QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-26
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 | 2020-02-21 (Latest revision 2020-01-21) | ||
| Replaces | draft-iyengar-quic-loss-recovery | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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draft-ietf-quic-recovery-26
QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track I. Swett, Ed.
Expires: 24 August 2020 Google
21 February 2020
QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-26
Abstract
This document describes loss detection and congestion control
mechanisms for QUIC.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic
(https://mailarchive.ietf.org/arch/search/?email_list=quic).
Working Group information can be found at https://github.com/quicwg
(https://github.com/quicwg); source code and issues list for this
draft can be found at https://github.com/quicwg/base-drafts/labels/-
recovery (https://github.com/quicwg/base-drafts/labels/-recovery).
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 24 August 2020.
Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document. Code Components
extracted from this document must include Simplified BSD License text
as described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 4
3. Design of the QUIC Transmission Machinery . . . . . . . . . . 5
3.1. Relevant Differences Between QUIC and TCP . . . . . . . . 5
3.1.1. Separate Packet Number Spaces . . . . . . . . . . . . 6
3.1.2. Monotonically Increasing Packet Numbers . . . . . . . 6
3.1.3. Clearer Loss Epoch . . . . . . . . . . . . . . . . . 6
3.1.4. No Reneging . . . . . . . . . . . . . . . . . . . . . 7
3.1.5. More ACK Ranges . . . . . . . . . . . . . . . . . . . 7
3.1.6. Explicit Correction For Delayed Acknowledgements . . 7
4. Estimating the Round-Trip Time . . . . . . . . . . . . . . . 7
4.1. Generating RTT samples . . . . . . . . . . . . . . . . . 7
4.2. Estimating min_rtt . . . . . . . . . . . . . . . . . . . 8
4.3. Estimating smoothed_rtt and rttvar . . . . . . . . . . . 9
5. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 10
5.1. Acknowledgement-based Detection . . . . . . . . . . . . . 10
5.1.1. Packet Threshold . . . . . . . . . . . . . . . . . . 11
5.1.2. Time Threshold . . . . . . . . . . . . . . . . . . . 11
5.2. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 12
5.2.1. Computing PTO . . . . . . . . . . . . . . . . . . . . 12
5.3. Handshakes and New Paths . . . . . . . . . . . . . . . . 13
5.3.1. Sending Probe Packets . . . . . . . . . . . . . . . . 14
5.3.2. Loss Detection . . . . . . . . . . . . . . . . . . . 15
5.4. Handling Retry Packets . . . . . . . . . . . . . . . . . 15
5.5. Discarding Keys and Packet State . . . . . . . . . . . . 15
6. Congestion Control . . . . . . . . . . . . . . . . . . . . . 16
6.1. Explicit Congestion Notification . . . . . . . . . . . . 16
6.2. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 17
6.3. Congestion Avoidance . . . . . . . . . . . . . . . . . . 17
6.4. Recovery Period . . . . . . . . . . . . . . . . . . . . . 17
6.5. Ignoring Loss of Undecryptable Packets . . . . . . . . . 17
6.6. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 18
6.7. Persistent Congestion . . . . . . . . . . . . . . . . . . 18
6.8. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 19
6.9. Under-utilizing the Congestion Window . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . 20
7.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 20
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7.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 20
7.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 20
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 21
9.1. Normative References . . . . . . . . . . . . . . . . . . 21
9.2. Informative References . . . . . . . . . . . . . . . . . 21
Appendix A. Loss Recovery Pseudocode . . . . . . . . . . . . . . 23
A.1. Tracking Sent Packets . . . . . . . . . . . . . . . . . . 23
A.1.1. Sent Packet Fields . . . . . . . . . . . . . . . . . 24
A.2. Constants of interest . . . . . . . . . . . . . . . . . . 24
A.3. Variables of interest . . . . . . . . . . . . . . . . . . 25
A.4. Initialization . . . . . . . . . . . . . . . . . . . . . 25
A.5. On Sending a Packet . . . . . . . . . . . . . . . . . . . 26
A.6. On Receiving an Acknowledgment . . . . . . . . . . . . . 26
A.7. On Packet Acknowledgment . . . . . . . . . . . . . . . . 28
A.8. Setting the Loss Detection Timer . . . . . . . . . . . . 28
A.9. On Timeout . . . . . . . . . . . . . . . . . . . . . . . 30
A.10. Detecting Lost Packets . . . . . . . . . . . . . . . . . 30
Appendix B. Congestion Control Pseudocode . . . . . . . . . . . 31
B.1. Constants of interest . . . . . . . . . . . . . . . . . . 31
B.2. Variables of interest . . . . . . . . . . . . . . . . . . 32
B.3. Initialization . . . . . . . . . . . . . . . . . . . . . 33
B.4. On Packet Sent . . . . . . . . . . . . . . . . . . . . . 33
B.5. On Packet Acknowledgement . . . . . . . . . . . . . . . . 33
B.6. On New Congestion Event . . . . . . . . . . . . . . . . . 34
B.7. Process ECN Information . . . . . . . . . . . . . . . . . 34
B.8. On Packets Lost . . . . . . . . . . . . . . . . . . . . . 35
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 35
C.1. Since draft-ietf-quic-recovery-25 . . . . . . . . . . . . 35
C.2. Since draft-ietf-quic-recovery-24 . . . . . . . . . . . . 35
C.3. Since draft-ietf-quic-recovery-23 . . . . . . . . . . . . 35
C.4. Since draft-ietf-quic-recovery-22 . . . . . . . . . . . . 36
C.5. Since draft-ietf-quic-recovery-21 . . . . . . . . . . . . 36
C.6. Since draft-ietf-quic-recovery-20 . . . . . . . . . . . . 36
C.7. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . . 36
C.8. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . . 37
C.9. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . . 37
C.10. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . . 38
C.11. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 38
C.12. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 38
C.13. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 39
C.14. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 39
C.15. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 39
C.16. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 39
C.17. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 39
C.18. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 39
C.19. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 40
C.20. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 40
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C.21. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 40
C.22. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 40
C.23. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 40
C.24. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 40
C.25. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 40
C.26. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 41
Appendix D. Contributors . . . . . . . . . . . . . . . . . . . . 41
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 41
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 41
1. Introduction
QUIC is a new multiplexed and secure transport atop UDP. QUIC builds
on decades of transport and security experience, and implements
mechanisms that make it attractive as a modern general-purpose
transport. The QUIC protocol is described in [QUIC-TRANSPORT].
QUIC implements the spirit of existing TCP congestion control and
loss recovery mechanisms, described in RFCs, various Internet-drafts,
and also those prevalent in the Linux TCP implementation. This
document describes QUIC congestion control and loss recovery, and
where applicable, attributes the TCP equivalent in RFCs, Internet-
drafts, academic papers, and/or TCP implementations.
2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
Definitions of terms that are used in this document:
Ack-eliciting Frames: All frames other than ACK, PADDING, and
CONNECTION_CLOSE are considered ack-eliciting.
Ack-eliciting Packets: Packets that contain ack-eliciting frames
elicit an ACK from the receiver within the maximum ack delay and
are called ack-eliciting packets.
In-flight: Packets are considered in-flight when they are ack-
eliciting or contain a PADDING frame, and they have been sent but
are not acknowledged, declared lost, or abandoned along with old
keys.
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3. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
indicates the encryption level and includes a packet sequence number
(referred to below as a packet number). The encryption level
indicates the packet number space, as described in [QUIC-TRANSPORT].
Packet numbers never repeat within a packet number space for the
lifetime of a connection. Packet numbers are sent in monotonically
increasing order within a space, preventing ambiguity.
This design obviates the need for disambiguating between
transmissions and retransmissions and eliminates significant
complexity from QUIC's interpretation of TCP loss detection
mechanisms.
QUIC packets can contain multiple frames of different types. The
recovery mechanisms ensure that data and frames that need reliable
delivery are acknowledged or declared lost and sent in new packets as
necessary. The types of frames contained in a packet affect recovery
and congestion control logic:
* All packets are acknowledged, though packets that contain no ack-
eliciting frames are only acknowledged along with ack-eliciting
packets.
* Long header packets that contain CRYPTO frames are critical to the
performance of the QUIC handshake and use shorter timers for
acknowledgement.
* Packets containing frames besides ACK or CONNECTION_CLOSE frames
count toward congestion control limits and are considered in-
flight.
* PADDING frames cause packets to contribute toward bytes in flight
without directly causing an acknowledgment to be sent.
3.1. Relevant Differences Between QUIC and TCP
Readers familiar with TCP's loss detection and congestion control
will find algorithms here that parallel well-known TCP ones.
Protocol differences between QUIC and TCP however contribute to
algorithmic differences. We briefly describe these protocol
differences below.
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3.1.1. Separate Packet Number Spaces
QUIC uses separate packet number spaces for each encryption level,
except 0-RTT and all generations of 1-RTT keys use the same packet
number space. Separate packet number spaces ensures acknowledgement
of packets sent with one level of encryption will not cause spurious
retransmission of packets sent with a different encryption level.
Congestion control and round-trip time (RTT) measurement are unified
across packet number spaces.
3.1.2. Monotonically Increasing Packet Numbers
TCP conflates transmission order at the sender with delivery order at
the receiver, which results in retransmissions of the same data
carrying the same sequence number, and consequently leads to
"retransmission ambiguity". QUIC separates the two. QUIC uses a
packet number to indicate transmission order. Application data is
sent in one or more streams and delivery order is determined by
stream offsets encoded within STREAM frames.
QUIC's packet number is strictly increasing within a packet number
space, and directly encodes transmission order. A higher packet
number signifies that the packet was sent later, and a lower packet
number signifies that the packet was sent earlier. When a packet
containing ack-eliciting frames is detected lost, QUIC rebundles
necessary frames in a new packet with a new packet number, removing
ambiguity about which packet is acknowledged when an ACK is received.
Consequently, more accurate RTT measurements can be made, spurious
retransmissions are trivially detected, and mechanisms such as Fast
Retransmit can be applied universally, based only on packet number.
This design point significantly simplifies loss detection mechanisms
for QUIC. Most TCP mechanisms implicitly attempt to infer
transmission ordering based on TCP sequence numbers - a non-trivial
task, especially when TCP timestamps are not available.
3.1.3. Clearer Loss Epoch
QUIC starts a loss epoch when a packet is lost and ends one when any
packet sent after the epoch starts is acknowledged. TCP waits for
the gap in the sequence number space to be filled, and so if a
segment is lost multiple times in a row, the loss epoch may not end
for several round trips. Because both should reduce their congestion
windows only once per epoch, QUIC will do it once for every round
trip that experiences loss, while TCP may only do it once across
multiple round trips.
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3.1.4. No Reneging
QUIC ACKs contain information that is similar to TCP SACK, but QUIC
does not allow any acked packet to be reneged, greatly simplifying
implementations on both sides and reducing memory pressure on the
sender.
3.1.5. More ACK Ranges
QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In
high loss environments, this speeds recovery, reduces spurious
retransmits, and ensures forward progress without relying on
timeouts.
3.1.6. Explicit Correction For Delayed Acknowledgements
QUIC endpoints measure the delay incurred between when a packet is
received and when the corresponding acknowledgment is sent, allowing
a peer to maintain a more accurate round-trip time estimate (see
Section 13.2 of [QUIC-TRANSPORT]).
4. Estimating the Round-Trip Time
At a high level, an endpoint measures the time from when a packet was
sent to when it is acknowledged as a round-trip time (RTT) sample.
The endpoint uses RTT samples and peer-reported host delays (see
Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical
description of the network path's RTT. An endpoint computes the
following three values for each path: the minimum value observed over
the lifetime of the path (min_rtt), an exponentially-weighted moving
average (smoothed_rtt), and the mean deviation (referred to as
"variation" in the rest of this document) in the observed RTT samples
(rttvar).
4.1. Generating RTT samples
An endpoint generates an RTT sample on receiving an ACK frame that
meets the following two conditions:
* the largest acknowledged packet number is newly acknowledged, and
* at least one of the newly acknowledged packets was ack-eliciting.
The RTT sample, latest_rtt, is generated as the time elapsed since
the largest acknowledged packet was sent:
latest_rtt = ack_time - send_time_of_largest_acked
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An RTT sample is generated using only the largest acknowledged packet
in the received ACK frame. This is because a peer reports ACK delays
for only the largest acknowledged packet in an ACK frame. While the
reported ACK delay is not used by the RTT sample measurement, it is
used to adjust the RTT sample in subsequent computations of
smoothed_rtt and rttvar Section 4.3.
To avoid generating multiple RTT samples for a single packet, an ACK
frame SHOULD NOT be used to update RTT estimates if it does not newly
acknowledge the largest acknowledged packet.
An RTT sample MUST NOT be generated on receiving an ACK frame that
does not newly acknowledge at least one ack-eliciting packet. A peer
usually does not send an ACK frame when only non-ack-eliciting
packets are received. Therefore an ACK frame that contains
acknowledgements for only non-ack-eliciting packets could include an
arbitrarily large Ack Delay value. Ignoring such ACK frames avoids
complications in subsequent smoothed_rtt and rttvar computations.
A sender might generate multiple RTT samples per RTT when multiple
ACK frames are received within an RTT. As suggested in [RFC6298],
doing so might result in inadequate history in smoothed_rtt and
rttvar. Ensuring that RTT estimates retain sufficient history is an
open research question.
4.2. Estimating min_rtt
min_rtt is the minimum RTT observed for a given network path.
min_rtt is set to the latest_rtt on the first RTT sample, and to the
lesser of min_rtt and latest_rtt on subsequent samples. In this
document, min_rtt is used by loss detection to reject implausibly
small rtt samples.
An endpoint uses only locally observed times in computing the min_rtt
and does not adjust for ACK delays reported by the peer. Doing so
allows the endpoint to set a lower bound for the smoothed_rtt based
entirely on what it observes (see Section 4.3), and limits potential
underestimation due to erroneously-reported delays by the peer.
The RTT for a network path may change over time. If a path's actual
RTT decreases, the min_rtt will adapt immediately on the first low
sample. If the path's actual RTT increases, the min_rtt will not
adapt to it, allowing future RTT samples that are smaller than the
new RTT be included in smoothed_rtt.
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4.3. Estimating smoothed_rtt and rttvar
smoothed_rtt is an exponentially-weighted moving average of an
endpoint's RTT samples, and rttvar is the variation in the RTT
samples, estimated using a mean variation.
The calculation of smoothed_rtt uses path latency after adjusting RTT
samples for acknowledgement delays. These delays are computed using
the ACK Delay field of the ACK frame as described in Section 19.3 of
[QUIC-TRANSPORT]. For packets sent in the ApplicationData packet
number space, a peer limits any delay in sending an acknowledgement
for an ack-eliciting packet to no greater than the value it
advertised in the max_ack_delay transport parameter. Consequently,
when a peer reports an Ack Delay that is greater than its
max_ack_delay, the delay is attributed to reasons out of the peer's
control, such as scheduler latency at the peer or loss of previous
ACK frames. Any delays beyond the peer's max_ack_delay are therefore
considered effectively part of path delay and incorporated into the
smoothed_rtt estimate.
When adjusting an RTT sample using peer-reported acknowledgement
delays, an endpoint:
* MUST ignore the Ack Delay field of the ACK frame for packets sent
in the Initial and Handshake packet number space.
* MUST use the lesser of the value reported in Ack Delay field of
the ACK frame and the peer's max_ack_delay transport parameter.
* MUST NOT apply the adjustment if the resulting RTT sample is
smaller than the min_rtt. This limits the underestimation that a
misreporting peer can cause to the smoothed_rtt.
On the first RTT sample for a network path, the smoothed_rtt is set
to the latest_rtt.
smoothed_rtt and rttvar are computed as follows, similar to
[RFC6298]. On the first RTT sample for a network path:
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:
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ack_delay = min(Ack Delay in ACK Frame, max_ack_delay)
adjusted_rtt = latest_rtt
if (min_rtt + ack_delay < latest_rtt):
adjusted_rtt = latest_rtt - ack_delay
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
5. Loss Detection
QUIC senders use acknowledgements to detect lost packets, and a probe
time out (see Section 5.2) to ensure acknowledgements are received.
This section provides a description of these algorithms.
If a packet is lost, the QUIC transport needs to recover from that
loss, such as by retransmitting the data, sending an updated frame,
or abandoning the frame. For more information, see Section 13.3 of
[QUIC-TRANSPORT].
5.1. Acknowledgement-based Detection
Acknowledgement-based loss detection implements the spirit of TCP's
Fast Retransmit [RFC5681], Early Retransmit [RFC5827], FACK [FACK],
SACK loss recovery [RFC6675], and RACK [RACK]. This section provides
an overview of how these algorithms are implemented in QUIC.
A packet is declared lost if it meets all the following conditions:
* The packet is unacknowledged, in-flight, and was sent prior to an
acknowledged packet.
* Either its packet number is kPacketThreshold smaller than an
acknowledged packet (Section 5.1.1), or it was sent long enough in
the past (Section 5.1.2).
The acknowledgement indicates that a packet sent later was delivered,
and the packet and time thresholds provide some tolerance for packet
reordering.
Spuriously declaring packets as lost leads to unnecessary
retransmissions and may result in degraded performance due to the
actions of the congestion controller upon detecting loss.
Implementations that detect spurious retransmissions and increase the
reordering threshold in packets or time MAY choose to start with
smaller initial reordering thresholds to minimize recovery latency.
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5.1.1. Packet Threshold
The RECOMMENDED initial value for the packet reordering threshold
(kPacketThreshold) is 3, based on best practices for TCP loss
detection [RFC5681] [RFC6675]. Implementations SHOULD NOT use a
packet threshold less than 3, to keep in line with TCP [RFC5681].
Some networks may exhibit higher degrees of reordering, causing a
sender to detect spurious losses. Implementers MAY use algorithms
developed for TCP, such as TCP-NCR [RFC4653], to improve QUIC's
reordering resilience.
5.1.2. Time Threshold
Once a later packet within the same packet number space has been
acknowledged, an endpoint SHOULD declare an earlier packet lost if it
was sent a threshold amount of time in the past. To avoid declaring
packets as lost too early, this time threshold MUST be set to at
least kGranularity. The time threshold is:
max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity)
If packets sent prior to the largest acknowledged packet cannot yet
be declared lost, then a timer SHOULD be set for the remaining time.
Using max(smoothed_rtt, latest_rtt) protects from the two following
cases:
* the latest RTT sample is lower than the smoothed RTT, perhaps due
to reordering where the acknowledgement encountered a shorter
path;
* the latest RTT sample is higher than the smoothed RTT, perhaps due
to a sustained increase in the actual RTT, but the smoothed RTT
has not yet caught up.
The RECOMMENDED time threshold (kTimeThreshold), expressed as a
round-trip time multiplier, is 9/8.
Implementations MAY experiment with absolute thresholds, thresholds
from previous connections, adaptive thresholds, or including RTT
variation. Smaller thresholds reduce reordering resilience and
increase spurious retransmissions, and larger thresholds increase
loss detection delay.
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5.2. Probe Timeout
A Probe Timeout (PTO) triggers sending one or two probe datagrams
when ack-eliciting packets are not acknowledged within the expected
period of time or the handshake has not been completed. A PTO
enables a connection to recover from loss of tail packets or
acknowledgements.
As with loss detection, the probe timeout is per packet number space.
The PTO algorithm used in QUIC implements the reliability functions
of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for
TCP [RFC5682]. The timeout computation is based on TCP's
retransmission timeout period [RFC6298].
5.2.1. Computing PTO
When an ack-eliciting packet is transmitted, the sender schedules a
timer for the PTO period as follows:
PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay
kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in
Appendix A.2 and Appendix A.3.
The PTO period is the amount of time that a sender ought to wait for
an acknowledgement of a sent packet. This time period includes the
estimated network roundtrip-time (smoothed_rtt), the variation in the
estimate (4*rttvar), and max_ack_delay, to account for the maximum
time by which a receiver might delay sending an acknowledgement.
When the PTO is armed for Initial or Handshake packet number spaces,
the max_ack_delay is 0, as specified in 13.2.1 of [QUIC-TRANSPORT].
The PTO value MUST be set to at least kGranularity, to avoid the
timer expiring immediately.
A sender computes its PTO timer every time an ack-eliciting packet is
sent. When ack-eliciting packets are in-flight in multiple packet
number spaces, the timer MUST be set for the packet number space with
the earliest timeout, except for ApplicationData, which MUST be
ignored until the handshake completes; see Section 4.1.1 of
[QUIC-TLS]. Not arming the PTO for ApplicationData prioritizes
completing the handshake and prevents the server from sending a 1-RTT
packet on a PTO before before it has the keys to process a 1-RTT
packet.
When a PTO timer expires, the PTO period MUST be set to twice its
current value. This exponential reduction in the sender's rate is
important because consecutive PTOs might be caused by loss of packets
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or acknowledgements due to severe congestion. Even when there are
ack-eliciting packets in-flight in multiple packet number spaces, the
exponential increase in probe timeout occurs across all spaces to
prevent excess load on the network. For example, a timeout in the
Initial packet number space doubles the length of the timeout in the
Handshake packet number space.
The life of a connection that is experiencing consecutive PTOs is
limited by the endpoint's idle timeout.
The probe timer MUST NOT be set if the time threshold Section 5.1.2
loss detection timer is set. The time threshold loss detection timer
is expected to both expire earlier than the PTO and be less likely to
spuriously retransmit data.
5.3. Handshakes and New Paths
The initial probe timeout for a new connection or new path SHOULD be
set to twice the initial RTT. Resumed connections over the same
network SHOULD use the previous connection's final smoothed RTT value
as the resumed connection's initial RTT. If no previous RTT is
available, the initial RTT SHOULD be set to 500ms, resulting in a 1
second initial timeout as recommended in [RFC6298].
A connection MAY use the delay between sending a PATH_CHALLENGE and
receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in
Appendix A.2) for a new path, but the delay SHOULD NOT be considered
an RTT sample.
Until the server has validated the client's address on the path, the
amount of data it can send is limited to three times the amount of
data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If
no data can be sent, then the PTO alarm MUST NOT be armed until
datagrams have been received from the client.
Since the server could be blocked until more packets are received
from the client, it is the client's responsibility to send packets to
unblock the server until it is certain that the server has finished
its address validation (see Section 8 of [QUIC-TRANSPORT]). That is,
the client MUST set the probe timer if the client has not received an
acknowledgement for one of its Handshake or 1-RTT packets.
Prior to handshake completion, when few to none RTT samples have been
generated, it is possible that the probe timer expiration is due to
an incorrect RTT estimate at the client. To allow the client to
improve its RTT estimate, the new packet that it sends MUST be ack-
eliciting. If Handshake keys are available to the client, it MUST
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send a Handshake packet, and otherwise it MUST send an Initial packet
in a UDP datagram of at least 1200 bytes.
Initial packets and Handshake packets could be never acknowledged,
but they are removed from bytes in flight when the Initial and
Handshake keys are discarded.
5.3.1. Sending Probe Packets
When a PTO timer expires, a sender MUST send at least one ack-
eliciting packet in the packet number space as a probe, unless there
is no data available to send. An endpoint MAY send up to two full-
sized datagrams containing ack-eliciting packets, to avoid an
expensive consecutive PTO expiration due to a single lost datagram or
transmit data from multiple packet number spaces.
In addition to sending data in the packet number space for which the
timer expired, the sender SHOULD send ack-eliciting packets from
other packet number spaces with in-flight data, coalescing packets if
possible.
When the PTO timer expires, and there is new or previously sent
unacknowledged data, it MUST be sent.
It is possible the sender has no new or previously-sent data to send.
As an example, consider the following sequence of events: new
application data is sent in a STREAM frame, deemed lost, then
retransmitted in a new packet, and then the original transmission is
acknowledged. When there is no data to send, the sender SHOULD send
a PING or other ack-eliciting frame in a single packet, re-arming the
PTO timer.
Alternatively, instead of sending an ack-eliciting packet, the sender
MAY mark any packets still in flight as lost. Doing so avoids
sending an additional packet, but increases the risk that loss is
declared too aggressively, resulting in an unnecessary rate reduction
by the congestion controller.
Consecutive PTO periods increase exponentially, and as a result,
connection recovery latency increases exponentially as packets
continue to be dropped in the network. Sending two packets on PTO
expiration increases resilience to packet drops, thus reducing the
probability of consecutive PTO events.
Probe packets sent on a PTO MUST be ack-eliciting. A probe packet
SHOULD carry new data when possible. A probe packet MAY carry
retransmitted unacknowledged data when new data is unavailable, when
flow control does not permit new data to be sent, or to
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opportunistically reduce loss recovery delay. Implementations MAY
use alternative strategies for determining the content of probe
packets, including sending new or retransmitted data based on the
application's priorities.
When the PTO timer expires multiple times and new data cannot be
sent, implementations must choose between sending the same payload
every time or sending different payloads. Sending the same payload
may be simpler and ensures the highest priority frames arrive first.
Sending different payloads each time reduces the chances of spurious
retransmission.
5.3.2. Loss Detection
Delivery or loss of packets in flight is established when an ACK
frame is received that newly acknowledges one or more packets.
A PTO timer expiration event does not indicate packet loss and MUST
NOT cause prior unacknowledged packets to be marked as lost. When an
acknowledgement is received that newly acknowledges packets, loss
detection proceeds as dictated by packet and time threshold
mechanisms; see Section 5.1.
5.4. Handling Retry Packets
A Retry packet causes a client to send another Initial packet,
effectively restarting the connection process. A Retry packet
indicates that the Initial was received, but not processed. A Retry
packet cannot be treated as an acknowledgment, because it does not
indicate that a packet was processed or specify the packet number.
Clients that receive a Retry packet reset congestion control and loss
recovery state, including resetting any pending timers. Other
connection state, in particular cryptographic handshake messages, is
retained; see Section 17.2.5 of [QUIC-TRANSPORT].
The client MAY compute an RTT estimate to the server as the time
period from when the first Initial was sent to when a Retry or a
Version Negotiation packet is received. The client MAY use this
value in place of its default for the initial RTT estimate.
5.5. Discarding Keys and Packet State
When packet protection keys are discarded (see Section 4.10 of
[QUIC-TLS]), all packets that were sent with those keys can no longer
be acknowledged because their acknowledgements cannot be processed
anymore. The sender MUST discard all recovery state associated with
those packets and MUST remove them from the count of bytes in flight.
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Endpoints stop sending and receiving Initial packets once they start
exchanging Handshake packets (see Section 17.2.2.1 of
[QUIC-TRANSPORT]). At this point, recovery state for all in-flight
Initial packets is discarded.
When 0-RTT is rejected, recovery state for all in-flight 0-RTT
packets is discarded.
If a server accepts 0-RTT, but does not buffer 0-RTT packets that
arrive before Initial packets, early 0-RTT packets will be declared
lost, but that is expected to be infrequent.
It is expected that keys are discarded after packets encrypted with
them would be acknowledged or declared lost. Initial secrets however
might be destroyed sooner, as soon as handshake keys are available
(see Section 4.10.1 of [QUIC-TLS]).
6. Congestion Control
This document specifies a Reno congestion controller for QUIC
[RFC6582].
The signals QUIC provides for congestion control are generic and are
designed to support different algorithms. Endpoints can unilaterally
choose a different algorithm to use, such as Cubic [RFC8312].
If an endpoint uses a different controller than that specified in
this document, the chosen controller MUST conform to the congestion
control guidelines specified in Section 3.1 of [RFC8085].
The algorithm in this document specifies and uses the controller's
congestion window in bytes.
An endpoint MUST NOT send a packet if it would cause bytes_in_flight
(see Appendix B.2) to be larger than the congestion window, unless
the packet is sent on a PTO timer expiration (see Section 5.2).
6.1. Explicit Congestion Notification
If a path has been verified to support ECN [RFC3168] [RFC8311], QUIC
treats a Congestion Experienced(CE) codepoint in the IP header as a
signal of congestion. This document specifies an endpoint's response
when its peer receives packets with the Congestion Experienced
codepoint.
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6.2. Slow Start
QUIC begins every connection in slow start and exits slow start upon
loss or upon increase in the ECN-CE counter. QUIC re-enters slow
start any time the congestion window is less than ssthresh, which
only occurs after persistent congestion is declared. While in slow
start, QUIC increases the congestion window by the number of bytes
acknowledged when each acknowledgment is processed.
6.3. Congestion Avoidance
Slow start exits to congestion avoidance. Congestion avoidance in
NewReno uses an additive increase multiplicative decrease (AIMD)
approach that increases the congestion window by one maximum packet
size per congestion window acknowledged. When a loss is detected,
NewReno halves the congestion window and sets the slow start
threshold to the new congestion window.
6.4. Recovery Period
A recovery period is entered when loss or ECN-CE marking of a packet
is detected. A recovery period ends when a packet sent during the
recovery period is acknowledged. This is slightly different from
TCP's definition of recovery, which ends when the lost packet that
started recovery is acknowledged.
The recovery period limits congestion window reduction to once per
round trip. During recovery, the congestion window remains unchanged
irrespective of new losses or increases in the ECN-CE counter.
6.5. Ignoring Loss of Undecryptable Packets
During the handshake, some packet protection keys might not be
available when a packet arrives. In particular, Handshake and 0-RTT
packets cannot be processed until the Initial packets arrive, and
1-RTT packets cannot be processed until the handshake completes.
Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets
that might arrive before the peer has packet protection keys to
process those packets.
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6.6. Probe Timeout
Probe packets MUST NOT be blocked by the congestion controller. A
sender MUST however count these packets as being additionally in
flight, since these packets add network load without establishing
packet loss. Note that sending probe packets might cause the
sender's bytes in flight to exceed the congestion window until an
acknowledgement is received that establishes loss or delivery of
packets.
6.7. Persistent Congestion
When an ACK frame is received that establishes loss of all in-flight
packets sent over a long enough period of time, the network is
considered to be experiencing persistent congestion. Commonly, this
can be established by consecutive PTOs, but since the PTO timer is
reset when a new ack-eliciting packet is sent, an explicit duration
must be used to account for those cases where PTOs do not occur or
are substantially delayed. This duration is computed as follows:
(smoothed_rtt + 4 * rttvar + max_ack_delay) *
kPersistentCongestionThreshold
For example, assume:
smoothed_rtt = 1 rttvar = 0 max_ack_delay = 0
kPersistentCongestionThreshold = 3
If an ack-eliciting packet is sent at time = 0, the following
scenario would illustrate persistent congestion:
+-----+------------------------+
| t=0 | Send Pkt #1 (App Data) |
+=====+========================+
| t=1 | Send Pkt #2 (PTO 1) |
+-----+------------------------+
| t=3 | Send Pkt #3 (PTO 2) |
+-----+------------------------+
| t=7 | Send Pkt #4 (PTO 3) |
+-----+------------------------+
| t=8 | Recv ACK of Pkt #4 |
+-----+------------------------+
Table 1
The first three packets are determined to be lost when the
acknowlegement of packet 4 is received at t=8. The congestion period
is calculated as the time between the oldest and newest lost packets:
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(3 - 0) = 3. The duration for persistent congestion is equal to: (1
* kPersistentCongestionThreshold) = 3. Because the threshold was
reached and because none of the packets between the oldest and the
newest packets are acknowledged, the network is considered to have
experienced persistent congestion.
When persistent congestion is established, the sender's congestion
window MUST be reduced to the minimum congestion window
(kMinimumWindow). This response of collapsing the congestion window
on persistent congestion is functionally similar to a sender's
response on a Retransmission Timeout (RTO) in TCP [RFC5681] after
Tail Loss Probes (TLP) [RACK].
6.8. Pacing
This document does not specify a pacer, but it is RECOMMENDED that a
sender pace sending of all in-flight packets based on input from the
congestion controller. For example, a pacer might distribute the
congestion window over the smoothed RTT when used with a window-based
controller, and a pacer might use the rate estimate of a rate-based
controller.
An implementation should take care to architect its congestion
controller to work well with a pacer. For instance, a pacer might
wrap the congestion controller and control the availability of the
congestion window, or a pacer might pace out packets handed to it by
the congestion controller. Timely delivery of ACK frames is
important for efficient loss recovery. Packets containing only ACK
frames should therefore not be paced, to avoid delaying their
delivery to the peer.
Sending multiple packets into the network without any delay between
them creates a packet burst that might cause short-term congestion
and losses. Implementations MUST either use pacing or limit such
bursts to the initial congestion window, which is recommended to be
the minimum of 10 * max_datagram_size and max(2* max_datagram_size,
14720)), where max_datagram_size is the current maximum size of a
datagram for the connection, not including UDP or IP overhead.
As an example of a well-known and publicly available implementation
of a flow pacer, implementers are referred to the Fair Queue packet
scheduler (fq qdisc) in Linux (3.11 onwards).
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6.9. Under-utilizing the Congestion Window
When bytes in flight is smaller than the congestion window and
sending is not pacing limited, the congestion window is under-
utilized. When this occurs, the congestion window SHOULD NOT be
increased in either slow start or congestion avoidance. This can
happen due to insufficient application data or flow control credit.
A sender MAY use the pipeACK method described in section 4.3 of
[RFC7661] to determine if the congestion window is sufficiently
utilized.
A sender that paces packets (see Section 6.8) might delay sending
packets and not fully utilize the congestion window due to this
delay. A sender should not consider itself application limited if it
would have fully utilized the congestion window without pacing delay.
A sender MAY implement alternative mechanisms to update its
congestion window after periods of under-utilization, such as those
proposed for TCP in [RFC7661].
7. Security Considerations
7.1. Congestion Signals
Congestion control fundamentally involves the consumption of signals
- both loss and ECN codepoints - from unauthenticated entities. On-
path attackers can spoof or alter these signals. An attacker can
cause endpoints to reduce their sending rate by dropping packets, or
alter send rate by changing ECN codepoints.
7.2. Traffic Analysis
Packets that carry only ACK frames can be heuristically identified by
observing packet size. Acknowledgement patterns may expose
information about link characteristics or application behavior.
Endpoints can use PADDING frames or bundle acknowledgments with other
frames to reduce leaked information.
7.3. Misreporting ECN Markings
A receiver can misreport ECN markings to alter the congestion
response of a sender. Suppressing reports of ECN-CE markings could
cause a sender to increase their send rate. This increase could
result in congestion and loss.
A sender MAY attempt to detect suppression of reports by marking
occasional packets that they send with ECN-CE. If a packet sent with
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ECN-CE is not reported as having been CE marked when the packet is
acknowledged, then the sender SHOULD disable ECN for that path.
Reporting additional ECN-CE markings will cause a sender to reduce
their sending rate, which is similar in effect to advertising reduced
connection flow control limits and so no advantage is gained by doing
so.
Endpoints choose the congestion controller that they use. Though
congestion controllers generally treat reports of ECN-CE markings as
equivalent to loss [RFC8311], the exact response for each controller
could be different. Failure to correctly respond to information
about ECN markings is therefore difficult to detect.
8. IANA Considerations
This document has no IANA actions. Yet.
9. References
9.1. Normative References
[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", Work in Progress, Internet-Draft, draft-ietf-quic-
tls-26, 21 February 2020,
<https://tools.ietf.org/html/draft-ietf-quic-tls-26>.
[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-26, 21 February
2020, <https://tools.ietf.org/html/draft-ietf-quic-
transport-26>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
9.2. Informative References
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[FACK] Mathis, M. and J. Mahdavi, "Forward Acknowledgement:
Refining TCP Congestion Control", ACM SIGCOMM , August
1996.
[RACK] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "RACK:
a time-based fast loss detection algorithm for TCP", Work
in Progress, Internet-Draft, draft-ietf-tcpm-rack-05, 26
April 2019, <http://www.ietf.org/internet-drafts/draft-
ietf-tcpm-rack-05.txt>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>.
[RFC4653] Bhandarkar, S., Reddy, A. L. N., Allman, M., and E.
Blanton, "Improving the Robustness of TCP to Non-
Congestion Events", RFC 4653, DOI 10.17487/RFC4653, August
2006, <https://www.rfc-editor.org/info/rfc4653>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>.
[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
"Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
Spurious Retransmission Timeouts with TCP", RFC 5682,
DOI 10.17487/RFC5682, September 2009,
<https://www.rfc-editor.org/info/rfc5682>.
[RFC5827] Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
P. Hurtig, "Early Retransmit for TCP and Stream Control
Transmission Protocol (SCTP)", RFC 5827,
DOI 10.17487/RFC5827, May 2010,
<https://www.rfc-editor.org/info/rfc5827>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<https://www.rfc-editor.org/info/rfc6298>.
[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012,
<https://www.rfc-editor.org/info/rfc6582>.
[RFC6675] Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
and Y. Nishida, "A Conservative Loss Recovery Algorithm
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Based on Selective Acknowledgment (SACK) for TCP",
RFC 6675, DOI 10.17487/RFC6675, August 2012,
<https://www.rfc-editor.org/info/rfc6675>.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
"Increasing TCP's Initial Window", RFC 6928,
DOI 10.17487/RFC6928, April 2013,
<https://www.rfc-editor.org/info/rfc6928>.
[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<https://www.rfc-editor.org/info/rfc7661>.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion
Notification (ECN) Experimentation", RFC 8311,
DOI 10.17487/RFC8311, January 2018,
<https://www.rfc-editor.org/info/rfc8311>.
[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
RFC 8312, DOI 10.17487/RFC8312, February 2018,
<https://www.rfc-editor.org/info/rfc8312>.
Appendix A. Loss Recovery Pseudocode
We now describe an example implementation of the loss detection
mechanisms described in Section 5.
A.1. Tracking Sent Packets
To correctly implement congestion control, a QUIC sender tracks every
ack-eliciting packet until the packet is acknowledged or lost. It is
expected that implementations will be able to access this information
by packet number and crypto context and store the per-packet fields
(Appendix A.1.1) for loss recovery and congestion control.
After a packet is declared lost, the endpoint can track it for an
amount of time comparable to the maximum expected packet reordering,
such as 1 RTT. This allows for detection of spurious
retransmissions.
Sent packets are tracked for each packet number space, and ACK
processing only applies to a single space.
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A.1.1. Sent Packet Fields
packet_number: The packet number of the sent packet.
ack_eliciting: A boolean that indicates whether a packet is ack-
eliciting. If true, it is expected that an acknowledgement will
be received, though the peer could delay sending the ACK frame
containing it by up to the MaxAckDelay.
in_flight: A boolean that indicates whether the packet counts
towards bytes in flight.
sent_bytes: The number of bytes sent in the packet, not including
UDP or IP overhead, but including QUIC framing overhead.
time_sent: The time the packet was sent.
A.2. Constants of interest
Constants used in loss recovery are based on a combination of RFCs,
papers, and common practice.
kPacketThreshold: Maximum reordering in packets before packet
threshold loss detection considers a packet lost. The RECOMMENDED
value is 3.
kTimeThreshold: Maximum reordering in time before time threshold
loss detection considers a packet lost. Specified as an RTT
multiplier. The RECOMMENDED value is 9/8.
kGranularity: Timer granularity. This is a system-dependent value.
However, implementations SHOULD use a value no smaller than 1ms.
kInitialRtt: The RTT used before an RTT sample is taken. The
RECOMMENDED value is 500ms.
kPacketNumberSpace: An enum to enumerate the three packet number
spaces.
enum kPacketNumberSpace {
Initial,
Handshake,
ApplicationData,
}
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A.3. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as
described in [RFC6298]
rttvar: The RTT variation, computed as described in [RFC6298]
min_rtt: The minimum RTT seen in the connection, ignoring ack delay.
max_ack_delay: The maximum amount of time by which the receiver
intends to delay acknowledgments for packets in the
ApplicationData packet number space. The actual ack_delay in a
received ACK frame may be larger due to late timers, reordering,
or lost ACK frames.
loss_detection_timer: Multi-modal timer used for loss detection.
pto_count: The number of times a PTO has been sent without receiving
an ack.
time_of_last_sent_ack_eliciting_packet[kPacketNumberSpace]: The time
the most recent ack-eliciting packet was sent.
largest_acked_packet[kPacketNumberSpace]: The largest packet number
acknowledged in the packet number space so far.
loss_time[kPacketNumberSpace]: The time at which the next packet in
that packet number space will be considered lost based on
exceeding the reordering window in time.
sent_packets[kPacketNumberSpace]: An association of packet numbers
in a packet number space to information about them. Described in
detail above in Appendix A.1.
A.4. Initialization
At the beginning of the connection, initialize the loss detection
variables as follows:
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loss_detection_timer.reset()
pto_count = 0
latest_rtt = 0
smoothed_rtt = 0
rttvar = 0
min_rtt = 0
max_ack_delay = 0
for pn_space in [ Initial, Handshake, ApplicationData ]:
largest_acked_packet[pn_space] = infinite
time_of_last_sent_ack_eliciting_packet[pn_space] = 0
loss_time[pn_space] = 0
A.5. On Sending a Packet
After a packet is sent, information about the packet is stored. The
parameters to OnPacketSent are described in detail above in
Appendix A.1.1.
Pseudocode for OnPacketSent follows:
OnPacketSent(packet_number, pn_space, ack_eliciting,
in_flight, sent_bytes):
sent_packets[pn_space][packet_number].packet_number =
packet_number
sent_packets[pn_space][packet_number].time_sent = now
sent_packets[pn_space][packet_number].ack_eliciting =
ack_eliciting
sent_packets[pn_space][packet_number].in_flight = in_flight
if (in_flight):
if (ack_eliciting):
time_of_last_sent_ack_eliciting_packet[pn_space] = now
OnPacketSentCC(sent_bytes)
sent_packets[pn_space][packet_number].size = sent_bytes
SetLossDetectionTimer()
A.6. On Receiving an Acknowledgment
When an ACK frame is received, it may newly acknowledge any number of
packets.
Pseudocode for OnAckReceived and UpdateRtt follow:
OnAckReceived(ack, pn_space):
if (largest_acked_packet[pn_space] == infinite):
largest_acked_packet[pn_space] = ack.largest_acked
else:
largest_acked_packet[pn_space] =
max(largest_acked_packet[pn_space], ack.largest_acked)
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// Nothing to do if there are no newly acked packets.
newly_acked_packets = DetermineNewlyAckedPackets(ack, pn_space)
if (newly_acked_packets.empty()):
return
// If the largest acknowledged is newly acked and
// at least one ack-eliciting was newly acked, update the RTT.
if (sent_packets[pn_space].contains(ack.largest_acked) &&
IncludesAckEliciting(newly_acked_packets)):
latest_rtt =
now - sent_packets[pn_space][ack.largest_acked].time_sent
ack_delay = 0
if (pn_space == ApplicationData):
ack_delay = ack.ack_delay
UpdateRtt(ack_delay)
// Process ECN information if present.
if (ACK frame contains ECN information):
ProcessECN(ack, pn_space)
for acked_packet in newly_acked_packets:
OnPacketAcked(acked_packet.packet_number, pn_space)
DetectLostPackets(pn_space)
pto_count = 0
SetLossDetectionTimer()
UpdateRtt(ack_delay):
// First RTT sample.
if (smoothed_rtt == 0):
min_rtt = latest_rtt
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
return
// min_rtt ignores ack delay.
min_rtt = min(min_rtt, latest_rtt)
// Limit ack_delay by max_ack_delay
ack_delay = min(ack_delay, max_ack_delay)
// Adjust for ack delay if plausible.
adjusted_rtt = latest_rtt
if (latest_rtt > min_rtt + ack_delay):
adjusted_rtt = latest_rtt - ack_delay
rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt)
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smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
A.7. On Packet Acknowledgment
When a packet is acknowledged for the first time, the following
OnPacketAcked function is called. Note that a single ACK frame may
newly acknowledge several packets. OnPacketAcked must be called once
for each of these newly acknowledged packets.
OnPacketAcked takes two parameters: acked_packet, which is the struct
detailed in Appendix A.1.1, and the packet number space that this ACK
frame was sent for.
Pseudocode for OnPacketAcked follows:
OnPacketAcked(acked_packet, pn_space):
if (acked_packet.in_flight):
OnPacketAckedCC(acked_packet)
sent_packets[pn_space].remove(acked_packet.packet_number)
A.8. Setting the Loss Detection Timer
QUIC loss detection uses a single timer for all timeout loss
detection. The duration of the timer is based on the timer's mode,
which is set in the packet and timer events further below. The
function SetLossDetectionTimer defined below shows how the single
timer is set.
This algorithm may result in the timer being set in the past,
particularly if timers wake up late. Timers set in the past SHOULD
fire immediately.
Pseudocode for SetLossDetectionTimer follows:
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GetEarliestTimeAndSpace(times):
time = times[Initial]
space = Initial
for pn_space in [ Handshake, ApplicationData ]:
if (times[pn_space] != 0 &&
(time == 0 || times[pn_space] < time) &&
# Skip ApplicationData until handshake completion.
(pn_space != ApplicationData ||
IsHandshakeComplete()):
time = times[pn_space];
space = pn_space
return time, space
PeerNotAwaitingAddressValidation():
# Assume clients validate the server's address implicitly.
if (endpoint is server):
return true
# Servers complete address validation when a
# protected packet is received.
return has received Handshake ACK ||
has received 1-RTT ACK
SetLossDetectionTimer():
earliest_loss_time, _ = GetEarliestTimeAndSpace(loss_time)
if (earliest_loss_time != 0):
// Time threshold loss detection.
loss_detection_timer.update(earliest_loss_time)
return
if (no ack-eliciting packets in flight &&
PeerNotAwaitingAddressValidation()):
loss_detection_timer.cancel()
return
// Use a default timeout if there are no RTT measurements
if (smoothed_rtt == 0):
timeout = 2 * kInitialRtt
else:
// Calculate PTO duration
timeout = smoothed_rtt + max(4 * rttvar, kGranularity) +
max_ack_delay
timeout = timeout * (2 ^ pto_count)
sent_time, _ = GetEarliestTimeAndSpace(
time_of_last_sent_ack_eliciting_packet)
loss_detection_timer.update(sent_time + timeout)
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A.9. On Timeout
When the loss detection timer expires, the timer's mode determines
the action to be performed.
Pseudocode for OnLossDetectionTimeout follows:
OnLossDetectionTimeout():
earliest_loss_time, pn_space =
GetEarliestTimeAndSpace(loss_time)
if (earliest_loss_time != 0):
// Time threshold loss Detection
DetectLostPackets(pn_space)
SetLossDetectionTimer()
return
if (endpoint is client without 1-RTT keys):
// Client sends an anti-deadlock packet: Initial is padded
// to earn more anti-amplification credit,
// a Handshake packet proves address ownership.
if (has Handshake keys):
SendOneAckElicitingHandshakePacket()
else:
SendOneAckElicitingPaddedInitialPacket()
else:
// PTO. Send new data if available, else retransmit old data.
// If neither is available, send a single PING frame.
_, pn_space = GetEarliestTimeAndSpace(
time_of_last_sent_ack_eliciting_packet)
SendOneOrTwoAckElicitingPackets(pn_space)
pto_count++
SetLossDetectionTimer()
A.10. Detecting Lost Packets
DetectLostPackets is called every time an ACK is received and
operates on the sent_packets for that packet number space.
Pseudocode for DetectLostPackets follows:
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DetectLostPackets(pn_space):
assert(largest_acked_packet[pn_space] != infinite)
loss_time[pn_space] = 0
lost_packets = {}
loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)
// Minimum time of kGranularity before packets are deemed lost.
loss_delay = max(loss_delay, kGranularity)
// Packets sent before this time are deemed lost.
lost_send_time = now() - loss_delay
foreach unacked in sent_packets[pn_space]:
if (unacked.packet_number > largest_acked_packet[pn_space]):
continue
// Mark packet as lost, or set time when it should be marked.
if (unacked.time_sent <= lost_send_time ||
largest_acked_packet[pn_space] >=
unacked.packet_number + kPacketThreshold):
sent_packets[pn_space].remove(unacked.packet_number)
if (unacked.in_flight):
lost_packets.insert(unacked)
else:
if (loss_time[pn_space] == 0):
loss_time[pn_space] = unacked.time_sent + loss_delay
else:
loss_time[pn_space] = min(loss_time[pn_space],
unacked.time_sent + loss_delay)
// Inform the congestion controller of lost packets and
// let it decide whether to retransmit immediately.
if (!lost_packets.empty()):
OnPacketsLost(lost_packets)
Appendix B. Congestion Control Pseudocode
We now describe an example implementation of the congestion
controller described in Section 6.
B.1. Constants of interest
Constants used in congestion control are based on a combination of
RFCs, papers, and common practice.
kInitialWindow: Default limit on the initial amount of data in
flight, in bytes. The RECOMMENDED value is the minimum of 10 *
max_datagram_size and max(2 * max_datagram_size, 14720)). This
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follows the analysis and recommendations in [RFC6928], increasing
the byte limit to account for the smaller 8 byte overhead of UDP
compared to the 20 byte overhead for TCP.
kMinimumWindow: Minimum congestion window in bytes. The RECOMMENDED
value is 2 * max_datagram_size.
kLossReductionFactor: Reduction in congestion window when a new loss
event is detected. The RECOMMENDED value is 0.5.
kPersistentCongestionThreshold: Period of time for persistent
congestion to be established, specified as a PTO multiplier. The
rationale for this threshold is to enable a sender to use initial
PTOs for aggressive probing, as TCP does with Tail Loss Probe
(TLP) [RACK], before establishing persistent congestion, as TCP
does with a Retransmission Timeout (RTO) [RFC5681]. The
RECOMMENDED value for kPersistentCongestionThreshold is 3, which
is approximately equivalent to having two TLPs before an RTO in
TCP.
B.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
max_datagram_size: The sender's current maximum payload size. Does
not include UDP or IP overhead. The max datagram size is used for
congestion window computations. An endpoint sets the value of
this variable based on its PMTU (see Section 14.1 of
[QUIC-TRANSPORT]), with a minimum value of 1200 bytes.
ecn_ce_counters[kPacketNumberSpace]: The highest value reported for
the ECN-CE counter in the packet number space by the peer in an
ACK frame. This value is used to detect increases in the reported
ECN-CE counter.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one ack-eliciting or PADDING frame, and have
not been acked or declared lost. The size does not include IP or
UDP overhead, but does include the QUIC header and AEAD overhead.
Packets only containing ACK frames do not count towards
bytes_in_flight to ensure congestion control does not impede
congestion feedback.
congestion_window: Maximum number of bytes-in-flight that may be
sent.
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congestion_recovery_start_time: The time when QUIC first detects
congestion due to loss or ECN, causing it to enter congestion
recovery. When a packet sent after this time is acknowledged,
QUIC exits congestion recovery.
ssthresh: Slow start threshold in bytes. When the congestion window
is below ssthresh, the mode is slow start and the window grows by
the number of bytes acknowledged.
B.3. Initialization
At the beginning of the connection, initialize the congestion control
variables as follows:
congestion_window = kInitialWindow
bytes_in_flight = 0
congestion_recovery_start_time = 0
ssthresh = infinite
for pn_space in [ Initial, Handshake, ApplicationData ]:
ecn_ce_counters[pn_space] = 0
B.4. On Packet Sent
Whenever a packet is sent, and it contains non-ACK frames, the packet
increases bytes_in_flight.
OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent
B.5. On Packet Acknowledgement
Invoked from loss detection's OnPacketAcked and is supplied with the
acked_packet from sent_packets.
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InCongestionRecovery(sent_time):
return sent_time <= congestion_recovery_start_time
OnPacketAckedCC(acked_packet):
// Remove from bytes_in_flight.
bytes_in_flight -= acked_packet.size
if (InCongestionRecovery(acked_packet.time_sent)):
// Do not increase congestion window in recovery period.
return
if (IsAppOrFlowControlLimited()):
// Do not increase congestion_window if application
// limited or flow control limited.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packet.size
else:
// Congestion avoidance.
congestion_window += max_datagram_size * acked_packet.size
/ congestion_window
B.6. On New Congestion Event
Invoked from ProcessECN and OnPacketsLost when a new congestion event
is detected. May start a new recovery period and reduces the
congestion window.
CongestionEvent(sent_time):
// Start a new congestion event if packet was sent after the
// start of the previous congestion recovery period.
if (!InCongestionRecovery(sent_time)):
congestion_recovery_start_time = Now()
congestion_window *= kLossReductionFactor
congestion_window = max(congestion_window, kMinimumWindow)
ssthresh = congestion_window
B.7. Process ECN Information
Invoked when an ACK frame with an ECN section is received from the
peer.
ProcessECN(ack, pn_space):
// If the ECN-CE counter reported by the peer has increased,
// this could be a new congestion event.
if (ack.ce_counter > ecn_ce_counters[pn_space]):
ecn_ce_counters[pn_space] = ack.ce_counter
CongestionEvent(sent_packets[ack.largest_acked].time_sent)
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B.8. On Packets Lost
Invoked from DetectLostPackets when packets are deemed lost.
InPersistentCongestion(largest_lost_packet):
pto = smoothed_rtt + max(4 * rttvar, kGranularity) +
max_ack_delay
congestion_period = pto * kPersistentCongestionThreshold
// Determine if all packets in the time period before the
// newest lost packet, including the edges, are marked
// lost
return AreAllPacketsLost(largest_lost_packet,
congestion_period)
OnPacketsLost(lost_packets):
// Remove lost packets from bytes_in_flight.
for (lost_packet : lost_packets):
bytes_in_flight -= lost_packet.size
largest_lost_packet = lost_packets.last()
CongestionEvent(largest_lost_packet.time_sent)
// Collapse congestion window if persistent congestion
if (InPersistentCongestion(largest_lost_packet)):
congestion_window = kMinimumWindow
Appendix C. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
Issue and pull request numbers are listed with a leading octothorp.
C.1. Since draft-ietf-quic-recovery-25
No significant changes.
C.2. 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.3. Since draft-ietf-quic-recovery-23
* Define under-utilizing the congestion window (#2630, #2686, #2675)
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* 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.4. 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.5. Since draft-ietf-quic-recovery-21
* No changes
C.6. Since draft-ietf-quic-recovery-20
* Path validation can be used as initial RTT value (#2644, #2687)
* max_ack_delay transport parameter defaults to 0 (#2638, #2646)
* Ack Delay only measures intentional delays induced by the
implementation (#2596, #2786)
C.7. 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)
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* 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.8. Since draft-ietf-quic-recovery-18
* Change IW byte limit to 14720 from 14600 (#2494)
* Update PTO calculation to match RFC6298 (#2480, #2489, #2490)
* Improve loss detection's description of multiple packet number
spaces and pseudocode (#2485, #2451, #2417)
* Declare persistent congestion even if non-probe packets are sent
and don't make persistent congestion more aggressive than RTO
verified was (#2365, #2244)
* Move pseudocode to the appendices (#2408)
* What to send on multiple PTOs (#2380)
C.9. 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)
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* Congestion and loss recovery state are discarded when keys are
discarded (#2327)
C.10. Since draft-ietf-quic-recovery-16
* Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP
and min crypto timeouts; eliminate timeout validation (#2114,
#2166, #2168, #1017)
* Redefine how congestion avoidance in terms of when the period
starts (#1928, #1930)
* Document what needs to be tracked for packets that are in flight
(#765, #1724, #1939)
* Integrate both time and packet thresholds into loss detection
(#1969, #1212, #934, #1974)
* Reduce congestion window after idle, unless pacing is used (#2007,
#2023)
* Disable RTT calculation for packets that don't elicit
acknowledgment (#2060, #2078)
* Limit ack_delay by max_ack_delay (#2060, #2099)
* Initial keys are discarded once Handshake keys are available
(#1951, #2045)
* Reorder ECN and loss detection in pseudocode (#2142)
* Only cancel loss detection timer if ack-eliciting packets are in
flight (#2093, #2117)
C.11. Since draft-ietf-quic-recovery-14
* Used max_ack_delay from transport params (#1796, #1782)
* Merge ACK and ACK_ECN (#1783)
C.12. Since draft-ietf-quic-recovery-13
* Corrected the lack of ssthresh reduction in CongestionEvent
pseudocode (#1598)
* Considerations for ECN spoofing (#1426, #1626)
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* 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.13. 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.14. Since draft-ietf-quic-recovery-11
No significant changes.
C.15. Since draft-ietf-quic-recovery-10
* Improved text on ack generation (#1139, #1159)
* Make references to TCP recovery mechanisms informational (#1195)
* Define time_of_last_sent_handshake_packet (#1171)
* Added signal from TLS the data it includes needs to be sent in a
Retry packet (#1061, #1199)
* Minimum RTT (min_rtt) is initialized with an infinite value
(#1169)
C.16. Since draft-ietf-quic-recovery-09
No significant changes.
C.17. Since draft-ietf-quic-recovery-08
* Clarified pacing and RTO (#967, #977)
C.18. 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)
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* Many editorial fixes.
C.19. Since draft-ietf-quic-recovery-06
No significant changes.
C.20. Since draft-ietf-quic-recovery-05
* Add more congestion control text (#776)
C.21. Since draft-ietf-quic-recovery-04
No significant changes.
C.22. Since draft-ietf-quic-recovery-03
No significant changes.
C.23. Since draft-ietf-quic-recovery-02
* Integrate F-RTO (#544, #409)
* Add congestion control (#545, #395)
* Require connection abort if a skipped packet was acknowledged
(#415)
* Simplify RTO calculations (#142, #417)
C.24. 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.25. Since draft-ietf-quic-recovery-00
* Improved description of constants and ACK behavior
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C.26. Since draft-iyengar-quic-loss-recovery-01
* Adopted as base for draft-ietf-quic-recovery
* Updated authors/editors list
* Added table of contents
Appendix D. Contributors
The IETF QUIC Working Group received an enormous amount of support
from many people. The following people provided substantive
contributions to this document: Alessandro Ghedini, Benjamin
Saunders, Gorry Fairhurst, 奥 一穂 (Kazuho Oku), Lars Eggert, Magnus
Westerlund, Marten Seemann, Martin Duke, Martin Thomson, Nick Banks,
Praveen Balasubramaniam.
Acknowledgments
Authors' Addresses
Jana Iyengar (editor)
Fastly
Email: jri.ietf@gmail.com
Ian Swett (editor)
Google
Email: ianswett@google.com
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