QUIC J. Iyengar, Ed.
Internet-Draft I. Swett, Ed.
Intended status: Standards Track Google
Expires: May 18, 2018 November 14, 2017
QUIC Loss Detection and Congestion Control
draft-ietf-quic-recovery-07
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 May 18, 2018.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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(https://trustee.ietf.org/license-info) in effect on the date of
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Notational Conventions . . . . . . . . . . . . . . . . . 3
2. Design of the QUIC Transmission Machinery . . . . . . . . . . 3
2.1. Relevant Differences Between QUIC and TCP . . . . . . . . 4
2.1.1. Monotonically Increasing Packet Numbers . . . . . . . 4
2.1.2. No Reneging . . . . . . . . . . . . . . . . . . . . . 5
2.1.3. More ACK Ranges . . . . . . . . . . . . . . . . . . . 5
2.1.4. Explicit Correction For Delayed Acks . . . . . . . . 5
3. Loss Detection . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Computing the RTT estimate . . . . . . . . . . . . . . . 5
3.2. Ack-based Detection . . . . . . . . . . . . . . . . . . . 5
3.2.1. Fast Retransmit . . . . . . . . . . . . . . . . . . . 6
3.2.2. Early Retransmit . . . . . . . . . . . . . . . . . . 6
3.3. Timer-based Detection . . . . . . . . . . . . . . . . . . 7
3.3.1. Tail Loss Probe . . . . . . . . . . . . . . . . . . . 7
3.3.2. Retransmission Timeout . . . . . . . . . . . . . . . 9
3.3.3. Handshake Timeout . . . . . . . . . . . . . . . . . . 10
3.4. Algorithm Details . . . . . . . . . . . . . . . . . . . . 10
3.4.1. Constants of interest . . . . . . . . . . . . . . . . 10
3.4.2. Variables of interest . . . . . . . . . . . . . . . . 11
3.4.3. Initialization . . . . . . . . . . . . . . . . . . . 12
3.4.4. On Sending a Packet . . . . . . . . . . . . . . . . . 13
3.4.5. On Ack Receipt . . . . . . . . . . . . . . . . . . . 13
3.4.6. On Packet Acknowledgment . . . . . . . . . . . . . . 14
3.4.7. Setting the Loss Detection Alarm . . . . . . . . . . 15
3.4.8. On Alarm Firing . . . . . . . . . . . . . . . . . . . 17
3.4.9. Detecting Lost Packets . . . . . . . . . . . . . . . 17
3.5. Discussion . . . . . . . . . . . . . . . . . . . . . . . 18
4. Congestion Control . . . . . . . . . . . . . . . . . . . . . 19
4.1. Slow Start . . . . . . . . . . . . . . . . . . . . . . . 19
4.2. Congestion Avoidance . . . . . . . . . . . . . . . . . . 19
4.3. Recovery Period . . . . . . . . . . . . . . . . . . . . . 19
4.4. Tail Loss Probe . . . . . . . . . . . . . . . . . . . . . 19
4.5. Retransmission Timeout . . . . . . . . . . . . . . . . . 20
4.6. Pacing Rate . . . . . . . . . . . . . . . . . . . . . . . 20
4.7. Pseudocode . . . . . . . . . . . . . . . . . . . . . . . 20
4.7.1. Constants of interest . . . . . . . . . . . . . . . . 20
4.7.2. Variables of interest . . . . . . . . . . . . . . . . 20
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4.7.3. Initialization . . . . . . . . . . . . . . . . . . . 21
4.7.4. On Packet Sent . . . . . . . . . . . . . . . . . . . 21
4.7.5. On Packet Acknowledgement . . . . . . . . . . . . . . 21
4.7.6. On Packets Lost . . . . . . . . . . . . . . . . . . . 22
4.7.7. On Retransmission Timeout Verified . . . . . . . . . 22
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 23
6.1. Normative References . . . . . . . . . . . . . . . . . . 23
6.2. Informative References . . . . . . . . . . . . . . . . . 24
6.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 24
Appendix A. Acknowledgments . . . . . . . . . . . . . . . . . . 24
Appendix B. Change Log . . . . . . . . . . . . . . . . . . . . . 24
B.1. Since draft-ietf-quic-recovery-06 . . . . . . . . . . . . 24
B.2. Since draft-ietf-quic-recovery-05 . . . . . . . . . . . . 24
B.3. Since draft-ietf-quic-recovery-04 . . . . . . . . . . . . 25
B.4. Since draft-ietf-quic-recovery-03 . . . . . . . . . . . . 25
B.5. Since draft-ietf-quic-recovery-02 . . . . . . . . . . . . 25
B.6. Since draft-ietf-quic-recovery-01 . . . . . . . . . . . . 25
B.7. Since draft-ietf-quic-recovery-00 . . . . . . . . . . . . 25
B.8. Since draft-iyengar-quic-loss-recovery-01 . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 25
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.
1.1. Notational Conventions
The words "MUST", "MUST NOT", "SHOULD", and "MAY" are used in this
document. It's not shouting; when they are capitalized, they have
the special meaning defined in [RFC2119].
2. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
includes a packet sequence number (referred to below as a packet
number). These packet numbers never repeat in the lifetime of a
connection, and are monotonically increasing, which makes duplicate
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detection trivial. This fundamental design decision obviates the
need for disambiguating between transmissions and retransmissions and
eliminates significant complexity from QUIC's interpretation of TCP
loss detection mechanisms.
Every packet may contain several frames. We outline the frames that
are important to the loss detection and congestion control machinery
below.
o Retransmittable frames are frames requiring reliable delivery.
The most common are STREAM frames, which typically contain
application data.
o Crypto handshake data is sent on stream 0, and uses the
reliability machinery of QUIC underneath.
o ACK frames contain acknowledgment information. QUIC uses a SACK-
based scheme, where acks express up to 256 ranges.
2.1. 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.
2.1.1. 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".
QUIC separates the two: QUIC uses a packet sequence number (referred
to as the "packet number") for transmissions, and any data that is to
be delivered to the receiving application(s) is sent in one or more
streams, with stream offsets encoded within STREAM frames inside of
packets that determine delivery order.
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
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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.
2.1.2. No Reneging
QUIC ACKs contain information that is equivalent 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.
2.1.3. More ACK Ranges
QUIC supports up to 256 ACK ranges, opposed to TCP's 3 SACK ranges.
In high loss environments, this speeds recovery.
2.1.4. 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.
3. 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.
3.1. Computing the RTT estimate
(To be filled)
3.2. 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.
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(TODO: Define unacknowledged packet, ackable packet, outstanding
bytes.)
3.2.1. 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) after the unacknowledged packet. Receipt of
the ack indicates that a later packet was received, while
kReorderingThreshold provides some tolerance for reordering of
packets in the network.
The RECOMMENDED initial value for kReorderingThreshold is 3.
We derive this default from recommendations for TCP loss recovery
[RFC5681] [RFC6675]. It is possible for networks to exhibit higher
degrees of reordering, causing a sender to detect spurious losses.
Detecting spurious losses leads to unnecessary retransmissions and
may result in degraded performance due to the actions of the
congestion controller upon detecting loss. Implementers MAY use
algorithms developed for TCP, such as TCP-NCR [RFC4653], to improve
QUIC's reordering resilience, though care should be taken to map TCP
specifics to QUIC correctly. Similarly, using time-based loss
detection to deal with reordering, such as in PR-TCP, should be more
readily usable in QUIC. Making QUIC deal with such networks is
important open research, and implementers are encouraged to explore
this space.
3.2.2. Early Retransmit
Unacknowledged packets close to the tail may have fewer than
kReorderingThreshold number of ackable 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 ackable packet triggers the
Early Retransmit process, as follows.
If there are unacknowledged ackable packets still pending, they ought
to be marked as lost. To compensate for the reduced reordering
resilience, the sender SHOULD set an alarm for a small period of
time. If the unacknowledged ackable packets are not acknowledged
during this time, then these packets MUST be marked as lost.
An endpoint SHOULD set the alarm such that a packet is marked as lost
no earlier than 1.25 * max(SRTT, latest_RTT) since when it was sent.
Using max(SRTT, latest_RTT) protects from the two following cases:
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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 1.25 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 multipler increases loss recovery
delay.
This mechanism is based on Early Retransmit for TCP [RFC5827].
However, [RFC5827] does not include the alarm described above. Early
Retransmit is prone to spurious retransmissions due to its reduced
reordering resilence without the alarm. This observation led Linux
TCP implementers to implement an alarm for TCP as well, and this
document incorporates this advancement.
3.3. Timer-based Detection
Timer-based loss detection implements the spirit of TCP's Tail Loss
Probe and Retransmission Timeout mechanisms.
3.3.1. 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 an alarm when the last ackable packet before
quiescence is transmitted. When this alarm fires, a Tail Loss Probe
(TLP) packet is sent to evoke an acknowledgement from the receiver.
The alarm duration, or Probe Timeout (PTO), is set based on the
following conditions:
o If there is exactly one unacknowledged packet, PTO SHOULD be
scheduled for max(2_SRTT, 1.5_SRTT+kDelayedAckTimeout)
o If there are more than one unacknowledged packets, PTO SHOULD be
scheduled for max(2*SRTT, 10ms).
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o If RTO is earlier, schedule a TLP alarm in its place. That is,
PTO SHOULD be scheduled for min(RTO, PTO).
kDelayedAckTimeout is the expected delayed ACK timer. When there is
exactly one unacknowledged packet, the alarm duration includes time
for an acknowledgment to be received, and additionally, a
kDelayedAckTimeout period to compensate for the delayed
acknowledgment timer at the receiver.
The RECOMMENDED value for kDelayedAckTimeout is 25ms.
(TODO: Add negotiability of delayed ack timeout.)
A PTO value of at least 2_SRTT ensures that the ACK is overdue.
Using a PTO of exactly 1_SRTT may generate spurious probes, and
2*SRTT is simply the next integral value of RTT.
(TODO: These values of 2 and 1.5 are a bit arbitrary. Reconsider
these.)
If the Retransmission Timeout (RTO, Section 3.3.2) period is smaller
than the computed PTO, then a PTO is scheduled for the smaller RTO
period.
To reduce latency, it is RECOMMENDED that the sender set and allow
the TLP alarm to fire twice before setting an RTO alarm. In other
words, when the TLP alarm fires the first time, a TLP packet is sent,
and it is RECOMMENDED that the TLP alarm be scheduled for a second
time. When the TLP alarm fires the second time, a second TLP packet
is sent, and an RTO alarm SHOULD be scheduled Section 3.3.2.
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 alarm 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 alarm fires.
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.
A sender will commonly not know that a packet being sent is a tail
packet. Consequently, a sender may have to arm or adjust the TLP
alarm on every sent ackable packet.
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3.3.2. Retransmission Timeout
A Retransmission Timeout (RTO) alarm 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, an alarm is scheduled for the RTO
period. When this alarm fires, the sender sends two packets, to
evoke acknowledgements from the receiver, and restarts the RTO alarm.
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, minRTO)
o When an RTO alarm fires, the RTO period is doubled.
The sender typically has incurred a high latency penalty by the time
an RTO alarm fires, 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
alarm is not considered a strong enough signal of packet loss. An
RTO alarm fires only when there's a prolonged period of network
silence, which could be caused by a change in the underlying network
RTT.
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.
A packet sent when an RTO alarm fires 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 alarm 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.
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3.3.3. Handshake Timeout
Handshake packets, which contain STREAM frames for stream 0, are
critical to QUIC transport and crypto negotiation, so a separate
alarm is used for them.
The handshake timeout SHOULD be set to twice the initial RTT.
There are no prior RTT samples within this connection. However, this
may be a resumed connection over the same network, in which case, a
client 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 the first handshake packet is sent, the sender SHOULD set an
alarm for the handshake timeout period.
When the alarm fires, the sender MUST retransmit all unacknowledged
handshake frames. The sender SHOULD double the handshake timeout and
set an alarm for this period.
On each consecutive firing of the handshake alarm, the sender SHOULD
double the handshake timeout period.
When an acknowledgement is received for a handshake packet, the new
RTT is computed and the alarm SHOULD be set for twice the newly
computed smoothed RTT.
Handshake frames may be cancelled by handshake state transitions. In
particular, all non-protected frames SHOULD no longer be transmitted
once packet protection is available.
(TODO: Work this section some more. Add text on client vs. server,
and on stateless retry.)
3.4. Algorithm Details
3.4.1. 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 (default 2): Maximum number of tail loss probes before an
RTO fires.
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kReorderingThreshold (default 3): Maximum reordering in packet
number space before FACK style loss detection considers a packet
lost.
kTimeReorderingFraction (default 1/8): Maximum reordering in time
space before time based loss detection considers a packet lost.
In fraction of an RTT.
kMinTLPTimeout (default 10ms): Minimum time in the future a tail
loss probe alarm may be set for.
kMinRTOTimeout (default 200ms): Minimum time in the future an RTO
alarm may be set for.
kDelayedAckTimeout (default 25ms): The length of the peer's delayed
ack timer.
kDefaultInitialRtt (default 100ms): The default RTT used before an
RTT sample is taken.
3.4.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
loss_detection_alarm: Multi-modal alarm used for loss detection.
handshake_count: The number of times the handshake packets have 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_packet: The time the most recent packet was sent.
largest_sent_packet: The packet number of the most recently sent
packet.
largest_acked_packet: The largest packet number acknowledged in an
ack frame.
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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]
reordering_threshold: The largest delta between the largest acked
retransmittable packet and a packet containing retransmittable
frames before it's 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, 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.
3.4.3. Initialization
At the beginning of the connection, initialize the loss detection
variables as follows:
loss_detection_alarm.reset()
handshake_count = 0
tlp_count = 0
rto_count = 0
if (UsingTimeLossDetection())
reordering_threshold = infinite
time_reordering_fraction = kTimeReorderingFraction
else:
reordering_threshold = kReorderingThreshold
time_reordering_fraction = infinite
loss_time = 0
smoothed_rtt = 0
rttvar = 0
largest_sent_before_rto = 0
time_of_last_sent_packet = 0
largest_sent_packet = 0
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3.4.4. 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 is_ack_only: A boolean that indicates whether a packet only
contains an ACK frame. If true, it is still expected an ack will
be received for this packet, but it is not congestion controlled.
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:
OnPacketSent(packet_number, is_ack_only, sent_bytes):
time_of_last_sent_packet = now
largest_sent_packet = packet_number
sent_packets[packet_number].packet_number = packet_number
sent_packets[packet_number].time = now
if !is_ack_only:
OnPacketSentCC(sent_bytes)
sent_packets[packet_number].bytes = sent_bytes
SetLossDetectionAlarm()
3.4.5. On Ack Receipt
When an ack is received, it may acknowledge 0 or more packets.
Pseudocode for OnAckReceived and UpdateRtt follow:
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OnAckReceived(ack):
largest_acked_packet = ack.largest_acked
// If the largest acked is newly acked, update the RTT.
if (sent_packets[ack.largest_acked]):
latest_rtt = now - sent_packets[ack.largest_acked].time
if (latest_rtt > ack.ack_delay):
latest_rtt -= ack.delay
UpdateRtt(latest_rtt)
// Find all newly acked packets.
for acked_packet in DetermineNewlyAckedPackets():
OnPacketAcked(acked_packet.packet_number)
DetectLostPackets(ack.largest_acked_packet)
SetLossDetectionAlarm()
UpdateRtt(latest_rtt):
// Based on {{RFC6298}}.
if (smoothed_rtt == 0):
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
else:
rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - latest_rtt)
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt
3.4.6. 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 packet
number of the newly acked packet, and returns a list of packet
numbers that are detected as lost.
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:
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OnPacketAcked(acked_packet_number):
OnPacketAckedCC(acked_packet_number)
// If a packet sent prior to RTO was acked, then the RTO
// was spurious. Otherwise, inform congestion control.
if (rto_count > 0 &&
acked_packet_number > largest_sent_before_rto)
OnRetransmissionTimeoutVerified()
handshake_count = 0
tlp_count = 0
rto_count = 0
sent_packets.remove(acked_packet_number)
3.4.7. Setting the Loss Detection Alarm
QUIC loss detection uses a single alarm for all timer-based loss
detection. The duration of the alarm is based on the alarm's mode,
which is set in the packet and timer events further below. The
function SetLossDetectionAlarm defined below shows how the single
timer is set based on the alarm mode.
3.4.7.1. Handshake Packets
The initial flight has no prior RTT sample. A client SHOULD remember
the previous RTT it observed when resumption is attempted and use
that for an initial RTT value. If no previous RTT is available, the
initial RTT defaults to 100ms.
Endpoints MUST retransmit handshake frames if not acknowledged within
a time limit. This time limit will start as the largest of twice the
RTT value and MinTLPTimeout. Each consecutive handshake
retransmission doubles the time limit, until an acknowledgement is
received.
Handshake frames may be cancelled by handshake state transitions. In
particular, all non-protected frames SHOULD be no longer be
transmitted once packet protection is available.
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 0RTT packets.
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3.4.7.2. Tail Loss Probe and Retransmission Timeout
Tail loss probes [LOSS-PROBE] and retransmission timeouts [RFC6298]
are an alarm based mechanism to recover from cases when there are
outstanding retransmittable packets, but an acknowledgement has not
been received in a timely manner.
3.4.7.3. Early Retransmit
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.
3.4.7.4. Pseudocode
Pseudocode for SetLossDetectionAlarm follows:
SetLossDetectionAlarm():
if (retransmittable packets are not outstanding):
loss_detection_alarm.cancel()
return
if (handshake packets are outstanding):
// Handshake retransmission alarm.
if (smoothed_rtt == 0):
alarm_duration = 2 * kDefaultInitialRtt
else:
alarm_duration = 2 * smoothed_rtt
alarm_duration = max(alarm_duration, kMinTLPTimeout)
alarm_duration = alarm_duration * (2 ^ handshake_count)
else if (loss_time != 0):
// Early retransmit timer or time loss detection.
alarm_duration = loss_time - now
else if (tlp_count < kMaxTLPs):
// Tail Loss Probe
if (retransmittable_packets_outstanding == 1):
alarm_duration = 1.5 * smoothed_rtt + kDelayedAckTimeout
else:
alarm_duration = kMinTLPTimeout
alarm_duration = max(alarm_duration, 2 * smoothed_rtt)
else:
// RTO alarm
alarm_duration = smoothed_rtt + 4 * rttvar
alarm_duration = max(alarm_duration, kMinRTOTimeout)
alarm_duration = alarm_duration * (2 ^ rto_count)
loss_detection_alarm.set(now + alarm_duration)
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3.4.8. On Alarm Firing
QUIC uses one loss recovery alarm, which when set, can be in one of
several modes. When the alarm fires, the mode determines the action
to be performed.
Pseudocode for OnLossDetectionAlarm follows:
OnLossDetectionAlarm():
if (handshake packets are outstanding):
// Handshake retransmission alarm.
RetransmitAllHandshakePackets()
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++
SetLossDetectionAlarm()
3.4.9. Detecting Lost Packets
Packets in QUIC are only considered lost once a larger packet number
is acknowledged. DetectLostPackets is called every time an ack is
received. If the loss detection alarm fires and the loss_time is
set, the previous largest acked packet is supplied.
3.4.9.1. Handshake Packets
The receiver MUST close the connection with an error of type
OPTIMISTIC_ACK when receiving an unprotected packet that acks
protected packets. The receiver MUST trust protected acks for
unprotected packets, however. Aside from this, loss detection for
handshake packets when an ack is processed is identical to other
packets.
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3.4.9.2. Pseudocode
DetectLostPackets takes one parameter, acked, which is the largest
acked packet.
Pseudocode for DetectLostPackets follows:
DetectLostPackets(largest_acked):
loss_time = 0
lost_packets = {}
delay_until_lost = infinite
if (time_reordering_fraction != infinite):
delay_until_lost =
(1 + time_reordering_fraction) * max(latest_rtt, smoothed_rtt)
else if (largest_acked.packet_number == largest_sent_packet):
// Early retransmit alarm.
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):
lost_packets.insert(unacked)
else if (delta > reordering_threshold)
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)
foreach (packet in lost_packets)
sent_packets.remove(packet.packet_number)
3.5. 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.
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4. Congestion Control
QUIC's congestion control is based on TCP NewReno[RFC6582] congestion
control to determine the congestion window and pacing rate. QUIC
congestion control is specified in bytes due to finer control and the
ease of appropriate byte counting[RFC3465].
4.1. Slow Start
QUIC begins every connection in slow start and exits slow start upon
loss. QUIC re-enters slow start after a retransmission timeout.
While in slow start, QUIC increases the congestion window by the
number of acknowledged bytes when each ack is processed.
4.2. 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 MSS of bytes 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.
4.3. Recovery Period
Recovery is a period of time beginning with detection of a lost
packet. 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 ending when the lost packet that
started recovery is acknowledged.
During recovery, the congestion window is not increased or decreased.
As such, multiple lost packets only decrease the congestion window
once as long as they're lost before exiting recovery. This causes
QUIC to decrease the congestion window multiple times if
retransmisions are lost, but limits the reduction to once per round
trip.
4.4. Tail Loss Probe
If recovery sends a tail loss probe, no change is made to the
congestion window or pacing rate. Acknowledgement or loss of tail
loss probes are treated like any other packet.
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4.5. Retransmission Timeout
When retransmissions are sent due to a retransmission timeout alarm,
no change is made to the congestion window or pacing rate until the
next acknowledgement arrives. When an ack arrives, if packets prior
to the first retransmission timeout are acknowledged, then the
congestion window remains the same. If no packets prior to the first
retransmission timeout are acknowledged, the retransmission timeout
has been validated and the congestion window must be reduced to the
minimum congestion window and slow start is begun.
4.6. Pacing Rate
The pacing rate is a function of the mode, the congestion window, and
the smoothed rtt. Specifically, the pacing rate is 2 times the
congestion window divided by the smoothed RTT during slow start and
1.25 times the congestion window divided by the smoothed RTT during
congestion avoidance. In order to fairly compete with flows that are
not pacing, it is recommended to not pace the first 10 sent packets
when exiting quiescence.
4.7. Pseudocode
4.7.1. 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.
kDefaultMss (default 1460 bytes): The default max packet size used
for calculating default and minimum congestion windows.
kInitialWindow (default 10 * kDefaultMss): Default limit on the
amount of outstanding data in bytes.
kMinimumWindow (default 2 * kDefaultMss): Default minimum congestion
window.
kLossReductionFactor (default 0.5): Reduction in congestion window
when a new loss event is detected.
4.7.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one retransmittable or PADDING frame, and
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have not been acked or declared lost. The size does not include
IP or UDP overhead. Packets only containing ack frames do not
count towards byte_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.
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.
4.7.3. 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
4.7.4. 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
4.7.5. On Packet Acknowledgement
Invoked from loss detection's OnPacketAcked and is supplied with
acked_packet from sent_packets.
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OnPacketAckedCC(acked_packet):
// Remove from bytes_in_flight.
bytes_in_flight -= acked_packet.bytes
if (acked_packet.packet_number < end_of_recovery):
// Do not increase congestion window in recovery period.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packets.bytes
else:
// Congestion avoidance.
congestion_window +=
kDefaultMss * acked_packets.bytes / congestion_window
4.7.6. 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 recovery epoch if the lost packet is larger
// than the end of the previous recovery epoch.
if (end_of_recovery < largest_lost_packet.packet_number):
end_of_recovery = largest_sent_packet
congestion_window *= kLossReductionFactor
congestion_window = max(congestion_window, kMinimumWindow)
ssthresh = congestion_window
4.7.7. On Retransmission Timeout Verified
QUIC decreases the congestion window to the minimum value once the
retransmission timeout has been verified.
OnRetransmissionTimeoutVerified()
congestion_window = kMinimumWindow
5. IANA Considerations
This document has no IANA actions. Yet.
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6. References
6.1. Normative References
[QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", draft-ietf-quic-
transport-07 (work in progress), November 2017.
[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>.
[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>.
[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>.
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6.2. Informative References
[LOSS-PROBE]
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.
[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>.
[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>.
[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.
6.3. 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. Acknowledgments
Appendix B. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
B.1. Since draft-ietf-quic-recovery-06
Nothing yet.
B.2. Since draft-ietf-quic-recovery-05
o Add more congestion control text (#776)
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B.3. Since draft-ietf-quic-recovery-04
No significant changes.
B.4. Since draft-ietf-quic-recovery-03
No significant changes.
B.5. 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)
B.6. 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
B.7. Since draft-ietf-quic-recovery-00
o Improved description of constants and ACK behavior
B.8. 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
Authors' Addresses
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Jana Iyengar (editor)
Google
Email: jri@google.com
Ian Swett (editor)
Google
Email: ianswett@google.com
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