QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track I. Swett, Ed.
Expires: April 6, 2019 Google
October 03, 2018
QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-15
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 April 6, 2019.
Copyright Notice
Copyright (c) 2018 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 4
3. Design of the QUIC Transmission Machinery . . . . . . . . . . 4
3.1. Relevant Differences Between QUIC and TCP . . . . . . . . 5
3.1.1. Separate Packet Number Spaces . . . . . . . . . . . . 5
3.1.2. Monotonically Increasing Packet Numbers . . . . . . . 5
3.1.3. No Reneging . . . . . . . . . . . . . . . . . . . . . 6
3.1.4. More ACK Ranges . . . . . . . . . . . . . . . . . . . 6
3.1.5. Explicit Correction For Delayed ACKs . . . . . . . . 6
4. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 7
4.1. Computing the RTT estimate . . . . . . . . . . . . . . . 7
4.2. Ack-based Detection . . . . . . . . . . . . . . . . . . . 7
4.2.1. Fast Retransmit . . . . . . . . . . . . . . . . . . . 7
4.2.2. Early Retransmit . . . . . . . . . . . . . . . . . . 8
4.3. Timer-based Detection . . . . . . . . . . . . . . . . . . 9
4.3.1. Crypto Handshake Timeout . . . . . . . . . . . . . . 9
4.3.2. Tail Loss Probe . . . . . . . . . . . . . . . . . . . 10
4.3.3. Retransmission Timeout . . . . . . . . . . . . . . . 11
4.4. Generating Acknowledgements . . . . . . . . . . . . . . . 12
4.4.1. Crypto Handshake Data . . . . . . . . . . . . . . . . 12
4.4.2. ACK Ranges . . . . . . . . . . . . . . . . . . . . . 13
4.4.3. Receiver Tracking of ACK Frames . . . . . . . . . . . 13
4.5. Pseudocode . . . . . . . . . . . . . . . . . . . . . . . 13
4.5.1. Constants of interest . . . . . . . . . . . . . . . . 13
4.5.2. Variables of interest . . . . . . . . . . . . . . . . 14
4.5.3. Initialization . . . . . . . . . . . . . . . . . . . 15
4.5.4. On Sending a Packet . . . . . . . . . . . . . . . . . 16
4.5.5. On Receiving an Acknowledgment . . . . . . . . . . . 17
4.5.6. On Packet Acknowledgment . . . . . . . . . . . . . . 19
4.5.7. Setting the Loss Detection Timer . . . . . . . . . . 19
4.5.8. On Timeout . . . . . . . . . . . . . . . . . . . . . 21
4.5.9. Detecting Lost Packets . . . . . . . . . . . . . . . 22
4.6. Discussion . . . . . . . . . . . . . . . . . . . . . . . 23
5. Congestion Control . . . . . . . . . . . . . . . . . . . . . 23
5.1. Explicit Congestion Notification . . . . . . . . . . . . 24
5.2. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 24
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5.3. Congestion Avoidance . . . . . . . . . . . . . . . . . . 24
5.4. Recovery Period . . . . . . . . . . . . . . . . . . . . . 24
5.5. Tail Loss Probe . . . . . . . . . . . . . . . . . . . . . 25
5.6. Retransmission Timeout . . . . . . . . . . . . . . . . . 25
5.7. Pacing . . . . . . . . . . . . . . . . . . . . . . . . . 25
5.8. Pseudocode . . . . . . . . . . . . . . . . . . . . . . . 26
5.8.1. Constants of interest . . . . . . . . . . . . . . . . 26
5.8.2. Variables of interest . . . . . . . . . . . . . . . . 26
5.8.3. Initialization . . . . . . . . . . . . . . . . . . . 27
5.8.4. On Packet Sent . . . . . . . . . . . . . . . . . . . 27
5.8.5. On Packet Acknowledgement . . . . . . . . . . . . . . 27
5.8.6. On New Congestion Event . . . . . . . . . . . . . . . 28
5.8.7. Process ECN Information . . . . . . . . . . . . . . . 28
5.8.8. On Packets Lost . . . . . . . . . . . . . . . . . . . 28
5.8.9. On Retransmission Timeout Verified . . . . . . . . . 29
6. Security Considerations . . . . . . . . . . . . . . . . . . . 29
6.1. Congestion Signals . . . . . . . . . . . . . . . . . . . 29
6.2. Traffic Analysis . . . . . . . . . . . . . . . . . . . . 29
6.3. Misreporting ECN Markings . . . . . . . . . . . . . . . . 29
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 30
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 30
8.1. Normative References . . . . . . . . . . . . . . . . . . 30
8.2. Informative References . . . . . . . . . . . . . . . . . 30
8.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 31
Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 32
A.1. Since draft-ietf-quic-recovery-14 . . . . . . . . . . . . 32
A.2. Since draft-ietf-quic-recovery-13 . . . . . . . . . . . . 32
A.3. Since draft-ietf-quic-recovery-12 . . . . . . . . . . . . 32
A.4. Since draft-ietf-quic-recovery-11 . . . . . . . . . . . . 32
A.5. Since draft-ietf-quic-recovery-10 . . . . . . . . . . . . 32
A.6. Since draft-ietf-quic-recovery-09 . . . . . . . . . . . . 33
A.7. Since draft-ietf-quic-recovery-08 . . . . . . . . . . . . 33
A.8. Since draft-ietf-quic-recovery-07 . . . . . . . . . . . . 33
A.9. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 33
A.10. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 33
A.11. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 33
A.12. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 33
A.13. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 33
A.14. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 34
A.15. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 34
A.16. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 34
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 34
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 34
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1. Introduction
QUIC is a new multiplexed and secure transport atop UDP. QUIC builds
on decades of transport and security experience, and implements
mechanisms that make it attractive as a modern general-purpose
transport. The QUIC protocol is described in [QUIC-TRANSPORT].
QUIC implements the spirit of known TCP loss recovery mechanisms,
described in RFCs, various Internet-drafts, and also those prevalent
in the Linux TCP implementation. This document describes QUIC
congestion control and loss recovery, and where applicable,
attributes the TCP equivalent in RFCs, Internet-drafts, academic
papers, and/or TCP implementations.
2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"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 an ACK frame.
In-flight: Packets are considered in-flight when they have been sent
and neither acknowledged nor declared lost, and they are not ACK-
only.
Retransmittable Frames: All frames besides ACK or PADDING are
considered retransmittable.
Retransmittable Packets: Packets that contain retransmittable frames
elicit an ACK from the receiver and are called retransmittable
packets.
3. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
indicates the encryption level and includes a packet sequence number
(referred to below as a packet number). The encryption level
indicates the packet number space, as described in [QUIC-TRANSPORT].
Packet numbers never repeat within a packet number space for the
lifetime of a connection. Packet numbers monotonically increase
within a space, preventing ambiguity.
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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 only ACK
and PADDING frames are not acknowledged immediately.
o Long header packets that contain CRYPTO frames are critical to the
performance of the QUIC handshake and use shorter timers for
acknowledgement and retransmission.
o Packets that contain only ACK frames do not count toward
congestion control limits and are not considered in-flight. Note
that this means PADDING frames cause packets to contribute toward
bytes in flight without directly causing an acknowledgment to be
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.
3.1.1. Separate Packet Number Spaces
QUIC uses separate packet number spaces for each encryption level,
except 0-RTT and all generations of 1-RTT keys use the same packet
number space. Separate packet number spaces ensures acknowledgement
of packets sent with one level of encryption will not cause spurious
retransmission of packets sent with a different encryption level.
Congestion control and RTT measurement are unified across packet
number spaces.
3.1.2. Monotonically Increasing Packet Numbers
TCP conflates transmission sequence number at the sender with
delivery sequence number at the receiver, which results in
retransmissions of the same data carrying the same sequence number,
and consequently to problems caused by "retransmission ambiguity".
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QUIC separates the two: QUIC uses a packet number for transmissions,
and any application data is sent in one or more streams, with
delivery order determined by stream offsets encoded within STREAM
frames.
QUIC's packet number is strictly increasing, and directly encodes
transmission order. A higher QUIC packet number signifies that the
packet was sent later, and a lower QUIC packet number signifies that
the packet was sent earlier. When a packet containing frames is
deemed lost, QUIC rebundles necessary frames in a new packet with a
new packet number, removing ambiguity about which packet is
acknowledged when an ACK is received. Consequently, more accurate
RTT measurements can be made, spurious retransmissions are trivially
detected, and mechanisms such as Fast Retransmit can be applied
universally, based only on packet number.
This design point significantly simplifies loss detection mechanisms
for QUIC. Most TCP mechanisms implicitly attempt to infer
transmission ordering based on TCP sequence numbers - a non-trivial
task, especially when TCP timestamps are not available.
3.1.3. No Reneging
QUIC ACKs contain information that is similar to TCP SACK, but QUIC
does not allow any acked packet to be reneged, greatly simplifying
implementations on both sides and reducing memory pressure on the
sender.
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 ACKs
QUIC ACKs explicitly encode the delay incurred at the receiver
between when a packet is received and when the corresponding ACK is
sent. This allows the receiver of the ACK to adjust for receiver
delays, specifically the delayed ack timer, when estimating the path
RTT. This mechanism also allows a receiver to measure and report the
delay from when a packet was received by the OS kernel, which is
useful in receivers which may incur delays such as context-switch
latency before a userspace QUIC receiver processes a received packet.
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4. Loss Detection
QUIC senders use both ack information and timeouts to detect lost
packets, and this section provides a description of these algorithms.
Estimating the network round-trip time (RTT) is critical to these
algorithms and is described first.
4.1. Computing the RTT estimate
RTT is calculated when an ACK frame arrives by computing the
difference between the current time and the time the largest newly
acked packet was sent. If no packets are newly acknowledged, RTT
cannot be calculated. When RTT is calculated, the ack delay field
from the ACK frame SHOULD be subtracted from the RTT as long as the
result is larger than the Min RTT. If the result is smaller than the
min_rtt, the RTT should be used, but the ack delay field should be
ignored.
Like TCP, QUIC calculates both smoothed RTT and RTT variance similar
to those specified in [RFC6298].
Min RTT is the minimum RTT measured over the connection, prior to
adjusting by ack delay. Ignoring ack delay for min RTT prevents
intentional or unintentional underestimation of min RTT, which in
turn prevents underestimating smoothed RTT.
4.2. Ack-based Detection
Ack-based loss detection implements the spirit of TCP's Fast
Retransmit [RFC5681], Early Retransmit [RFC5827], FACK, and SACK loss
recovery [RFC6675]. This section provides an overview of how these
algorithms are implemented in QUIC.
4.2.1. Fast Retransmit
An unacknowledged packet is marked as lost when an acknowledgment is
received for a packet that was sent a threshold number of packets
(kReorderingThreshold) and/or a threshold amount of time after the
unacknowledged packet. Receipt of the acknowledgement indicates that
a later packet was received, while the reordering threshold provides
some tolerance for reordering of packets in the network.
The RECOMMENDED initial value for kReorderingThreshold is 3, based on
TCP loss recovery [RFC5681] [RFC6675]. Some networks may exhibit
higher degrees of reordering, causing a sender to detect spurious
losses. Spuriously declaring packets lost leads to unnecessary
retransmissions and may result in degraded performance due to the
actions of the congestion controller upon detecting loss.
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Implementers MAY use algorithms developed for TCP, such as TCP-NCR
[RFC4653], to improve QUIC's reordering resilience.
QUIC implementations can use time-based loss detection to handle
reordering based on time elapsed since the packet was sent. This may
be used either as a replacement for a packet reordering threshold or
in addition to it. The RECOMMENDED time threshold, expressed as a
fraction of the round-trip time (kTimeReorderingFraction), is 1/8.
4.2.2. Early Retransmit
Unacknowledged packets close to the tail may have fewer than
kReorderingThreshold retransmittable packets sent after them. Loss
of such packets cannot be detected via Fast Retransmit. To enable
ack-based loss detection of such packets, receipt of an
acknowledgment for the last outstanding retransmittable packet
triggers the Early Retransmit process, as follows.
If there are unacknowledged in-flight packets still pending, they
should be marked as lost. To compensate for the reduced reordering
resilience, the sender SHOULD set a timer for a small period of time.
If the unacknowledged in-flight packets are not acknowledged during
this time, then these packets MUST be marked as lost.
An endpoint SHOULD set the timer such that a packet is marked as lost
no earlier than 1.125 * max(SRTT, latest_RTT) since when it was sent.
Using max(SRTT, latest_RTT) protects from the two following cases:
o the latest RTT sample is lower than the SRTT, perhaps due to
reordering where packet whose ack triggered the Early Retransit
process encountered a shorter path;
o the latest RTT sample is higher than the SRTT, perhaps due to a
sustained increase in the actual RTT, but the smoothed SRTT has
not yet caught up.
The 1.125 multiplier increases reordering resilience. Implementers
MAY experiment with using other multipliers, bearing in mind that a
lower multiplier reduces reordering resilience and increases spurious
retransmissions, and a higher multiplier increases loss recovery
delay.
This mechanism is based on Early Retransmit for TCP [RFC5827].
However, [RFC5827] does not include the timer described above. Early
Retransmit is prone to spurious retransmissions due to its reduced
reordering resilence without the timer. This observation led Linux
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TCP implementers to implement a timer for TCP as well, and this
document incorporates this advancement.
4.3. Timer-based Detection
Timer-based loss detection recovers from losses that cannot be
handled by ack-based loss detection. It uses a single timer which
switches between a handshake retransmission timer, a Tail Loss Probe
timer and Retransmission Timeout mechanisms.
4.3.1. Crypto Handshake Timeout
Data in CRYPTO frames is critical to QUIC transport and crypto
negotiation, so a more aggressive timeout is used to retransmit it.
Below, the term "handshake packet" is used to refer to packets
containing CRYPTO frames, not packets with the specific long header
packet type Handshake.
The initial handshake timeout SHOULD be set to twice the initial RTT.
At the beginning, there are no prior RTT samples within a connection.
Resumed connections over the same network SHOULD use the previous
connection's final smoothed RTT value as the resumed connection's
initial RTT.
If no previous RTT is available, or if the network changes, the
initial RTT SHOULD be set to 100ms.
When CRYPTO frames are sent, the sender SHOULD set a timer for the
handshake timeout period. Upon timeout, the sender MUST retransmit
all unacknowledged CRYPTO data by calling
RetransmitAllUnackedHandshakeData(). On each consecutive expiration
of the handshake timer without receiving an acknowledgement for a new
packet, the sender SHOULD double the handshake timeout and set a
timer for this period.
When CRYPTO frames are outstanding, the TLP and RTO timers are not
active unless the CRYPTO frames were sent at 1-RTT encryption.
When an acknowledgement is received for a handshake packet, the new
RTT is computed and the timer SHOULD be set for twice the newly
computed smoothed RTT.
4.3.1.1. Retry and Version Negotiation
A Retry or Version Negotiation packet causes a client to send another
Initial packet, effectively restarting the connection process.
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Either packet indicates that the Initial was received but not
processed. Neither packet can be treated as an acknowledgment for
the Initial, but they MAY be used to improve the RTT estimate.
4.3.2. Tail Loss Probe
The algorithm described in this section is an adaptation of the Tail
Loss Probe algorithm proposed for TCP [TLP].
A packet sent at the tail is particularly vulnerable to slow loss
detection, since acks of subsequent packets are needed to trigger
ack-based detection. To ameliorate this weakness of tail packets,
the sender schedules a timer when the last retransmittable packet
before quiescence is transmitted. Upon timeout, a Tail Loss Probe
(TLP) packet is sent to evoke an acknowledgement from the receiver.
The timer duration, or Probe Timeout (PTO), is set based on the
following conditions:
o PTO SHOULD be scheduled for max(1.5*SRTT+MaxAckDelay,
kMinTLPTimeout)
o If RTO (Section 4.3.3) is earlier, schedule a TLP in its place.
That is, PTO SHOULD be scheduled for min(RTO, PTO).
QUIC includes MaxAckDelay in all probe timeouts, because it assumes
the ack delay may come into play, regardless of the number of packets
outstanding. TCP's TLP assumes if at least 2 packets are
outstanding, acks will not be delayed.
A PTO value of at least 1.5*SRTT ensures that the ACK is overdue.
The 1.5 is based on [TLP], but implementations MAY experiment with
other constants.
To reduce latency, it is RECOMMENDED that the sender set and allow
the TLP timer to fire twice before setting an RTO timer. In other
words, when the TLP timer expires the first time, a TLP packet is
sent, and it is RECOMMENDED that the TLP timer be scheduled for a
second time. When the TLP timer expires the second time, a second
TLP packet is sent, and an RTO timer SHOULD be scheduled
Section 4.3.3.
A TLP packet SHOULD carry new data when possible. If new data is
unavailable or new data cannot be sent due to flow control, a TLP
packet MAY retransmit unacknowledged data to potentially reduce
recovery time. Since a TLP timer is used to send a probe into the
network prior to establishing any packet loss, prior unacknowledged
packets SHOULD NOT be marked as lost when a TLP timer expires.
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A sender may not know that a packet being sent is a tail packet.
Consequently, a sender may have to arm or adjust the TLP timer on
every sent retransmittable packet.
4.3.3. Retransmission Timeout
A Retransmission Timeout (RTO) timer is the final backstop for loss
detection. The algorithm used in QUIC is based on the RTO algorithm
for TCP [RFC5681] and is additionally resilient to spurious RTO
events [RFC5682].
When the last TLP packet is sent, a timer is set for the RTO period.
When this timer expires, the sender sends two packets, to evoke
acknowledgements from the receiver, and restarts the RTO timer.
Similar to TCP [RFC6298], the RTO period is set based on the
following conditions:
o When the final TLP packet is sent, the RTO period is set to
max(SRTT + 4*RTTVAR + MaxAckDelay, kMinRTOTimeout)
o When an RTO timer expires, the RTO period is doubled.
The sender typically has incurred a high latency penalty by the time
an RTO timer expires, and this penalty increases exponentially in
subsequent consecutive RTO events. Sending a single packet on an RTO
event therefore makes the connection very sensitive to single packet
loss. Sending two packets instead of one significantly increases
resilience to packet drop in both directions, thus reducing the
probability of consecutive RTO events.
QUIC's RTO algorithm differs from TCP in that the firing of an RTO
timer is not considered a strong enough signal of packet loss, so
does not result in an immediate change to congestion window or
recovery state. An RTO timer expires only when there's a prolonged
period of network silence, which could be caused by a change in the
underlying network RTT.
QUIC also diverges from TCP by including MaxAckDelay in the RTO
period. Since QUIC corrects for this delay in its SRTT and RTTVAR
computations, it is necessary to add this delay explicitly in the TLP
and RTO computation.
When an acknowledgment is received for a packet sent on an RTO event,
any unacknowledged packets with lower packet numbers than those
acknowledged MUST be marked as lost. If an acknowledgement for a
packet sent on an RTO is received at the same time packets sent prior
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to the first RTO are acknowledged, the RTO is considered spurious and
standard loss detection rules apply.
A packet sent when an RTO timer expires MAY carry new data if
available or unacknowledged data to potentially reduce recovery time.
Since this packet is sent as a probe into the network prior to
establishing any packet loss, prior unacknowledged packets SHOULD NOT
be marked as lost.
A packet sent on an RTO timer MUST NOT be blocked by the sender's
congestion controller. A sender MUST however count these bytes as
additional bytes in flight, since this packet adds network load
without establishing packet loss.
4.4. Generating Acknowledgements
QUIC SHOULD delay sending acknowledgements in response to packets,
but MUST NOT excessively delay acknowledgements of packets containing
frames other than ACK. Specifically, implementations MUST attempt to
enforce a maximum ack delay to avoid causing the peer spurious
timeouts. The maximum ack delay is communicated in the
"max_ack_delay" transport parameter and the default value is 25ms.
An acknowledgement SHOULD be sent immediately upon receipt of a
second packet but the delay SHOULD NOT exceed the maximum ack delay.
QUIC recovery algorithms do not assume the peer generates an
acknowledgement immediately when receiving a second full-packet.
Out-of-order packets SHOULD be acknowledged more quickly, in order to
accelerate loss recovery. The receiver SHOULD send an immediate ACK
when it receives a new packet which is not one greater than the
largest received packet number.
Similarly, packets marked with the ECN Congestion Experienced (CE)
codepoint in the IP header SHOULD be acknowledged immediately, to
reduce the peer's response time to congestion events.
As an optimization, a receiver MAY process multiple packets before
sending any ACK frames in response. In this case they can determine
whether an immediate or delayed acknowledgement should be generated
after processing incoming packets.
4.4.1. Crypto Handshake Data
In order to quickly complete the handshake and avoid spurious
retransmissions due to handshake timeouts, handshake packets SHOULD
use a very short ack delay, such as 1ms. ACK frames MAY be sent
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immediately when the crypto stack indicates all data for that
encryption level has been received.
4.4.2. ACK Ranges
When an ACK frame is sent, one or more ranges of acknowledged packets
are included. Including older packets reduces the chance of spurious
retransmits caused by losing previously sent ACK frames, at the cost
of larger ACK frames.
ACK frames SHOULD always acknowledge the most recently received
packets, and the more out-of-order the packets are, the more
important it is to send an updated ACK frame quickly, to prevent the
peer from declaring a packet as lost and spuriously retransmitting
the frames it contains.
Below is one recommended approach for determining what packets to
include in an ACK frame.
4.4.3. Receiver Tracking of ACK Frames
When a packet containing an ACK frame is sent, the largest
acknowledged in that frame may be saved. When a packet containing an
ACK frame is acknowledged, the receiver can stop acknowledging
packets less than or equal to the largest acknowledged in the sent
ACK frame.
In cases without ACK frame loss, this algorithm allows for a minimum
of 1 RTT of reordering. In cases with ACK frame loss, this approach
does not guarantee that every acknowledgement is seen by the sender
before it is no longer included in the ACK frame. Packets could be
received out of order and all subsequent ACK frames containing them
could be lost. In this case, the loss recovery algorithm may cause
spurious retransmits, but the sender will continue making forward
progress.
4.5. Pseudocode
4.5.1. Constants of interest
Constants used in loss recovery are based on a combination of RFCs,
papers, and common practice. Some may need to be changed or
negotiated in order to better suit a variety of environments.
kMaxTLPs: Maximum number of tail loss probes before an RTO expires.
The RECOMMENDED value is 2.
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kReorderingThreshold: Maximum reordering in packet number space
before FACK style loss detection considers a packet lost. The
RECOMMENDED value is 3.
kTimeReorderingFraction: Maximum reordering in time space before
time based loss detection considers a packet lost. In fraction of
an RTT. The RECOMMENDED value is 1/8.
kUsingTimeLossDetection: Whether time based loss detection is in
use. If false, uses FACK style loss detection. The RECOMMENDED
value is false.
kMinTLPTimeout: Minimum time in the future a tail loss probe timer
may be set for. The RECOMMENDED value is 10ms.
kMinRTOTimeout: Minimum time in the future an RTO timer may be set
for. The RECOMMENDED value is 200ms.
kDelayedAckTimeout: The length of the peer's delayed ack timer. The
RECOMMENDED value is 25ms.
kInitialRtt: The RTT used before an RTT sample is taken. The
RECOMMENDED value is 100ms.
4.5.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
loss_detection_timer: Multi-modal timer used for loss detection.
handshake_count: The number of times all unacknowledged handshake
data has been retransmitted without receiving an ack.
tlp_count: The number of times a tail loss probe has been sent
without receiving an ack.
rto_count: The number of times an RTO has been sent without
receiving an ack.
largest_sent_before_rto: The last packet number sent prior to the
first retransmission timeout.
time_of_last_sent_retransmittable_packet: The time the most recent
retransmittable packet was sent.
time_of_last_sent_handshake_packet: The time the most recent packet
containing a CRYPTO frame was sent.
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largest_sent_packet: The packet number of the most recently sent
packet.
largest_acked_packet: The largest packet number acknowledged in an
ACK frame.
latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as
described in [RFC6298]
rttvar: The RTT variance, computed as described in [RFC6298]
min_rtt: The minimum RTT seen in the connection, ignoring ack delay.
max_ack_delay: The maximum amount of time by which the receiver
intends to delay acknowledgments, in milliseconds. The actual
ack_delay in a received ACK frame may be larger due to late
timers, reordering, or lost ACKs.
reordering_threshold: The largest packet number gap between the
largest acknowledged retransmittable packet and an unacknowledged
retransmittable packet before it is declared lost.
time_reordering_fraction: The reordering window as a fraction of
max(smoothed_rtt, latest_rtt).
loss_time: The time at which the next packet will be considered lost
based on early transmit or exceeding the reordering window in
time.
sent_packets: An association of packet numbers to information about
them, including a number field indicating the packet number, a
time field indicating the time a packet was sent, a boolean
indicating whether the packet is ack-only, a boolean indicating
whether it counts towards bytes in flight, and a bytes field
indicating the packet's size. sent_packets is ordered by packet
number, and packets remain in sent_packets until acknowledged or
lost. A sent_packets data structure is maintained per packet
number space, and ACK processing only applies to a single space.
4.5.3. Initialization
At the beginning of the connection, initialize the loss detection
variables as follows:
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loss_detection_timer.reset()
handshake_count = 0
tlp_count = 0
rto_count = 0
if (kUsingTimeLossDetection)
reordering_threshold = infinite
time_reordering_fraction = kTimeReorderingFraction
else:
reordering_threshold = kReorderingThreshold
time_reordering_fraction = infinite
loss_time = 0
smoothed_rtt = 0
rttvar = 0
min_rtt = infinite
largest_sent_before_rto = 0
time_of_last_sent_retransmittable_packet = 0
time_of_last_sent_handshake_packet = 0
largest_sent_packet = 0
4.5.4. On Sending a Packet
After any packet is sent, be it a new transmission or a rebundled
transmission, the following OnPacketSent function is called. The
parameters to OnPacketSent are as follows:
o packet_number: The packet number of the sent packet.
o ack_only: A boolean that indicates whether a packet contains only
ACK or PADDING frame(s). If true, it is still expected an ack
will be received for this packet, but it is not retransmittable.
o in_flight: A boolean that indicates whether the packet counts
towards bytes in flight.
o is_handshake_packet: 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.
o sent_bytes: The number of bytes sent in the packet, not including
UDP or IP overhead, but including QUIC framing overhead.
Pseudocode for OnPacketSent follows:
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OnPacketSent(packet_number, ack_only, in_flight,
is_handshake_packet, sent_bytes):
largest_sent_packet = packet_number
sent_packets[packet_number].packet_number = packet_number
sent_packets[packet_number].time = now
sent_packets[packet_number].ack_only = ack_only
sent_packets[packet_number].in_flight = in_flight
if !ack_only:
if is_handshake_packet:
time_of_last_sent_handshake_packet = now
time_of_last_sent_retransmittable_packet = now
OnPacketSentCC(sent_bytes)
sent_packets[packet_number].bytes = sent_bytes
SetLossDetectionTimer()
4.5.5. On Receiving an Acknowledgment
When an ACK frame is received, it may newly acknowledge any number of
packets.
Pseudocode for OnAckReceived and UpdateRtt follow:
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OnAckReceived(ack):
largest_acked_packet = ack.largest_acked
// If the largest acknowledged is newly acked,
// update the RTT.
if (sent_packets[ack.largest_acked]):
latest_rtt = now - sent_packets[ack.largest_acked].time
UpdateRtt(latest_rtt, ack.ack_delay)
// Find all newly acked packets in this ACK frame
newly_acked_packets = DetermineNewlyAckedPackets(ack)
for acked_packet in newly_acked_packets:
OnPacketAcked(acked_packet.packet_number)
if !newly_acked_packets.empty():
// Find the smallest newly acknowledged packet
smallest_newly_acked =
FindSmallestNewlyAcked(newly_acked_packets)
// If any packets sent prior to RTO were acked, then the
// RTO was spurious. Otherwise, inform congestion control.
if (rto_count > 0 &&
smallest_newly_acked > largest_sent_before_rto):
OnRetransmissionTimeoutVerified(smallest_newly_acked)
handshake_count = 0
tlp_count = 0
rto_count = 0
DetectLostPackets(ack.largest_acked_packet)
SetLossDetectionTimer()
// Process ECN information if present.
if (ACK frame contains ECN information):
ProcessECN(ack)
UpdateRtt(latest_rtt, ack_delay):
// min_rtt ignores ack delay.
min_rtt = min(min_rtt, latest_rtt)
// Adjust for ack delay if it's plausible.
if (latest_rtt - min_rtt > ack_delay):
latest_rtt -= ack_delay
// Based on {{RFC6298}}.
if (smoothed_rtt == 0):
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
else:
rttvar_sample = abs(smoothed_rtt - latest_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt
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4.5.6. On Packet Acknowledgment
When a packet is acked for the first time, the following
OnPacketAcked function is called. Note that a single ACK frame may
newly acknowledge several packets. OnPacketAcked must be called once
for each of these newly acked packets.
OnPacketAcked takes one parameter, acked_packet, which is the struct
of the newly acked packet.
If this is the first acknowledgement following RTO, check if the
smallest newly acknowledged packet is one sent by the RTO, and if so,
inform congestion control of a verified RTO, similar to F-RTO
[RFC5682].
Pseudocode for OnPacketAcked follows:
OnPacketAcked(acked_packet):
if (!acked_packet.is_ack_only):
OnPacketAckedCC(acked_packet)
sent_packets.remove(acked_packet.packet_number)
4.5.7. Setting the Loss Detection Timer
QUIC loss detection uses a single timer for all timer-based loss
detection. The duration of the timer is based on the timer's mode,
which is set in the packet and timer events further below. The
function SetLossDetectionTimer defined below shows how the single
timer is set.
4.5.7.1. Handshake Timer
When a connection has unacknowledged handshake data, the handshake
timer is set and when it expires, all unacknowledgedd handshake data
is retransmitted.
When stateless rejects are in use, the connection is considered
immediately closed once a reject is sent, so no timer is set to
retransmit the reject.
Version negotiation packets are always stateless, and MUST be sent
once per handshake packet that uses an unsupported QUIC version, and
MAY be sent in response to 0-RTT packets.
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4.5.7.2. Tail Loss Probe and Retransmission Timer
Tail loss probes [TLP] and retransmission timeouts [RFC6298] are
timer based mechanisms to recover from cases when there are
outstanding retransmittable packets, but an acknowledgement has not
been received in a timely manner.
The TLP and RTO timers are armed when there is no unacknowledged
handshake data. The TLP timer is set until the max number of TLP
packets have been sent, and then the RTO timer is set.
4.5.7.3. Early Retransmit Timer
Early retransmit [RFC5827] is implemented with a 1/4 RTT timer. It
is part of QUIC's time based loss detection, but is always enabled,
even when only packet reordering loss detection is enabled.
4.5.7.4. Pseudocode
Pseudocode for SetLossDetectionTimer follows:
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SetLossDetectionTimer():
// Don't arm timer if there are no retransmittable packets
// in flight.
if (bytes_in_flight == 0):
loss_detection_timer.cancel()
return
if (handshake packets are outstanding):
// Handshake retransmission timer.
if (smoothed_rtt == 0):
timeout = 2 * kInitialRtt
else:
timeout = 2 * smoothed_rtt
timeout = max(timeout, kMinTLPTimeout)
timeout = timeout * (2 ^ handshake_count)
loss_detection_timer.set(
time_of_last_sent_handshake_packet + timeout)
return;
else if (loss_time != 0):
// Early retransmit timer or time loss detection.
timeout = loss_time -
time_of_last_sent_retransmittable_packet
else:
// RTO or TLP timer
// Calculate RTO duration
timeout =
smoothed_rtt + 4 * rttvar + max_ack_delay
timeout = max(timeout, kMinRTOTimeout)
timeout = timeout * (2 ^ rto_count)
if (tlp_count < kMaxTLPs):
// Tail Loss Probe
tlp_timeout = max(1.5 * smoothed_rtt
+ max_ack_delay, kMinTLPTimeout)
timeout = min(tlp_timeout, timeout)
loss_detection_timer.set(
time_of_last_sent_retransmittable_packet + timeout)
4.5.8. On Timeout
QUIC uses one loss recovery timer, which when set, can be in one of
several modes. When the timer expires, the mode determines the
action to be performed.
Pseudocode for OnLossDetectionTimeout follows:
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OnLossDetectionTimeout():
if (handshake packets are outstanding):
// Handshake timeout.
RetransmitAllUnackedHandshakeData()
handshake_count++
else if (loss_time != 0):
// Early retransmit or Time Loss Detection
DetectLostPackets(largest_acked_packet)
else if (tlp_count < kMaxTLPs):
// Tail Loss Probe.
SendOnePacket()
tlp_count++
else:
// RTO.
if (rto_count == 0)
largest_sent_before_rto = largest_sent_packet
SendTwoPackets()
rto_count++
SetLossDetectionTimer()
4.5.9. Detecting Lost Packets
Packets in QUIC are only considered lost once a larger packet number
in the same packet number space is acknowledged. DetectLostPackets
is called every time an ack is received and operates on the
sent_packets for that packet number space. If the loss detection
timer expires and the loss_time is set, the previous largest acked
packet is supplied.
4.5.9.1. Pseudocode
DetectLostPackets takes one parameter, acked, which is the largest
acked packet.
Pseudocode for DetectLostPackets follows:
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DetectLostPackets(largest_acked):
loss_time = 0
lost_packets = {}
delay_until_lost = infinite
if (kUsingTimeLossDetection):
delay_until_lost =
(1 + time_reordering_fraction) *
max(latest_rtt, smoothed_rtt)
else if (largest_acked.packet_number == largest_sent_packet):
// Early retransmit timer.
delay_until_lost = 9/8 * max(latest_rtt, smoothed_rtt)
foreach (unacked < largest_acked.packet_number):
time_since_sent = now() - unacked.time_sent
delta = largest_acked.packet_number - unacked.packet_number
if (time_since_sent > delay_until_lost ||
delta > reordering_threshold):
sent_packets.remove(unacked.packet_number)
if (!unacked.is_ack_only):
lost_packets.insert(unacked)
else if (loss_time == 0 && delay_until_lost != infinite):
loss_time = now() + delay_until_lost - time_since_sent
// Inform the congestion controller of lost packets and
// lets it decide whether to retransmit immediately.
if (!lost_packets.empty()):
OnPacketsLost(lost_packets)
4.6. Discussion
The majority of constants were derived from best common practices
among widely deployed TCP implementations on the internet.
Exceptions follow.
A shorter delayed ack time of 25ms was chosen because longer delayed
acks can delay loss recovery and for the small number of connections
where less than packet per 25ms is delivered, acking every packet is
beneficial to congestion control and loss recovery.
The default initial RTT of 100ms was chosen because it is slightly
higher than both the median and mean min_rtt typically observed on
the public internet.
5. Congestion Control
QUIC's congestion control is based on TCP NewReno [RFC6582]. NewReno
is a congestion window based congestion control. QUIC specifies the
congestion window in bytes rather than packets due to finer control
and the ease of appropriate byte counting [RFC3465].
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QUIC hosts MUST NOT send packets if they would increase
bytes_in_flight (defined in Section 5.8.2) beyond the available
congestion window, unless the packet is a probe packet sent after the
TLP or RTO timer expires, as described in Section 4.3.2 and
Section 4.3.3.
Implementations MAY use other congestion control algorithms, and
endpoints MAY use different algorithms from one another. The signals
QUIC provides for congestion control are generic and are designed to
support different algorithms.
5.1. Explicit Congestion Notification
If a path has been verified to support ECN, QUIC treats a Congestion
Experienced codepoint in the IP header as a signal of congestion.
This document specifies an endpoint's response when its peer receives
packets with the Congestion Experienced codepoint. As discussed in
[RFC8311], endpoints are permitted to experiment with other response
functions.
5.2. Slow Start
QUIC begins every connection in slow start and exits slow start upon
loss or upon increase in the ECN-CE counter. QUIC re-enters slow
start anytime the congestion window is less than ssthresh, which
typically only occurs after an RTO. While in slow start, QUIC
increases the congestion window by the number of bytes acknowledged
when each ack is processed.
5.3. Congestion Avoidance
Slow start exits to congestion avoidance. Congestion avoidance in
NewReno uses an additive increase multiplicative decrease (AIMD)
approach that increases the congestion window by one maximum packet
size per congestion window acknowledged. When a loss is detected,
NewReno halves the congestion window and sets the slow start
threshold to the new congestion window.
5.4. Recovery Period
Recovery is a period of time beginning with detection of a lost
packet or an increase in the ECN-CE counter. Because QUIC
retransmits stream data and control frames, not packets, it defines
the end of recovery as a packet sent after the start of recovery
being acknowledged. This is slightly different from TCP's definition
of recovery, which ends when the lost packet that started recovery is
acknowledged.
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The recovery period limits congestion window reduction to once per
round trip. During recovery, the congestion window remains unchanged
irrespective of new losses or increases in the ECN-CE counter.
5.5. Tail Loss Probe
A TLP packet MUST NOT be blocked by the sender's congestion
controller. The sender MUST however count these bytes as additional
bytes-in-flight, since a TLP adds network load without establishing
packet loss.
Acknowledgement or loss of tail loss probes are treated like any
other packet.
5.6. Retransmission Timeout
When retransmissions are sent due to a retransmission timeout timer,
no change is made to the congestion window until the next
acknowledgement arrives. The retransmission timeout is considered
spurious when this acknowledgement acknowledges packets sent prior to
the first retransmission timeout. The retransmission timeout is
considered valid when this acknowledgement acknowledges no packets
sent prior to the first retransmission timeout. In this case, the
congestion window MUST be reduced to the minimum congestion window
and slow start is re-entered.
5.7. Pacing
This document does not specify a pacer, but it is RECOMMENDED that a
sender pace sending of all in-flight packets based on input from the
congestion controller. For example, a pacer might distribute the
congestion window over the SRTT when used with a window-based
controller, and a pacer might use the rate estimate of a rate-based
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).
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5.8. Pseudocode
5.8.1. Constants of interest
Constants used in congestion control are based on a combination of
RFCs, papers, and common practice. Some may need to be changed or
negotiated in order to better suit a variety of environments.
kMaxDatagramSize: The sender's maximum payload size. Does not
include UDP or IP overhead. The max packet size is used for
calculating initial and minimum congestion windows. The
RECOMMENDED value is 1200 bytes.
kInitialWindow: Default limit on the initial amount of outstanding
data in bytes. Taken from [RFC6928]. The RECOMMENDED value is
the minimum of 10 * kMaxDatagramSize and max(2* kMaxDatagramSize,
14600)).
kMinimumWindow: Minimum congestion window in bytes. The RECOMMENDED
value is 2 * kMaxDatagramSize.
kLossReductionFactor: Reduction in congestion window when a new loss
event is detected. The RECOMMENDED value is 0.5.
5.8.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
ecn_ce_counter: The highest value reported for the ECN-CE counter by
the peer in an ACK frame. This variable is used to detect
increases in the reported ECN-CE counter.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one retransmittable or PADDING frame, and
have not been acked or declared lost. The size does not include
IP or UDP overhead, but does include the QUIC header and AEAD
overhead. Packets only containing ACK frames do not count towards
bytes_in_flight to ensure congestion control does not impede
congestion feedback.
congestion_window: Maximum number of bytes-in-flight that may be
sent.
end_of_recovery: The largest packet number sent when QUIC detects a
loss. When a larger packet is acknowledged, QUIC exits recovery.
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ssthresh: Slow start threshold in bytes. When the congestion window
is below ssthresh, the mode is slow start and the window grows by
the number of bytes acknowledged.
5.8.3. Initialization
At the beginning of the connection, initialize the congestion control
variables as follows:
congestion_window = kInitialWindow
bytes_in_flight = 0
end_of_recovery = 0
ssthresh = infinite
ecn_ce_counter = 0
5.8.4. On Packet Sent
Whenever a packet is sent, and it contains non-ACK frames, the packet
increases bytes_in_flight.
OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent
5.8.5. On Packet Acknowledgement
Invoked from loss detection's OnPacketAcked and is supplied with
acked_packet from sent_packets.
InRecovery(packet_number):
return packet_number <= end_of_recovery
OnPacketAckedCC(acked_packet):
// Remove from bytes_in_flight.
bytes_in_flight -= acked_packet.bytes
if (InRecovery(acked_packet.packet_number)):
// Do not increase congestion window in recovery period.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packet.bytes
else:
// Congestion avoidance.
congestion_window += kMaxDatagramSize * acked_packet.bytes
/ congestion_window
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5.8.6. On New Congestion Event
Invoked from ProcessECN and OnPacketsLost when a new congestion event
is detected. Starts a new recovery period and reduces the congestion
window.
CongestionEvent(packet_number):
// Start a new congestion event if packet_number
// is larger than the end of the previous recovery epoch.
if (!InRecovery(packet_number)):
end_of_recovery = largest_sent_packet
congestion_window *= kLossReductionFactor
congestion_window = max(congestion_window, kMinimumWindow)
ssthresh = congestion_window
5.8.7. Process ECN Information
Invoked when an ACK frame with an ECN section is received from the
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
// Start a new congestion event if the last acknowledged
// packet is past the end of the previous recovery epoch.
CongestionEvent(ack.largest_acked_packet)
5.8.8. On Packets Lost
Invoked by loss detection from DetectLostPackets when new packets are
detected lost.
OnPacketsLost(lost_packets):
// Remove lost packets from bytes_in_flight.
for (lost_packet : lost_packets):
bytes_in_flight -= lost_packet.bytes
largest_lost_packet = lost_packets.last()
// Start a new congestion epoch if the last lost packet
// is past the end of the previous recovery epoch.
CongestionEvent(largest_lost_packet.packet_number)
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5.8.9. On Retransmission Timeout Verified
QUIC decreases the congestion window to the minimum value once the
retransmission timeout has been verified and removes any packets sent
before the newly acknowledged RTO packet.
OnRetransmissionTimeoutVerified(packet_number)
congestion_window = kMinimumWindow
// Declare all packets prior to packet_number lost.
for (sent_packet: sent_packets):
if (sent_packet.packet_number < packet_number):
bytes_in_flight -= lost_packet.bytes
sent_packets.remove(sent_packet.packet_number)
6. Security Considerations
6.1. Congestion Signals
Congestion control fundamentally involves the consumption of signals
- both loss and ECN codepoints - from unauthenticated entities. On-
path attackers can spoof or alter these signals. An attacker can
cause endpoints to reduce their sending rate by dropping packets, or
alter send rate by changing ECN codepoints.
6.2. Traffic Analysis
Packets that carry only ACK frames can be heuristically identified by
observing packet size. Acknowledgement patterns may expose
information about link characteristics or application behavior.
Endpoints can use PADDING frames or bundle acknowledgments with other
frames to reduce leaked information.
6.3. Misreporting ECN Markings
A receiver can misreport ECN markings to alter the congestion
response of a sender. Suppressing reports of ECN-CE markings could
cause a sender to increase their send rate. This increase could
result in congestion and loss.
A sender MAY attempt to detect suppression of reports by marking
occasional packets that they send with ECN-CE. If a packet marked
with ECN-CE is not reported as having been marked when the packet is
acknowledged, the sender SHOULD then disable ECN for that path.
Reporting additional ECN-CE markings will cause a sender to reduce
their sending rate, which is similar in effect to advertising reduced
connection flow control limits and so no advantage is gained by doing
so.
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Endpoints choose the congestion controller that they use. Though
congestion controllers generally treat reports of ECN-CE markings as
equivalent to loss [RFC8311], the exact response for each controller
could be different. Failure to correctly respond to information
about ECN markings is therefore difficult to detect.
7. IANA Considerations
This document has no IANA actions. Yet.
8. References
8.1. Normative References
[QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", draft-ietf-quic-
transport-15 (work in progress), October 2018.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<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>.
8.2. Informative References
[RFC3465] Allman, M., "TCP Congestion Control with Appropriate Byte
Counting (ABC)", RFC 3465, DOI 10.17487/RFC3465, February
2003, <https://www.rfc-editor.org/info/rfc3465>.
[RFC4653] Bhandarkar, S., Reddy, A., Allman, M., and E. Blanton,
"Improving the Robustness of TCP to Non-Congestion
Events", RFC 4653, DOI 10.17487/RFC4653, August 2006,
<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>.
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Internet-Draft QUIC Loss Detection October 2018
[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>.
[TLP] Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
"Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
Tail Losses", draft-dukkipati-tcpm-tcp-loss-probe-01 (work
in progress), February 2013.
8.3. URIs
[1] https://mailarchive.ietf.org/arch/search/?email_list=quic
[2] https://github.com/quicwg
[3] https://github.com/quicwg/base-drafts/labels/-recovery
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Appendix A. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
A.1. Since draft-ietf-quic-recovery-14
o Used max_ack_delay from transport params (#1796, #1782)
o Merge ACK and ACK_ECN (#1783)
A.2. Since draft-ietf-quic-recovery-13
o Corrected the lack of ssthresh reduction in CongestionEvent
pseudocode (#1598)
o Considerations for ECN spoofing (#1426, #1626)
o Clarifications for PADDING and congestion control (#837, #838,
#1517, #1531, #1540)
o Reduce early retransmission timer to RTT/8 (#945, #1581)
o Packets are declared lost after an RTO is verified (#935, #1582)
A.3. Since draft-ietf-quic-recovery-12
o Changes to manage separate packet number spaces and encryption
levels (#1190, #1242, #1413, #1450)
o Added ECN feedback mechanisms and handling; new ACK_ECN frame
(#804, #805, #1372)
A.4. Since draft-ietf-quic-recovery-11
No significant changes.
A.5. Since draft-ietf-quic-recovery-10
o Improved text on ack generation (#1139, #1159)
o Make references to TCP recovery mechanisms informational (#1195)
o Define time_of_last_sent_handshake_packet (#1171)
o Added signal from TLS the data it includes needs to be sent in a
Retry packet (#1061, #1199)
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o Minimum RTT (min_rtt) is initialized with an infinite value
(#1169)
A.6. Since draft-ietf-quic-recovery-09
No significant changes.
A.7. Since draft-ietf-quic-recovery-08
o Clarified pacing and RTO (#967, #977)
A.8. Since draft-ietf-quic-recovery-07
o Include Ack Delay in RTO(and TLP) computations (#981)
o Ack Delay in SRTT computation (#961)
o Default RTT and Slow Start (#590)
o Many editorial fixes.
A.9. Since draft-ietf-quic-recovery-06
No significant changes.
A.10. Since draft-ietf-quic-recovery-05
o Add more congestion control text (#776)
A.11. Since draft-ietf-quic-recovery-04
No significant changes.
A.12. Since draft-ietf-quic-recovery-03
No significant changes.
A.13. Since draft-ietf-quic-recovery-02
o Integrate F-RTO (#544, #409)
o Add congestion control (#545, #395)
o Require connection abort if a skipped packet was acknowledged
(#415)
o Simplify RTO calculations (#142, #417)
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A.14. Since draft-ietf-quic-recovery-01
o Overview added to loss detection
o Changes initial default RTT to 100ms
o Added time-based loss detection and fixes early retransmit
o Clarified loss recovery for handshake packets
o Fixed references and made TCP references informative
A.15. Since draft-ietf-quic-recovery-00
o Improved description of constants and ACK behavior
A.16. Since draft-iyengar-quic-loss-recovery-01
o Adopted as base for draft-ietf-quic-recovery
o Updated authors/editors list
o Added table of contents
Acknowledgments
Authors' Addresses
Jana Iyengar (editor)
Fastly
Email: jri.ietf@gmail.com
Ian Swett (editor)
Google
Email: ianswett@google.com
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