QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track I. Swett, Ed.
Expires: October 25, 2019 Google
April 23, 2019
QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-20
Abstract
This document describes loss detection and congestion control
mechanisms for QUIC.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic [1].
Working Group information can be found at https://github.com/quicwg
[2]; source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/-recovery [3].
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 25, 2019.
Copyright Notice
Copyright (c) 2019 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 4
3. Design of the QUIC Transmission Machinery . . . . . . . . . . 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. No Reneging . . . . . . . . . . . . . . . . . . . . . 6
3.1.4. More ACK Ranges . . . . . . . . . . . . . . . . . . . 7
3.1.5. Explicit Correction For Delayed Acknowledgements . . 7
4. Generating Acknowledgements . . . . . . . . . . . . . . . . . 7
4.1. Crypto Handshake Data . . . . . . . . . . . . . . . . . . 7
4.2. ACK Ranges . . . . . . . . . . . . . . . . . . . . . . . 8
4.3. Receiver Tracking of ACK Frames . . . . . . . . . . . . . 8
4.4. Measuring and Reporting Host Delay . . . . . . . . . . . 8
5. Estimating the Round-Trip Time . . . . . . . . . . . . . . . 9
5.1. Generating RTT samples . . . . . . . . . . . . . . . . . 9
5.2. Estimating min_rtt . . . . . . . . . . . . . . . . . . . 10
5.3. Estimating smoothed_rtt and rttvar . . . . . . . . . . . 10
6. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 11
6.1. Acknowledgement-based Detection . . . . . . . . . . . . . 11
6.1.1. Packet Threshold . . . . . . . . . . . . . . . . . . 12
6.1.2. Time Threshold . . . . . . . . . . . . . . . . . . . 12
6.2. Crypto Retransmission Timeout . . . . . . . . . . . . . . 13
6.2.1. Retry and Version Negotiation . . . . . . . . . . . . 14
6.2.2. Discarding Keys and Packet State . . . . . . . . . . 14
6.3. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 15
6.3.1. Computing PTO . . . . . . . . . . . . . . . . . . . . 15
6.3.2. Sending Probe Packets . . . . . . . . . . . . . . . . 16
6.3.3. Loss Detection . . . . . . . . . . . . . . . . . . . 17
6.4. Discussion . . . . . . . . . . . . . . . . . . . . . . . 17
7. Congestion Control . . . . . . . . . . . . . . . . . . . . . 17
7.1. Explicit Congestion Notification . . . . . . . . . . . . 17
7.2. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 18
7.3. Congestion Avoidance . . . . . . . . . . . . . . . . . . 18
7.4. Recovery Period . . . . . . . . . . . . . . . . . . . . . 18
7.5. Ignoring Loss of Undecryptable Packets . . . . . . . . . 18
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7.6. Probe Timeout . . . . . . . . . . . . . . . . . . . . . . 18
7.7. Persistent Congestion . . . . . . . . . . . . . . . . . . 19
7.8. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 20
7.9. Under-utilizing the Congestion Window . . . . . . . . . . 20
8. Security Considerations . . . . . . . . . . . . . . . . . . . 21
8.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 21
8.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 21
8.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 21
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 22
10.1. Normative References . . . . . . . . . . . . . . . . . . 22
10.2. Informative References . . . . . . . . . . . . . . . . . 22
10.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 24
Appendix A. Loss Recovery Pseudocode . . . . . . . . . . . . . . 24
A.1. Tracking Sent Packets . . . . . . . . . . . . . . . . . . 24
A.1.1. Sent Packet Fields . . . . . . . . . . . . . . . . . 24
A.2. Constants of interest . . . . . . . . . . . . . . . . . . 25
A.3. Variables of interest . . . . . . . . . . . . . . . . . . 25
A.4. Initialization . . . . . . . . . . . . . . . . . . . . . 26
A.5. On Sending a Packet . . . . . . . . . . . . . . . . . . . 27
A.6. On Receiving an Acknowledgment . . . . . . . . . . . . . 27
A.7. On Packet Acknowledgment . . . . . . . . . . . . . . . . 29
A.8. Setting the Loss Detection Timer . . . . . . . . . . . . 29
A.9. On Timeout . . . . . . . . . . . . . . . . . . . . . . . 31
A.10. Detecting Lost Packets . . . . . . . . . . . . . . . . . 31
Appendix B. Congestion Control Pseudocode . . . . . . . . . . . 32
B.1. Constants of interest . . . . . . . . . . . . . . . . . . 32
B.2. Variables of interest . . . . . . . . . . . . . . . . . . 33
B.3. Initialization . . . . . . . . . . . . . . . . . . . . . 34
B.4. On Packet Sent . . . . . . . . . . . . . . . . . . . . . 34
B.5. On Packet Acknowledgement . . . . . . . . . . . . . . . . 34
B.6. On New Congestion Event . . . . . . . . . . . . . . . . . 35
B.7. Process ECN Information . . . . . . . . . . . . . . . . . 35
B.8. On Packets Lost . . . . . . . . . . . . . . . . . . . . . 36
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 36
C.1. Since draft-ietf-quic-recovery-19 . . . . . . . . . . . . 36
C.2. Since draft-ietf-quic-recovery-18 . . . . . . . . . . . . 37
C.3. Since draft-ietf-quic-recovery-17 . . . . . . . . . . . . 37
C.4. Since draft-ietf-quic-recovery-16 . . . . . . . . . . . . 38
C.5. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 38
C.6. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 38
C.7. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 39
C.8. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 39
C.9. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 39
C.10. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 39
C.11. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 39
C.12. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 39
C.13. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 40
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C.14. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 40
C.15. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 40
C.16. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 40
C.17. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 40
C.18. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 40
C.19. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 40
C.20. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 40
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 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-only: Any packet containing only one or more ACK frame(s).
In-flight: Packets are considered in-flight when they have been sent
and neither acknowledged nor declared lost, and they are not ACK-
only.
Ack-eliciting Frames: All frames besides ACK or PADDING 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.
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Crypto Packets: Packets containing CRYPTO data sent in Initial or
Handshake packets.
Out-of-order Packets: Packets that do not increase the largest
received packet number for its packet number space by exactly one.
Packets arrive out of order when earlier packets are lost or
delayed.
3. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
indicates the encryption level and includes a packet sequence number
(referred to below as a packet number). The encryption level
indicates the packet number space, as described in [QUIC-TRANSPORT].
Packet numbers never repeat within a packet number space for the
lifetime of a connection. Packet numbers monotonically increase
within a space, preventing ambiguity.
This design obviates the need for disambiguating between
transmissions and retransmissions and eliminates significant
complexity from QUIC's interpretation of TCP loss detection
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:
o All packets are acknowledged, though packets that contain no ack-
eliciting frames are only acknowledged along with ack-eliciting
packets.
o Long header packets that contain CRYPTO frames are critical to the
performance of the QUIC handshake and use shorter timers for
acknowledgement and retransmission.
o Packets that contain only ACK frames do not count toward
congestion control limits and are not considered in-flight.
o 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
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algorithmic differences. We briefly describe these protocol
differences below.
3.1.1. Separate Packet Number Spaces
QUIC uses separate packet number spaces for each encryption level,
except 0-RTT and all generations of 1-RTT keys use the same packet
number space. Separate packet number spaces ensures acknowledgement
of packets sent with one level of encryption will not cause spurious
retransmission of packets sent with a different encryption level.
Congestion control and 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, and any application
data is sent in one or more streams, with delivery order determined
by stream offsets encoded within STREAM frames.
QUIC's packet number is strictly increasing 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. 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.
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3.1.4. More ACK Ranges
QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In
high loss environments, this speeds recovery, reduces spurious
retransmits, and ensures forward progress without relying on
timeouts.
3.1.5. 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 4.4).
4. Generating Acknowledgements
An acknowledgement SHOULD be sent immediately upon receipt of a
second ack-eliciting packet. QUIC recovery algorithms do not assume
the peer sends an ACK immediately when receiving a second ack-
eliciting packet.
In order to accelerate loss recovery and reduce timeouts, the
receiver SHOULD send an immediate ACK after it receives an out-of-
order packet. It could send immediate ACKs for in-order packets for
a period of time that SHOULD NOT exceed 1/8 RTT unless more out-of-
order packets arrive. If every packet arrives out-of- order, then an
immediate ACK SHOULD be sent for every received packet.
Similarly, packets marked with the ECN Congestion Experienced (CE)
codepoint in the IP header SHOULD be acknowledged immediately, to
reduce the peer's response time to congestion events.
As an optimization, a receiver MAY process multiple packets before
sending any ACK frames in response. In this case the receiver can
determine whether an immediate or delayed acknowledgement should be
generated after processing incoming packets.
4.1. Crypto Handshake Data
In order to quickly complete the handshake and avoid spurious
retransmissions due to crypto retransmission timeouts, crypto packets
SHOULD use a very short ack delay, such as the local timer
granularity. ACK frames SHOULD be sent immediately when the crypto
stack indicates all data for that packet number space has been
received.
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4.2. ACK Ranges
When an ACK frame is sent, one or more ranges of acknowledged packets
are included. Including older packets reduces the chance of spurious
retransmits caused by losing previously sent ACK frames, at the cost
of larger ACK frames.
ACK frames SHOULD always acknowledge the most recently received
packets, and the more out-of-order the packets are, the more
important it is to send an updated ACK frame quickly, to prevent the
peer from declaring a packet as lost and spuriously retransmitting
the frames it contains.
Below is one recommended approach for determining what packets to
include in an ACK frame.
4.3. Receiver Tracking of ACK Frames
When a packet containing an ACK frame is sent, the largest
acknowledged in that frame may be saved. When a packet containing an
ACK frame is acknowledged, the receiver can stop acknowledging
packets less than or equal to the largest acknowledged in the sent
ACK frame.
In cases without ACK frame loss, this algorithm allows for a minimum
of 1 RTT of reordering. In cases with ACK frame loss and reordering,
this approach does not guarantee that every acknowledgement is seen
by the sender before it is no longer included in the ACK frame.
Packets could be received out of order and all subsequent ACK frames
containing them could be lost. In this case, the loss recovery
algorithm may cause spurious retransmits, but the sender will
continue making forward progress.
4.4. Measuring and Reporting Host Delay
An endpoint measures the delay incurred between when a packet is
received and when the corresponding acknowledgment is sent. The
endpoint encodes this host delay for the largest acknowledged packet
in the Ack Delay field of an ACK frame (see Section 19.3 of
[QUIC-TRANSPORT]). This allows the receiver of the ACK to adjust for
any host delays, which is important for delayed acknowledgements,
when estimating the path RTT. In certain deployments, a packet might
be held in the OS kernel or elsewhere on the host before being
processed by the QUIC stack. Where possible, an endpoint MAY include
these delays when populating the Ack Delay field in an ACK frame.
An endpoint MUST NOT excessively delay acknowledgements of ack-
eliciting packets. The maximum ack delay is communicated in the
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max_ack_delay transport parameter, see Section 18.1 of
[QUIC-TRANSPORT]. max_ack_delay implies an explicit contract: an
endpoint promises to never delay acknowledgments of an ack-eliciting
packet by more than the indicated value. If it does, any excess
accrues to the RTT estimate and could result in spurious
retransmissions from the peer.
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
(Section 4.4) to generate a statistical description of the
connection's RTT. An endpoint computes the following three values:
the minimum value observed over the lifetime of the connection
(min_rtt), an exponentially-weighted moving average (smoothed_rtt),
and the variance 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:
o the largest acknowledged packet number is newly acknowledged, and
o 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 host
delays for only the largest acknowledged packet in an ACK frame.
While the reported host 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.
To avoid generating multiple RTT samples using the same 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
does not send an ACK frame on receiving only non-ack-eliciting
packets, so an ACK frame that is subsequently sent can include an
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arbitrarily large Ack Delay field. 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 over the lifetime of the
connection. min_rtt is set to the latest_rtt on the first sample in
a connection, and to the lesser of min_rtt and latest_rtt on
subsequent samples.
An endpoint uses only locally observed times in computing the min_rtt
and does not adjust for host delays reported by the peer
(Section 4.4). 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.
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 endpoint's estimated
variance in the RTT samples.
smoothed_rtt uses path latency after adjusting RTT samples for peer-
reported host delays (Section 4.4). A peer limits any delay in
sending an acknowledgement for an ack-eliciting packet to no greater
than the advertised 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:
o 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
(Section 4.4).
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o 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 in a connection, 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 in a connection:
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
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
6. Loss Detection
QUIC senders use both ack information and timeouts to detect lost
packets, and 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 13.2 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:
o The packet is unacknowledged, in-flight, and was sent prior to an
acknowledged packet.
o 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).
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The acknowledgement indicates that a packet sent later was delivered,
while 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.
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].
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.
6.1.2. Time Threshold
Once a later packet has been acknowledged, an endpoint SHOULD declare
an earlier packet lost if it was sent a threshold amount of time in
the past. The time threshold is computed as kTimeThreshold *
max(SRTT, latest_RTT). If packets sent prior to the largest
acknowledged packet cannot yet be declared lost, then a timer SHOULD
be set for the remaining time.
The RECOMMENDED time threshold (kTimeThreshold), expressed as a
round-trip time multiplier, is 9/8.
Using max(SRTT, latest_RTT) protects from the two following cases:
o the latest RTT sample is lower than the SRTT, perhaps due to
reordering where the acknowledgement encountered a shorter path;
o the latest RTT sample is higher than the SRTT, perhaps due to a
sustained increase in the actual RTT, but the smoothed SRTT has
not yet caught up.
An endpoint might consistently record RTT samples as 0 in extremely
low latency networks, leading to a smoothed_rtt of 0. Consequently,
the endpoint could declare all earlier packets as lost immediately
upon receiving an acknowledgement for a later packet. That is, the
endpoint would not provide any reordering tolerance. To avoid
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declaring packets as lost too early, the time threshold MUST be set
to at least kGranularity (defined in Appendix A.2).
Implementations MAY experiment with absolute thresholds, thresholds
from previous connections, adaptive thresholds, or including RTT
variance. Smaller thresholds reduce reordering resilience and
increase spurious retransmissions, and larger thresholds increase
loss detection delay.
6.2. Crypto Retransmission Timeout
Data in CRYPTO frames is critical to QUIC transport and crypto
negotiation, so a more aggressive timeout is used to retransmit it.
The initial crypto retransmission timeout SHOULD be set to twice the
initial RTT.
At the beginning, there are no prior RTT samples within a connection.
Resumed connections over the same network SHOULD use the previous
connection's final smoothed RTT value as the resumed connection's
initial RTT. If no previous RTT is available, or if the network
changes, the initial RTT SHOULD be set to 500ms, resulting in a 1
second initial handshake timeout as recommended in [RFC6298].
When a crypto packet is sent, the sender MUST set a timer for twice
the smoothed RTT. This timer MUST be updated when a new crypto
packet is sent and when an acknowledgement is received which computes
a new RTT sample. Upon timeout, the sender MUST retransmit all
unacknowledged CRYPTO data if possible. The sender MUST NOT declare
in-flight crypto packets as lost when the crypto timer expires.
On each consecutive expiration of the crypto timer without receiving
an acknowledgement for a new packet, the sender MUST double the
crypto retransmission timeout and set a timer for this period.
Until the server has validated the client's address on the path, the
amount of data it can send is limited, as specified in Section 8.1 of
[QUIC-TRANSPORT]. If not all unacknowledged CRYPTO data can be sent,
then all unacknowledged CRYPTO data sent in Initial packets should be
retransmitted. If no data can be sent, then no alarm should be armed
until data has been received from the client.
Because the server could be blocked until more packets are received,
the client MUST ensure that the crypto retransmission timer is set if
there is unacknowledged crypto data or if the client does not yet
have 1-RTT keys. If the crypto retransmission timer expires before
the client has 1-RTT keys, it is possible that the client may not
have any crypto data to retransmit. However, the client MUST send a
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new packet, containing only PING or PADDING frames if necessary, to
allow the server to continue sending data. 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.
The crypto retransmission timer is not 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 crypto
retransmission timeout and be less likely to spuriously retransmit
data. The Initial and Handshake packet number spaces will typically
contain a small number of packets, so losses are less likely to be
detected using packet-threshold loss detection.
When the crypto retransmission timer is active, the probe timer
(Section 6.3) is not active.
6.2.1. Retry and Version Negotiation
A Retry or Version Negotiation packet causes a client to send another
Initial packet, effectively restarting the connection process and
resetting congestion control and loss recovery state, including
resetting any pending timers. Either packet indicates that the
Initial was received but not processed. Neither packet can be
treated as an acknowledgment for the Initial.
The client MAY however 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 to seed the RTT estimator for a subsequent connection attempt
to the server.
6.2.2. Discarding Keys and Packet State
When packet protection keys are discarded (see Section 4.9 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.
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.
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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 of [QUIC-TLS]).
6.3. Probe Timeout
A Probe Timeout (PTO) triggers a probe packet when ack-eliciting data
is in flight but an acknowledgement is not received within the
expected period of time. A PTO enables a connection to recover from
loss of tail packets or acks. The PTO algorithm used in QUIC
implements the reliability functions of Tail Loss Probe [TLP] [RACK],
RTO [RFC5681] and F-RTO algorithms for TCP [RFC5682], and the timeout
computation is based on TCP's retransmission timeout period
[RFC6298].
6.3.1. 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 variance in the
estimate (4*rttvar), and max_ack_delay, to account for the maximum
time by which a receiver might delay sending an acknowledgement.
The PTO value MUST be set to at least kGranularity, to avoid the
timer expiring immediately.
When a PTO timer expires, the sender probes the network as described
in the next section. The PTO period MUST be set to twice its current
value. This exponential reduction in the sender's rate is important
because the PTOs might be caused by loss of packets or
acknowledgements due to severe congestion.
A sender computes its PTO timer every time an ack-eliciting packet is
sent. A sender might choose to optimize this by setting the timer
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fewer times if it knows that more ack-eliciting packets will be sent
within a short period of time.
6.3.2. Sending Probe Packets
When a PTO timer expires, a sender MUST send at least one ack-
eliciting packet as a probe, unless there is no data available to
send. An endpoint MAY send up to two ack-eliciting packets, to avoid
an expensive consecutive PTO expiration due to a single packet loss.
It is possible that 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. In the absence of any new application data, a PTO
timer expiration now would find the sender with no new or previously-
sent data to send.
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
opportunistically reduce loss recovery delay. Implementations MAY
use alternate 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.
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6.3.3. 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 6.1.
6.4. Discussion
The majority of constants were derived from best common practices
among widely deployed TCP implementations on the internet.
Exceptions follow.
A shorter delayed ack time of 25ms was chosen because longer delayed
acks can delay loss recovery and for the small number of connections
where less than packet per 25ms is delivered, acking every packet is
beneficial to congestion control and loss recovery.
7. Congestion Control
QUIC's congestion control is based on TCP NewReno [RFC6582]. NewReno
is a congestion window based congestion control. QUIC specifies the
congestion window in bytes rather than packets due to finer control
and the ease of appropriate byte counting [RFC3465].
QUIC hosts MUST NOT send packets if they would increase
bytes_in_flight (defined in Appendix B.2) beyond the available
congestion window, unless the packet is a probe packet sent after a
PTO timer expires, as described in Section 6.3.
Implementations MAY use other congestion control algorithms, such as
Cubic [RFC8312], and endpoints MAY use different algorithms from one
another. The signals QUIC provides for congestion control are
generic and are designed to support different algorithms.
7.1. Explicit Congestion Notification
If a path has been verified to support ECN, QUIC treats a Congestion
Experienced codepoint in the IP header as a signal of congestion.
This document specifies an endpoint's response when its peer receives
packets with the Congestion Experienced codepoint. As discussed in
[RFC8311], endpoints are permitted to experiment with other response
functions.
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7.2. Slow Start
QUIC begins every connection in slow start and exits slow start upon
loss or upon increase in the ECN-CE counter. QUIC re-enters slow
start anytime the congestion window is less than ssthresh, which
typically only occurs after an PTO. While in slow start, QUIC
increases the congestion window by the number of bytes acknowledged
when each acknowledgment is processed.
7.3. 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.
7.4. Recovery Period
Recovery is a period of time beginning with detection of a lost
packet or an increase in the ECN-CE counter. Because QUIC does not
retransmit packets, it defines the end of recovery as a packet sent
after the start of recovery being acknowledged. This is slightly
different from TCP's definition of recovery, which ends when the lost
packet that started recovery is 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.
7.5. 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.
7.6. 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
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acknowledgement is received that establishes loss or delivery of
packets.
7.7. 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 eck-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 |
+-----+------------------------+
The first three packets are determined to be lost when the ACK 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
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on persistent congestion is functionally similar to a sender's
response on a Retransmission Timeout (RTO) in TCP [RFC5681] after
Tail Loss Probes (TLP) [TLP].
7.8. 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 SRTT when used with a window-based
controller, and a pacer might use the rate estimate of a rate-based
controller.
An implementation should take care to architect its congestion
controller to work well with a pacer. For instance, a pacer might
wrap the congestion controller and control the availability of the
congestion window, or a pacer might pace out packets handed to it by
the congestion controller. Timely delivery of ACK frames is
important for efficient loss recovery. Packets containing only ACK
frames should therefore not be paced, to avoid delaying their
delivery to the peer.
As an example of a well-known and publicly available implementation
of a flow pacer, implementers are referred to the Fair Queue packet
scheduler (fq qdisc) in Linux (3.11 onwards).
7.9. Under-utilizing the Congestion Window
A congestion window that is under-utilized 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 7.8) 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.
Bursting more than an intial window's worth of data into the network
might cause short-term congestion and losses. Implemementations
SHOULD either use pacing or reduce their congestion window to limit
such bursts.
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A sender MAY implement alternate 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 marked
with ECN-CE is not reported as having been marked when the packet is
acknowledged, the sender SHOULD then disable ECN for that path.
Reporting additional ECN-CE markings will cause a sender to reduce
their sending rate, which is similar in effect to advertising reduced
connection flow control limits and so no advantage is gained by doing
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. Yet.
10. References
10.1. Normative References
[QUIC-TLS]
Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", draft-ietf-quic-tls-20 (work in progress), April
2019.
[QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", draft-ietf-quic-
transport-20 (work in progress), April 2019.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[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>.
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",
draft-ietf-tcpm-rack-04 (work in progress), July 2018.
[RFC3465] Allman, M., "TCP Congestion Control with Appropriate Byte
Counting (ABC)", RFC 3465, DOI 10.17487/RFC3465, February
2003, <https://www.rfc-editor.org/info/rfc3465>.
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[RFC4653] Bhandarkar, S., Reddy, A., Allman, M., and E. Blanton,
"Improving the Robustness of TCP to Non-Congestion
Events", RFC 4653, DOI 10.17487/RFC4653, August 2006,
<https://www.rfc-editor.org/info/rfc4653>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>.
[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
"Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
Spurious Retransmission Timeouts with TCP", RFC 5682,
DOI 10.17487/RFC5682, September 2009,
<https://www.rfc-editor.org/info/rfc5682>.
[RFC5827] Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
P. Hurtig, "Early Retransmit for TCP and Stream Control
Transmission Protocol (SCTP)", RFC 5827,
DOI 10.17487/RFC5827, May 2010,
<https://www.rfc-editor.org/info/rfc5827>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<https://www.rfc-editor.org/info/rfc6298>.
[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012,
<https://www.rfc-editor.org/info/rfc6582>.
[RFC6675] Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
and Y. Nishida, "A Conservative Loss Recovery Algorithm
Based on Selective Acknowledgment (SACK) for TCP",
RFC 6675, DOI 10.17487/RFC6675, August 2012,
<https://www.rfc-editor.org/info/rfc6675>.
[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>.
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[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>.
[TLP] Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
"Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
Tail Losses", draft-dukkipati-tcpm-tcp-loss-probe-01 (work
in progress), February 2013.
10.3. URIs
[1] https://mailarchive.ietf.org/arch/search/?email_list=quic
[2] https://github.com/quicwg
[3] https://github.com/quicwg/base-drafts/labels/-recovery
Appendix A. 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, it SHOULD be tracked 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.
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in_flight: A boolean that indicates whether the packet counts
towards bytes in flight.
is_crypto_packet: A boolean that indicates whether the packet
contains cryptographic handshake messages critical to the
completion of the QUIC handshake. In this version of QUIC, this
includes any packet with the long header that includes a CRYPTO
frame.
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. Some may need to be changed or
negotiated in order to better suit a variety of environments.
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,
}
A.3. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
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loss_detection_timer: Multi-modal timer used for loss detection.
crypto_count: The number of times all unacknowledged CRYPTO data has
been retransmitted without receiving an ack.
pto_count: The number of times a PTO has been sent without receiving
an ack.
time_of_last_sent_ack_eliciting_packet: The time the most recent
ack-eliciting packet was sent.
time_of_last_sent_crypto_packet: The time the most recent crypto
packet was sent.
largest_acked_packet[kPacketNumberSpace]: The largest packet number
acknowledged in the packet number space so far.
latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as
described in [RFC6298]
rttvar: The RTT variance, computed as described in [RFC6298]
min_rtt: The minimum RTT seen in the connection, ignoring ack delay.
max_ack_delay: The maximum amount of time by which the receiver
intends to delay acknowledgments, in milliseconds. The actual
ack_delay in a received ACK frame may be larger due to late
timers, reordering, or lost ACKs.
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()
crypto_count = 0
pto_count = 0
latest_rtt = 0
smoothed_rtt = 0
rttvar = 0
min_rtt = 0
time_of_last_sent_ack_eliciting_packet = 0
time_of_last_sent_crypto_packet = 0
for pn_space in [ Initial, Handshake, ApplicationData ]:
largest_acked_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, is_crypto_packet, 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 (is_crypto_packet):
time_of_last_sent_crypto_packet = now
if (ack_eliciting):
time_of_last_sent_ack_eliciting_packet = 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):
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][ack.largest_acked] &&
IncludesAckEliciting(newly_acked_packets))
latest_rtt =
now - sent_packets[pn_space][ack.largest_acked].time_sent
UpdateRtt(ack.ack_delay)
// Process ECN information if present.
if (ACK frame contains ECN information):
ProcessECN(ack)
for acked_packet in newly_acked_packets:
OnPacketAcked(acked_packet.packet_number, pn_space)
DetectLostPackets(pn_space)
crypto_count = 0
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)
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
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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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// Returns the earliest loss_time and the packet number
// space it's from. Returns 0 if all times are 0.
GetEarliestLossTime():
time = loss_time[Initial]
space = Initial
for pn_space in [ Handshake, ApplicationData ]:
if loss_time[pn_space] != 0 &&
(time == 0 || loss_time[pn_space] < time):
time = loss_time[pn_space];
space = pn_space
return time, space
SetLossDetectionTimer():
loss_time, _ = GetEarliestLossTime()
if (loss_time != 0):
// Time threshold loss detection.
loss_detection_timer.update(loss_time)
return
if (has unacknowledged crypto data
|| endpoint is client without 1-RTT keys):
// Crypto retransmission timer.
if (smoothed_rtt == 0):
timeout = 2 * kInitialRtt
else:
timeout = 2 * smoothed_rtt
timeout = max(timeout, kGranularity)
timeout = timeout * (2 ^ crypto_count)
loss_detection_timer.update(
time_of_last_sent_crypto_packet + timeout)
return
// Don't arm timer if there are no ack-eliciting packets
// in flight.
if (no ack-eliciting packets in flight):
loss_detection_timer.cancel()
return
// Calculate PTO duration
timeout =
smoothed_rtt + max(4 * rttvar, kGranularity) + max_ack_delay
timeout = timeout * (2 ^ pto_count)
loss_detection_timer.update(
time_of_last_sent_ack_eliciting_packet + 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():
loss_time, pn_space = GetEarliestLossTime()
if (loss_time != 0):
// Time threshold loss Detection
DetectLostPackets(pn_space)
// Retransmit crypto data if no packets were lost
// and there is crypto data to retransmit.
else if (has unacknowledged crypto data):
// Crypto retransmission timeout.
RetransmitUnackedCryptoData()
crypto_count++
else 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):
SendOneHandshakePacket()
else:
SendOnePaddedInitialPacket()
crypto_count++
else:
// PTO. Send new data if available, else retransmit old data.
// If neither is available, send a single PING frame.
SendOneOrTwoPackets()
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):
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
// Packets with packet numbers before this are deemed lost.
lost_pn = largest_acked_packet[pn_space] - kPacketThreshold
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 ||
unacked.packet_number <= lost_pn):
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 7.
B.1. Constants of interest
Constants used in congestion control are based on a combination of
RFCs, papers, and common practice. Some may need to be changed or
negotiated in order to better suit a variety of environments.
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kMaxDatagramSize: The sender's maximum payload size. Does not
include UDP or IP overhead. The max packet size is used for
calculating initial and minimum congestion windows. The
RECOMMENDED value is 1200 bytes.
kInitialWindow: Default limit on the initial amount of data in
flight, in bytes. Taken from [RFC6928], but increased slightly to
account for the smaller 8 byte overhead of UDP vs 20 bytes for
TCP. The RECOMMENDED value is the minimum of 10 *
kMaxDatagramSize and max(2* kMaxDatagramSize, 14720)).
kMinimumWindow: Minimum congestion window in bytes. The RECOMMENDED
value is 2 * kMaxDatagramSize.
kLossReductionFactor: Reduction in congestion window when a new loss
event is detected. The RECOMMENDED value is 0.5.
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) [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.
ecn_ce_counter: The highest value reported for the ECN-CE counter by
the peer in an ACK frame. This variable is used to detect
increases in the reported ECN-CE counter.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one 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
ecn_ce_counter = 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 (IsAppLimited())
// Do not increase congestion_window if application
// limited.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packet.size
else:
// Congestion avoidance.
congestion_window += kMaxDatagramSize * 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):
// If the ECN-CE counter reported by the peer has increased,
// this could be a new congestion event.
if (ack.ce_counter > ecn_ce_counter):
ecn_ce_counter = ack.ce_counter
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 window before the
// newest lost packet, including the edges, are marked
// lost
return IsWindowLost(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-19
o Send a PING if the PTO timer fires and there's nothing to send
(#2624)
o Set loss delay to at least kGranularity (#2617)
o Merge application limited and sending after idle sections. Always
limit burst size instead of requiring resetting CWND to initial
CWND after idle (#2605)
o Rewrite RTT estimation, allow RTT samples where a newly acked
packet is ack-eliciting but the largest_acked is not (#2592)
o Don't arm the handshake timer if there is no handshake data
(#2590)
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o Clarify that the time threshold loss alarm takes precedence over
the crypto handshake timer (#2590, #2620)
o Change initial RTT to 500ms to align with RFC6298 (#2184)
C.2. Since draft-ietf-quic-recovery-18
o Change IW byte limit to 14720 from 14600 (#2494)
o Update PTO calculation to match RFC6298 (#2480, #2489, #2490)
o Improve loss detection's description of multiple packet number
spaces and pseudocode (#2485, #2451, #2417)
o Declare persistent congestion even if non-probe packets are sent
and don't make persistent congestion more aggressive than RTO
verified was (#2365, #2244)
o Move pseudocode to the appendices (#2408)
o What to send on multiple PTOs (#2380)
C.3. Since draft-ietf-quic-recovery-17
o After Probe Timeout discard in-flight packets or send another
(#2212, #1965)
o Endpoints discard initial keys as soon as handshake keys are
available (#1951, #2045)
o 0-RTT state is discarded when 0-RTT is rejected (#2300)
o Loss detection timer is cancelled when ack-eliciting frames are in
flight (#2117, #2093)
o Packets are declared lost if they are in flight (#2104)
o After becoming idle, either pace packets or reset the congestion
controller (#2138, 2187)
o Process ECN counts before marking packets lost (#2142)
o Mark packets lost before resetting crypto_count and pto_count
(#2208, #2209)
o Congestion and loss recovery state are discarded when keys are
discarded (#2327)
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C.4. Since draft-ietf-quic-recovery-16
o Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP
and min crypto timeouts; eliminate timeout validation (#2114,
#2166, #2168, #1017)
o Redefine how congestion avoidance in terms of when the period
starts (#1928, #1930)
o Document what needs to be tracked for packets that are in flight
(#765, #1724, #1939)
o Integrate both time and packet thresholds into loss detection
(#1969, #1212, #934, #1974)
o Reduce congestion window after idle, unless pacing is used (#2007,
#2023)
o Disable RTT calculation for packets that don't elicit
acknowledgment (#2060, #2078)
o Limit ack_delay by max_ack_delay (#2060, #2099)
o Initial keys are discarded once Handshake are avaialble (#1951,
#2045)
o Reorder ECN and loss detection in pseudocode (#2142)
o Only cancel loss detection timer if ack-eliciting packets are in
flight (#2093, #2117)
C.5. Since draft-ietf-quic-recovery-14
o Used max_ack_delay from transport params (#1796, #1782)
o Merge ACK and ACK_ECN (#1783)
C.6. Since draft-ietf-quic-recovery-13
o Corrected the lack of ssthresh reduction in CongestionEvent
pseudocode (#1598)
o Considerations for ECN spoofing (#1426, #1626)
o Clarifications for PADDING and congestion control (#837, #838,
#1517, #1531, #1540)
o Reduce early retransmission timer to RTT/8 (#945, #1581)
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o Packets are declared lost after an RTO is verified (#935, #1582)
C.7. Since draft-ietf-quic-recovery-12
o Changes to manage separate packet number spaces and encryption
levels (#1190, #1242, #1413, #1450)
o Added ECN feedback mechanisms and handling; new ACK_ECN frame
(#804, #805, #1372)
C.8. Since draft-ietf-quic-recovery-11
No significant changes.
C.9. Since draft-ietf-quic-recovery-10
o Improved text on ack generation (#1139, #1159)
o Make references to TCP recovery mechanisms informational (#1195)
o Define time_of_last_sent_handshake_packet (#1171)
o Added signal from TLS the data it includes needs to be sent in a
Retry packet (#1061, #1199)
o Minimum RTT (min_rtt) is initialized with an infinite value
(#1169)
C.10. Since draft-ietf-quic-recovery-09
No significant changes.
C.11. Since draft-ietf-quic-recovery-08
o Clarified pacing and RTO (#967, #977)
C.12. Since draft-ietf-quic-recovery-07
o Include Ack Delay in RTO(and TLP) computations (#981)
o Ack Delay in SRTT computation (#961)
o Default RTT and Slow Start (#590)
o Many editorial fixes.
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C.13. Since draft-ietf-quic-recovery-06
No significant changes.
C.14. Since draft-ietf-quic-recovery-05
o Add more congestion control text (#776)
C.15. Since draft-ietf-quic-recovery-04
No significant changes.
C.16. Since draft-ietf-quic-recovery-03
No significant changes.
C.17. Since draft-ietf-quic-recovery-02
o Integrate F-RTO (#544, #409)
o Add congestion control (#545, #395)
o Require connection abort if a skipped packet was acknowledged
(#415)
o Simplify RTO calculations (#142, #417)
C.18. Since draft-ietf-quic-recovery-01
o Overview added to loss detection
o Changes initial default RTT to 100ms
o Added time-based loss detection and fixes early retransmit
o Clarified loss recovery for handshake packets
o Fixed references and made TCP references informative
C.19. Since draft-ietf-quic-recovery-00
o Improved description of constants and ACK behavior
C.20. Since draft-iyengar-quic-loss-recovery-01
o Adopted as base for draft-ietf-quic-recovery
o Updated authors/editors list
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o Added table of contents
Acknowledgments
Authors' Addresses
Jana Iyengar (editor)
Fastly
Email: jri.ietf@gmail.com
Ian Swett (editor)
Google
Email: ianswett@google.com
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