys):
return (now() + duration), Handshake
else:
return (now() + duration), Initial
pto_timeout = infinite
pto_space = Initial
for space in [ Initial, Handshake, ApplicationData ]:
if (no in-flight packets in space):
continue;
if (space == ApplicationData):
// Skip ApplicationData until handshake complete.
if (handshake is not complete):
return pto_timeout, pto_space
// Include max_ack_delay and backoff for ApplicationData.
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duration += max_ack_delay * (2 ^ pto_count)
t = time_of_last_ack_eliciting_packet[space] + duration
if (t < pto_timeout):
pto_timeout = t
pto_space = space
return pto_timeout, pto_space
PeerCompletedAddressValidation():
# Assume clients validate the server's address implicitly.
if (endpoint is server):
return true
# Servers complete address validation when a
# protected packet is received.
return has received Handshake ACK ||
has received 1-RTT ACK ||
has received HANDSHAKE_DONE
SetLossDetectionTimer():
earliest_loss_time, _ = GetLossTimeAndSpace()
if (earliest_loss_time != 0):
// Time threshold loss detection.
loss_detection_timer.update(earliest_loss_time)
return
if (server is at anti-amplification limit):
// The server's timer is not set if nothing can be sent.
loss_detection_timer.cancel()
return
if (no ack-eliciting packets in flight &&
PeerCompletedAddressValidation()):
// There is nothing to detect lost, so no timer is set.
// However, the client needs to arm the timer if the
// server might be blocked by the anti-amplification limit.
loss_detection_timer.cancel()
return
// Determine which PN space to arm PTO for.
timeout, _ = GetPtoTimeAndSpace()
loss_detection_timer.update(timeout)
A.9. On Timeout
When the loss detection timer expires, the timer's mode determines
the action to be performed.
Pseudocode for OnLossDetectionTimeout follows:
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OnLossDetectionTimeout():
earliest_loss_time, pn_space = GetLossTimeAndSpace()
if (earliest_loss_time != 0):
// Time threshold loss Detection
lost_packets = DetectLostPackets(pn_space)
assert(!lost_packets.empty())
OnPacketsLost(lost_packets)
SetLossDetectionTimer()
return
if (bytes_in_flight > 0):
// PTO. Send new data if available, else retransmit old data.
// If neither is available, send a single PING frame.
_, pn_space = GetPtoTimeAndSpace()
SendOneOrTwoAckElicitingPackets(pn_space)
else:
assert(endpoint is client without 1-RTT keys)
// Client sends an anti-deadlock packet: Initial is padded
// to earn more anti-amplification credit,
// a Handshake packet proves address ownership.
if (has Handshake keys):
SendOneAckElicitingHandshakePacket()
else:
SendOneAckElicitingPaddedInitialPacket()
pto_count++
SetLossDetectionTimer()
A.10. Detecting Lost Packets
DetectAndRemoveLostPackets is called every time an ACK is received or
the time threshold loss detection timer expires. This function
operates on the sent_packets for that packet number space and returns
a list of packets newly detected as lost.
Pseudocode for DetectAndRemoveLostPackets follows:
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DetectAndRemoveLostPackets(pn_space):
assert(largest_acked_packet[pn_space] != infinite)
loss_time[pn_space] = 0
lost_packets = {}
loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)
// Minimum time of kGranularity before packets are deemed lost.
loss_delay = max(loss_delay, kGranularity)
// Packets sent before this time are deemed lost.
lost_send_time = now() - loss_delay
foreach unacked in sent_packets[pn_space]:
if (unacked.packet_number > largest_acked_packet[pn_space]):
continue
// Mark packet as lost, or set time when it should be marked.
if (unacked.time_sent <= lost_send_time ||
largest_acked_packet[pn_space] >=
unacked.packet_number + kPacketThreshold):
sent_packets[pn_space].remove(unacked.packet_number)
if (unacked.in_flight):
lost_packets.insert(unacked)
else:
if (loss_time[pn_space] == 0):
loss_time[pn_space] = unacked.time_sent + loss_delay
else:
loss_time[pn_space] = min(loss_time[pn_space],
unacked.time_sent + loss_delay)
return lost_packets
Appendix B. Congestion Control Pseudocode
We now describe an example implementation of the congestion
controller described in Section 7.
B.1. Constants of interest
Constants used in congestion control are based on a combination of
RFCs, papers, and common practice.
kInitialWindow: Default limit on the initial bytes in flight as
described in Section 7.2.
kMinimumWindow: Minimum congestion window in bytes as described in
Section 7.2.
kLossReductionFactor: Reduction in congestion window when a new loss
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event is detected. The Section 7 section recommends a value is
0.5.
kPersistentCongestionThreshold: Period of time for persistent
congestion to be established, specified as a PTO multiplier. The
Section 7.8 section recommends a value of 3.
B.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
max_datagram_size: The sender's current maximum payload size. Does
not include UDP or IP overhead. The max datagram size is used for
congestion window computations. An endpoint sets the value of
this variable based on its PMTU (see Section 14.1 of
[QUIC-TRANSPORT]), with a minimum value of 1200 bytes.
ecn_ce_counters[kPacketNumberSpace]: The highest value reported for
the ECN-CE counter in the packet number space by the peer in an
ACK frame. This value is used to detect increases in the reported
ECN-CE counter.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one ack-eliciting or PADDING frame, and have
not been acked or declared lost. The size does not include IP or
UDP overhead, but does include the QUIC header and AEAD overhead.
Packets only containing ACK frames do not count towards
bytes_in_flight to ensure congestion control does not impede
congestion feedback.
congestion_window: Maximum number of bytes-in-flight that may be
sent.
congestion_recovery_start_time: The time when QUIC first detects
congestion due to loss or ECN, causing it to enter congestion
recovery. When a packet sent after this time is acknowledged,
QUIC exits congestion recovery.
ssthresh: Slow start threshold in bytes. When the congestion window
is below ssthresh, the mode is slow start and the window grows by
the number of bytes acknowledged.
B.3. Initialization
At the beginning of the connection, initialize the congestion control
variables as follows:
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congestion_window = kInitialWindow
bytes_in_flight = 0
congestion_recovery_start_time = 0
ssthresh = infinite
for pn_space in [ Initial, Handshake, ApplicationData ]:
ecn_ce_counters[pn_space] = 0
B.4. On Packet Sent
Whenever a packet is sent, and it contains non-ACK frames, the packet
increases bytes_in_flight.
OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent
B.5. On Packet Acknowledgement
Invoked from loss detection's OnAckReceived and is supplied with the
newly acked_packets from sent_packets.
InCongestionRecovery(sent_time):
return sent_time <= congestion_recovery_start_time
OnPacketsAcked(acked_packets):
for (packet in acked_packets):
// Remove from bytes_in_flight.
bytes_in_flight -= packet.size
if (InCongestionRecovery(packet.time_sent)):
// Do not increase congestion window in recovery period.
return
if (IsAppOrFlowControlLimited()):
// Do not increase congestion_window if application
// limited or flow control limited.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += packet.size
return
// Congestion avoidance.
congestion_window += max_datagram_size * acked_packet.size
/ congestion_window
B.6. On New Congestion Event
Invoked from ProcessECN and OnPacketsLost when a new congestion event
is detected. May start a new recovery period and reduces the
congestion window.
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CongestionEvent(sent_time):
// Start a new congestion event if packet was sent after the
// start of the previous congestion recovery period.
if (!InCongestionRecovery(sent_time)):
congestion_recovery_start_time = now()
congestion_window *= kLossReductionFactor
congestion_window = max(congestion_window, kMinimumWindow)
ssthresh = congestion_window
// A packet can be sent to speed up loss recovery.
MaybeSendOnePacket()
B.7. Process ECN Information
Invoked when an ACK frame with an ECN section is received from the
peer.
ProcessECN(ack, pn_space):
// If the ECN-CE counter reported by the peer has increased,
// this could be a new congestion event.
if (ack.ce_counter > ecn_ce_counters[pn_space]):
ecn_ce_counters[pn_space] = ack.ce_counter
CongestionEvent(sent_packets[ack.largest_acked].time_sent)
B.8. On Packets Lost
Invoked from DetectLostPackets when packets are deemed 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):
// Remove lost packets from bytes_in_flight.
for (lost_packet : lost_packets):
bytes_in_flight -= lost_packet.size
CongestionEvent(lost_packets.largest().time_sent)
// Collapse congestion window if persistent congestion
if (InPersistentCongestion(lost_packets)):
congestion_window = kMinimumWindow
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B.9. Upon dropping Initial or Handshake keys
When Initial or Handshake keys are discarded, packets from the space
are discarded and loss detection state is updated.
Pseudocode for OnPacketNumberSpaceDiscarded follows:
OnPacketNumberSpaceDiscarded(pn_space):
assert(pn_space != ApplicationData)
// Remove any unacknowledged packets from flight.
foreach packet in sent_packets[pn_space]:
if packet.in_flight
bytes_in_flight -= size
sent_packets[pn_space].clear()
// Reset the loss detection and PTO timer
time_of_last_ack_eliciting_packet[pn_space] = 0
loss_time[pn_space] = 0
pto_count = 0
SetLossDetectionTimer()
Appendix C. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
Issue and pull request numbers are listed with a leading octothorp.
C.1. Since draft-ietf-quic-recovery-28
* Refactored pseudocode to correct PTO calculation (#3564, #3674,
#3681)
C.2. Since draft-ietf-quic-recovery-27
* Added recommendations for speeding up handshake under some loss
conditions (#3078, #3080)
* PTO count is reset when handshake progress is made (#3272, #3415)
* PTO count is not reset by a client when the server might be
awaiting address validation (#3546, #3551)
* Recommend repairing losses immediately after entering the recovery
period (#3335, #3443)
* Clarified what loss conditions can be ignored during the handshake
(#3456, #3450)
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* Allow, but don't recommend, using RTT from previous connection to
seed RTT (#3464, #3496)
* Recommend use of adaptive loss detection thresholds (#3571, #3572)
C.3. Since draft-ietf-quic-recovery-26
No changes.
C.4. Since draft-ietf-quic-recovery-25
No significant changes.
C.5. Since draft-ietf-quic-recovery-24
* Require congestion control of some sort (#3247, #3244, #3248)
* Set a minimum reordering threshold (#3256, #3240)
* PTO is specific to a packet number space (#3067, #3074, #3066)
C.6. Since draft-ietf-quic-recovery-23
* Define under-utilizing the congestion window (#2630, #2686, #2675)
* PTO MUST send data if possible (#3056, #3057)
* Connection Close is not ack-eliciting (#3097, #3098)
* MUST limit bursts to the initial congestion window (#3160)
* Define the current max_datagram_size for congestion control
(#3041, #3167)
C.7. Since draft-ietf-quic-recovery-22
* PTO should always send an ack-eliciting packet (#2895)
* Unify the Handshake Timer with the PTO timer (#2648, #2658, #2886)
* Move ACK generation text to transport draft (#1860, #2916)
C.8. Since draft-ietf-quic-recovery-21
* No changes
C.9. Since draft-ietf-quic-recovery-20
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* Path validation can be used as initial RTT value (#2644, #2687)
* max_ack_delay transport parameter defaults to 0 (#2638, #2646)
* Ack Delay only measures intentional delays induced by the
implementation (#2596, #2786)
C.10. Since draft-ietf-quic-recovery-19
* Change kPersistentThreshold from an exponent to a multiplier
(#2557)
* Send a PING if the PTO timer fires and there's nothing to send
(#2624)
* Set loss delay to at least kGranularity (#2617)
* Merge application limited and sending after idle sections. Always
limit burst size instead of requiring resetting CWND to initial
CWND after idle (#2605)
* Rewrite RTT estimation, allow RTT samples where a newly acked
packet is ack-eliciting but the largest_acked is not (#2592)
* Don't arm the handshake timer if there is no handshake data
(#2590)
* Clarify that the time threshold loss alarm takes precedence over
the crypto handshake timer (#2590, #2620)
* Change initial RTT to 500ms to align with RFC6298 (#2184)
C.11. Since draft-ietf-quic-recovery-18
* Change IW byte limit to 14720 from 14600 (#2494)
* Update PTO calculation to match RFC6298 (#2480, #2489, #2490)
* Improve loss detection's description of multiple packet number
spaces and pseudocode (#2485, #2451, #2417)
* Declare persistent congestion even if non-probe packets are sent
and don't make persistent congestion more aggressive than RTO
verified was (#2365, #2244)
* Move pseudocode to the appendices (#2408)
* What to send on multiple PTOs (#2380)
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C.12. Since draft-ietf-quic-recovery-17
* After Probe Timeout discard in-flight packets or send another
(#2212, #1965)
* Endpoints discard initial keys as soon as handshake keys are
available (#1951, #2045)
* 0-RTT state is discarded when 0-RTT is rejected (#2300)
* Loss detection timer is cancelled when ack-eliciting frames are in
flight (#2117, #2093)
* Packets are declared lost if they are in flight (#2104)
* After becoming idle, either pace packets or reset the congestion
controller (#2138, 2187)
* Process ECN counts before marking packets lost (#2142)
* Mark packets lost before resetting crypto_count and pto_count
(#2208, #2209)
* Congestion and loss recovery state are discarded when keys are
discarded (#2327)
C.13. Since draft-ietf-quic-recovery-16
* Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP
and min crypto timeouts; eliminate timeout validation (#2114,
#2166, #2168, #1017)
* Redefine how congestion avoidance in terms of when the period
starts (#1928, #1930)
* Document what needs to be tracked for packets that are in flight
(#765, #1724, #1939)
* Integrate both time and packet thresholds into loss detection
(#1969, #1212, #934, #1974)
* Reduce congestion window after idle, unless pacing is used (#2007,
#2023)
* Disable RTT calculation for packets that don't elicit
acknowledgment (#2060, #2078)
* Limit ack_delay by max_ack_delay (#2060, #2099)
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* Initial keys are discarded once Handshake keys are available
(#1951, #2045)
* Reorder ECN and loss detection in pseudocode (#2142)
* Only cancel loss detection timer if ack-eliciting packets are in
flight (#2093, #2117)
C.14. Since draft-ietf-quic-recovery-14
* Used max_ack_delay from transport params (#1796, #1782)
* Merge ACK and ACK_ECN (#1783)
C.15. Since draft-ietf-quic-recovery-13
* Corrected the lack of ssthresh reduction in CongestionEvent
pseudocode (#1598)
* Considerations for ECN spoofing (#1426, #1626)
* Clarifications for PADDING and congestion control (#837, #838,
#1517, #1531, #1540)
* Reduce early retransmission timer to RTT/8 (#945, #1581)
* Packets are declared lost after an RTO is verified (#935, #1582)
C.16. Since draft-ietf-quic-recovery-12
* Changes to manage separate packet number spaces and encryption
levels (#1190, #1242, #1413, #1450)
* Added ECN feedback mechanisms and handling; new ACK_ECN frame
(#804, #805, #1372)
C.17. Since draft-ietf-quic-recovery-11
No significant changes.
C.18. Since draft-ietf-quic-recovery-10
* Improved text on ack generation (#1139, #1159)
* Make references to TCP recovery mechanisms informational (#1195)
* Define time_of_last_sent_handshake_packet (#1171)
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* Added signal from TLS the data it includes needs to be sent in a
Retry packet (#1061, #1199)
* Minimum RTT (min_rtt) is initialized with an infinite value
(#1169)
C.19. Since draft-ietf-quic-recovery-09
No significant changes.
C.20. Since draft-ietf-quic-recovery-08
* Clarified pacing and RTO (#967, #977)
C.21. Since draft-ietf-quic-recovery-07
* Include Ack Delay in RTO(and TLP) computations (#981)
* Ack Delay in SRTT computation (#961)
* Default RTT and Slow Start (#590)
* Many editorial fixes.
C.22. Since draft-ietf-quic-recovery-06
No significant changes.
C.23. Since draft-ietf-quic-recovery-05
* Add more congestion control text (#776)
C.24. Since draft-ietf-quic-recovery-04
No significant changes.
C.25. Since draft-ietf-quic-recovery-03
No significant changes.
C.26. Since draft-ietf-quic-recovery-02
* Integrate F-RTO (#544, #409)
* Add congestion control (#545, #395)
* Require connection abort if a skipped packet was acknowledged
(#415)
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* Simplify RTO calculations (#142, #417)
C.27. Since draft-ietf-quic-recovery-01
* Overview added to loss detection
* Changes initial default RTT to 100ms
* Added time-based loss detection and fixes early retransmit
* Clarified loss recovery for handshake packets
* Fixed references and made TCP references informative
C.28. Since draft-ietf-quic-recovery-00
* Improved description of constants and ACK behavior
C.29. Since draft-iyengar-quic-loss-recovery-01
* Adopted as base for draft-ietf-quic-recovery
* Updated authors/editors list
* Added table of contents
Appendix D. Contributors
The IETF QUIC Working Group received an enormous amount of support
from many people. The following people provided substantive
contributions to this document: Alessandro Ghedini, Benjamin
Saunders, Gorry Fairhurst, 奥 一穂 (Kazuho Oku), Lars Eggert, Magnus
Westerlund, Marten Seemann, Martin Duke, Martin Thomson, Nick Banks,
Praveen Balasubramaniam.
Acknowledgments
Authors' Addresses
Jana Iyengar (editor)
Fastly
Email: jri.ietf@gmail.com
Ian Swett (editor)
Google
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Email: ianswett@google.com
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