---
title: QUIC Loss Detection and Congestion Control
abbrev: QUIC Loss Detection
docname: draft-ietf-quic-recovery-latest
date: {DATE}
category: std
ipr: trust200902
area: Transport
workgroup: QUIC

stand_alone: yes
pi: [toc, sortrefs, symrefs, docmapping]

author:
 -
    ins: J. Iyengar
    name: Jana Iyengar
    org: Google
    email: jri@google.com
    role: editor
 -
    ins: I. Swett
    name: Ian Swett
    org: Google
    email: ianswett@google.com
    role: editor

normative:

  QUIC-TRANSPORT:
    title: "QUIC: A UDP-Based Multiplexed and Secure Transport"
    date: {DATE}
    seriesinfo:
      Internet-Draft: draft-ietf-quic-transport-latest
    author:
      -
        ins: J. Iyengar
        name: Jana Iyengar
        org: Google
        role: editor
      -
        ins: M. Thomson
        name: Martin Thomson
        org: Mozilla
        role: editor

--- abstract

This document describes loss detection and congestion control mechanisms for
QUIC.

--- note_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>.

Working Group information can be found at <https://github.com/quicwg>; source
code and issues list for this draft can be found at
<https://github.com/quicwg/base-drafts/labels/recovery>.

--- middle

# 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.


## 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}}.


# 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 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.

* Retransmittable frames are frames requiring reliable delivery.  The most
  common are STREAM frames, which typically contain application data.

* Crypto handshake data is sent on stream 0, and uses the reliability
  machinery of QUIC underneath.

* ACK frames contain acknowledgment information.  QUIC uses a SACK-based
  scheme, where acks express up to 256 ranges.  The ACK frame also includes a
  receive timestamp for each packet newly acked.

## 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.

### 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 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.

### 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.

### More ACK Ranges

QUIC supports up to 256 ACK ranges, opposed to TCP's 3 SACK ranges.  In high
loss environments, this speeds recovery.

### 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.


# Loss Detection

## Overview {#overview}

QUIC uses a combination of ack information and alarms to detect lost packets.
An unacknowledged QUIC packet is marked as lost in one of the following ways:

  * A packet is marked as lost if at least one packet that was sent a threshold
    number of packets (kReorderingThreshold) after it has been
    acknowledged. This indicates that the unacknowledged packet is either lost
    or reordered beyond the specified threshold. This mechanism combines both
    TCP's FastRetransmit and FACK mechanisms.

  * If a packet is near the tail, where fewer than kReorderingThreshold packets
    are sent after it, the sender cannot expect to detect loss based on the
    previous mechanism. In this case, a sender uses both ack information and an
    alarm to detect loss. Specifically, when the last sent packet is
    acknowledged, the sender waits a short period of time to allow for
    reordering and then marks any unacknowledged packets as lost. This mechanism
    is based on the Linux implementation of TCP Early Retransmit.

  * If a packet is sent at the tail, there are no packets sent after it, and the
    sender cannot use ack information to detect its loss. The sender therefore
    relies on an alarm to detect such tail losses. This mechanism is based on
    TCP's Tail Loss Probe.

  * If all else fails, a Retransmission Timeout (RTO) alarm is always set when
    any retransmittable packet is outstanding. When this alarm fires, all
    unacknowledged packets are marked as lost.

  * Instead of a packet threshold to tolerate reordering, a QUIC sender may use
    a time threshold. This allows for senders to be tolerant of short periods of
    significant reordering. In this mechanism, a QUIC sender marks a packet as
    lost when a packet larger than it is acknowledged and a threshold amount of
    time has passed since the packet was sent.

  * Handshake packets, which contain STREAM frames for stream 0, are
    critical to QUIC transport and crypto negotiation, so a separate alarm
    period is used for them.


## Algorithm Details

### 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.

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.

### 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.

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.

### 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
~~~

### 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:

* packet_number: The packet number of the sent packet.

* is_retransmittable: A boolean that indicates whether the packet contains at
  least one frame requiring reliable deliver.  The retransmittability of various
  QUIC frames is described in {{QUIC-TRANSPORT}}.  If false, it is still
  acceptable for an ack to be received for this packet.  However, a caller MUST
  NOT set is_retransmittable to true if an ack is not expected.

* sent_bytes: The number of bytes sent in the packet.

Pseudocode for OnPacketSent follows:

~~~
 OnPacketSent(packet_number, is_retransmittable, 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_retransmittable:
     sent_packets[packet_number].bytes = sent_bytes
     SetLossDetectionAlarm()
~~~

### On Ack Receipt

When an ack is received, it may acknowledge 0 or more packets.

Pseudocode for OnAckReceived and UpdateRtt follow:

~~~
   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 * (smoothed_rtt - latest_rtt)
       smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt
~~~

### 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:

~~~
   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)
~~~

### 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.

#### 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.

#### Tail Loss Probe and Retransmission Timeout

Tail loss probes {{?LOSS-PROBE=I-D.dukkipati-tcpm-tcp-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.

#### 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.

#### 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)
~~~

### 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()
~~~

### 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.

#### Handshake Packets

The receiver MUST ignore unprotected packets that ack 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.

#### 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
       packet_delta = largest_acked.packet_number - unacked.packet_number
       if (time_since_sent > delay_until_lost):
         lost_packets.insert(unacked)
       else if (packet_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)
~~~

## 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.


# Congestion Control

QUIC's congestion control is based on TCP NewReno{{?RFC6582}}
congestion control to determine the congestion window and pacing rate.

## Slow Start

QUIC begins every connection in slow start and exits slow start upon
loss. While in slow start, QUIC increases the congestion window by the
number of acknowledged bytes when each ack is processed.

## Recovery

Recovery is a period of time beginning with detection of a lost packet.
It ends when all packets outstanding at the time recovery began have been
acknowledged or lost. During recovery, the congestion window is not
increased or decreased.

## 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.


## 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 have not been acked or
  declared lost. The size does not include IP or UDP overhead.
  Ack only frames do not count towards byte_in_flight.

congestion_window:
: Maximum number of bytes in flight that may be sent.

end_of_recovery:
: The packet number after which QUIC will no longer be in recovery.

ssthresh
: Slow start threshold in bytes.  When the congestion window is below
  ssthresh, it grows by the number of bytes acknowledged for each ack.

## Initialization

At the beginning of the connection, initialize the loss detection variables as
follows:

~~~
   congestion_window = kInitialWindow
   bytes_in_flight = 0
   end_of_recovery = 0
   ssthresh = infinite
~~~

## On Packet Acknowledgement

Invoked from loss detection's OnPacketAcked and is supplied with
acked_packet from sent_packets.

Pseudocode for OnPacketAckedCC follows:

~~~
   OnPacketAckedCC(acked_packet):
     if (acked_packet.packet_number < end_of_recovery):
       return
     if (congestion_window < ssthresh):
       congestion_window += acket_packets.bytes
     else:
       congestion_window +=
           acked_packets.bytes / congestion_window
~~~

## On Packets Lost

Invoked by loss detection from DetectLostPackets when new packets
are detected lost.

~~~
   OnPacketsLost(lost_packets):
     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
~~~

## On Retransmission Timeout Verified

QUIC decreases the congestion window to the minimum value once the
retransmission timeout has been confirmed to not be spurious when
the first post-RTO acknowledgement is processed.

~~~
   OnRetransmissionTimeoutVerified()
     congestion_window = kMinimumWindow
~~~

## Pacing Packets

QUIC sends a packet if there is available congestion window and
sending the packet does not exceed the pacing rate.

TimeToSend returns infinite if the congestion controller is
congestion window limited, a time in the past if the packet can be
sent immediately, and a time in the future if sending is pacing
limited.

~~~
   TimeToSend(packet_size):
     if (bytes_in_flight + packet_size > congestion_window)
       return infinite
     return time_of_last_sent_packet +
         (packet_size * smoothed_rtt) / congestion_window
~~~


# IANA Considerations

This document has no IANA actions.  Yet.


--- back

# Acknowledgments


# Change Log

> **RFC Editor's Note:**  Please remove this section prior to
> publication of a final version of this document.

## Since draft-ietf-quic-recovery-04

No significant changes.

## Since draft-ietf-quic-recovery-03

No significant changes.

## Since draft-ietf-quic-recovery-02

- Integrate F-RTO (#544, #409)
- Add congestion control (#545, #395)
- Require connection abort if a skipped packet was acknowledged (#415)
- Simplify RTO calculations (#142, #417)


## Since draft-ietf-quic-recovery-01

- Overview added to loss detection
- Changes initial default RTT to 100ms
- Added time-based loss detection and fixes early retransmit
- Clarified loss recovery for handshake packets
- Fixed references and made TCP references informative


## Since draft-ietf-quic-recovery-00

- Improved description of constants and ACK behavior


## Since draft-iyengar-quic-loss-recovery-01

- Adopted as base for draft-ietf-quic-recovery
- Updated authors/editors list
- Added table of contents