---
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: Fastly
    email: jri.ietf@gmail.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: Fastly
        role: editor
      -
        ins: M. Thomson
        name: Martin Thomson
        org: Mozilla
        role: editor

informative:

  FACK:
    title: "Forward Acknowledgement: Refining TCP Congestion Control"
    author:
      - ins: M. Mathis
      - ins: J. Mahdavi
    date: 1996-08
    seriesinfo: ACM SIGCOMM

--- 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.


# Conventions and Definitions

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14 {{!RFC2119}} {{!RFC8174}}
when, and only when, they appear in all capitals, as shown here.

Definitions of terms that are used in this document:

ACK-only:

: Any packet containing only one or more ACK frame(s).

In-flight:

: Packets are considered in-flight when they have been sent
  and neither acknowledged nor declared lost, and they are not
  ACK-only.

Ack-eliciting Frames:

: All frames besides ACK or PADDING are considered ack-eliciting.

Ack-eliciting Packets:

: Packets that contain ack-eliciting frames elicit an ACK from the receiver
  within the maximum ack delay and are called ack-eliciting packets.

Crypto Packets:

: Packets containing CRYPTO data sent in Initial or Handshake
  packets.

# Design of the QUIC Transmission Machinery

All transmissions in QUIC are sent with a packet-level header, which indicates
the encryption level and includes a packet sequence number (referred to below as
a packet number).  The encryption level indicates the packet number space, as
described in {{QUIC-TRANSPORT}}.  Packet numbers never repeat within a packet
number space for the lifetime of a connection.  Packet numbers monotonically
increase within a space, preventing ambiguity.

This design obviates the need for disambiguating between transmissions and
retransmissions and eliminates significant complexity from QUIC's interpretation
of TCP loss detection mechanisms.

QUIC packets can contain multiple frames of different types. The recovery
mechanisms ensure that data and frames that need reliable delivery are
acknowledged or declared lost and sent in new packets as necessary. The types
of frames contained in a packet affect recovery and congestion control logic:

* All packets are acknowledged, though packets that contain no
  ack-eliciting frames are only acknowledged along with ack-eliciting
  packets.

* Long header packets that contain CRYPTO frames are critical to the
  performance of the QUIC handshake and use shorter timers for
  acknowledgement and retransmission.

* 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.

## 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.

### 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.

### Monotonically Increasing Packet Numbers

TCP conflates transmission order at the sender with delivery order at the
receiver, which results in retransmissions of the same data carrying the same
sequence number, and consequently leads to "retransmission ambiguity".  QUIC
separates the two: QUIC uses a packet number to indicate transmission order,
and any application data is sent in one or more streams, with delivery order
determined by stream offsets encoded within STREAM frames.

QUIC's packet number is strictly increasing within a packet number space,
and directly encodes transmission order.  A higher packet number signifies
that the packet was sent later, and a lower packet number signifies that
the packet was sent earlier.  When a packet containing ack-eliciting
frames is detected lost, QUIC rebundles necessary frames in a new packet
with a new packet number, removing ambiguity about which packet is
acknowledged when an ACK is received.  Consequently, more accurate RTT
measurements can be made, spurious retransmissions are trivially detected, and
mechanisms such as Fast Retransmit can be applied universally, based only on
packet number.

This design point significantly simplifies loss detection mechanisms for QUIC.
Most TCP mechanisms implicitly attempt to infer transmission ordering based on
TCP sequence numbers - a non-trivial task, especially when TCP timestamps are
not available.

### 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.

### 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.

### 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.


# Generating Acknowledgements

QUIC SHOULD delay sending acknowledgements in response to packets, but MUST NOT
excessively delay acknowledgements of ack-eliciting packets. 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.

## Crypto Handshake Data

In order to quickly complete the handshake and avoid spurious retransmissions
due to crypto retransmission timeouts, crypto packets SHOULD use a very short
ack delay, such as 1ms.  ACK frames MAY be sent immediately when the crypto
stack indicates all data for that packet number space has been received.

## 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.

## Receiver Tracking of ACK Frames

When a packet containing an ACK frame is sent, the largest acknowledged in that
frame may be saved.  When a packet containing an ACK frame is acknowledged, the
receiver can stop acknowledging packets less than or equal to the largest
acknowledged in the sent ACK frame.

In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT
of reordering. In cases with ACK frame loss and reordering, this approach does
not guarantee that every acknowledgement is seen by the sender before it is no
longer included in the ACK frame. Packets could be received out of order and
all subsequent ACK frames containing them could be lost. In this case, the
loss recovery algorithm may cause spurious retransmits, but the sender will
continue making forward progress.

# 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 acked packet was sent.  An RTT sample
MUST NOT be taken for a packet that is not newly acknowledged or not
ack-eliciting.

When RTT is calculated, the ack delay field from the ACK frame SHOULD be limited
to the max_ack_delay specified by the peer.  Limiting ack_delay to max_ack_delay
ensures a peer specifying an extremely small max_ack_delay doesn't cause more
spurious timeouts than a peer that correctly specifies max_ack_delay. It 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.

# 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.

If a packet is lost, the QUIC transport needs to recover from that loss, such
as by retransmitting the data, sending an updated frame, or abandoning the
frame.  For more information, see Section 13.2 of {{QUIC-TRANSPORT}}.

## Acknowledgement-based Detection {#ack-loss-detection}

Acknowledgement-based loss detection implements the spirit of TCP's Fast
Retransmit {{?RFC5681}}, Early Retransmit {{?RFC5827}}, FACK {{FACK}}, SACK loss
recovery {{?RFC6675}}, and RACK {{?RACK=I-D.ietf-tcpm-rack}}. This section
provides an overview of how these algorithms are implemented in QUIC.

A packet is declared lost if it meets all the following conditions:

* The packet is unacknowledged, in-flight, and was sent prior to an
  acknowledged packet.

* Either its packet number is kPacketThreshold smaller than an acknowledged
  packet ({{packet-threshold}}), or it was sent long enough in the past
  ({{time-threshold}}).

The acknowledgement indicates that a packet sent later was delivered, while the
packet and time thresholds provide some tolerance for packet reordering.

Spuriously declaring packets as lost leads to unnecessary retransmissions and
may result in degraded performance due to the actions of the congestion
controller upon detecting loss.  Implementations that detect spurious
retransmissions and increase the reordering threshold in packets or time MAY
choose to start with smaller initial reordering thresholds to minimize recovery
latency.

### Packet Threshold

The RECOMMENDED initial value for the packet reordering threshold
(kPacketThreshold) is 3, based on best practices for TCP loss detection
{{?RFC5681}} {{?RFC6675}}.

Some networks may exhibit higher degrees of reordering, causing a sender to
detect spurious losses.  Implementers MAY use algorithms developed for TCP, such
as TCP-NCR {{?RFC4653}}, to improve QUIC's reordering resilience.

### Time Threshold {#time-threshold}

Once a later packet has been acknowledged, an endpoint SHOULD declare an earlier
packet lost if it was sent a threshold amount of time in the past. The time
threshold is computed as kTimeThreshold * max(SRTT, latest_RTT).
If packets sent prior to the largest acknowledged packet cannot yet be declared
lost, then a timer SHOULD be set for the remaining time.

The RECOMMENDED time threshold (kTimeThreshold), expressed as a round-trip time
multiplier, is 9/8.

Using max(SRTT, latest_RTT) protects from the two following cases:

* the latest RTT sample is lower than the SRTT, perhaps due to reordering where
  packet whose ack triggered the Early Retransmit process encountered a shorter
  path;

* 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.

Implementers MAY experiment with using other reordering thresholds, including
absolute thresholds, bearing in mind that a lower multiplier reduces reordering
resilience and increases spurious retransmissions, and a higher multiplier
increases loss detection delay.


## Timeout Loss Detection

Timeout loss detection recovers from losses that cannot be handled by
acknowledgement-based loss detection.  It uses a single timer which switches
between a crypto retransmission timer and a probe timer.

### Crypto Retransmission Timeout

Data in CRYPTO frames is critical to QUIC transport and crypto negotiation, so a
more aggressive timeout is used to retransmit it.

The initial crypto retransmission timeout SHOULD be set to twice the initial
RTT.

At the beginning, there are no prior RTT samples within a connection.  Resumed
connections over the same network SHOULD use the previous connection's final
smoothed RTT value as the resumed connection's initial RTT.  If no previous RTT
is available, or if the network changes, the initial RTT SHOULD be set to 100ms.
When an acknowledgement is received, a new RTT is computed and the timer
SHOULD be set for twice the newly computed smoothed RTT.

When crypto packets are sent, the sender MUST set a timer for the crypto
timeout period.  Upon timeout, the sender MUST retransmit all unacknowledged
CRYPTO data if possible.

Until the server has validated the client's address on the path, the amount of
data it can send is limited, as specified in {{QUIC-TRANSPORT}}.  If not all
unacknowledged CRYPTO data can be sent, then all unacknowledged CRYPTO data sent
in Initial packets should be retransmitted.  If no data can be sent, then no
alarm should be armed until data has been received from the client.

Because the server could be blocked until more packets are received, the client
MUST start the crypto retransmission timer even if there is no unacknowledged
CRYPTO data.  If the timer expires and the client has no CRYPTO data to
retransmit and does not have Handshake keys, it SHOULD send an Initial packet in
a UDP datagram of at least 1200 bytes.  If the client has Handshake keys, it
SHOULD send a Handshake packet.

On each consecutive expiration of the crypto timer without receiving an
acknowledgement for a new packet, the sender SHOULD double the crypto
retransmission timeout and set a timer for this period.

When crypto packets are in flight, the probe timer ({{pto}}) is not active.


#### Retry and Version Negotiation

A Retry or Version Negotiation packet causes a client to send another Initial
packet, effectively restarting the connection process and resetting congestion
control and loss recovery state, including resetting any pending timers.  Either
packet indicates that the Initial was received but not processed.  Neither
packet can be treated as an acknowledgment for the Initial.


#### Discarding Initial State {#discard-initial}

As described in Section 17.5.1 of {{QUIC-TRANSPORT}}, endpoints stop sending and
receiving Initial packets once they start exchanging Handshake packets.  At this
point, all loss recovery state for the Initial packet number space is also
discarded. Packets that are in flight for the packet number space are not
declared as either acknowledged or lost.  After discarding state, new Initial
packets will not be sent.

The client MAY however compute an RTT estimate to the server as the time period
from when the first Initial was sent to when a Retry or a Version Negotiation
packet is received.  The client MAY use this value to seed the RTT estimator for
a subsequent connection attempt to the server.


### Probe Timeout {#pto}

A Probe Timeout (PTO) triggers a probe packet when ack-eliciting data is in
flight but an acknowledgement is not received within the expected period of
time.  A PTO enables a connection to recover from loss of tail packets or acks.
The PTO algorithm used in QUIC implements the reliability functions of Tail Loss
Probe {{?TLP=I-D.dukkipati-tcpm-tcp-loss-probe}} {{?RACK}}, RTO {{?RFC5681}} and
F-RTO algorithms for TCP {{?RFC5682}}, and the timeout computation is based on
TCP's retransmission timeout period {{?RFC6298}}.

#### Computing PTO

When an ack-eliciting packet is transmitted, the sender schedules a timer for
the PTO period as follows:

~~~
PTO = max(smoothed_rtt + 4*rttvar + max_ack_delay, kGranularity)
~~~

kGranularity, smoothed_rtt, rttvar, and max_ack_delay are defined in
{{ld-consts-of-interest}} and {{ld-vars-of-interest}}.

The PTO period is the amount of time that a sender ought to wait for an
acknowledgement of a sent packet.  This time period includes the estimated
network roundtrip-time (smoothed_rtt), the variance in the estimate (4*rttvar),
and max_ack_delay, to account for the maximum time by which a receiver might
delay sending an acknowledgement.

The PTO value MUST be set to at least kGranularity, to avoid the timer expiring
immediately.

When a PTO timer expires, the PTO period MUST be set to twice its current value.
This exponential reduction in the sender's rate is important because the PTOs
might be caused by loss of packets or acknowledgements due to severe congestion.

A sender computes its PTO timer every time an ack-eliciting packet is sent. A
sender might choose to optimize this by setting the timer fewer times if it
knows that more ack-eliciting packets will be sent within a short period of
time.

#### Sending Probe Packets

When a PTO timer expires, the sender MUST send one ack-eliciting packet as a
probe. A sender MAY send up to two ack-eliciting packets, to avoid an expensive
consecutive PTO expiration due to a single packet loss.

Consecutive PTO periods increase exponentially, and as a result, connection
recovery latency increases exponentially as packets continue to be dropped in
the network.  Sending two packets on PTO expiration increases resilience to
packet drops, thus reducing the probability of consecutive PTO events.

Probe packets sent on a PTO MUST be ack-eliciting.  A probe packet SHOULD carry
new data when possible.  A probe packet MAY carry retransmitted unacknowledged
data when new data is unavailable, when flow control does not permit new data to
be sent, or to opportunistically reduce loss recovery delay.  Implementations
MAY use alternate strategies for determining the content of probe packets,
including sending new or retransmitted data based on the application's
priorities.


#### Loss Detection {#pto-loss}

Delivery or loss of packets in flight is established when an ACK frame is
received that newly acknowledges one or more packets.

A PTO timer expiration event does not indicate packet loss and MUST NOT cause
prior unacknowledged packets to be marked as lost.  After a PTO timer has
expired, an endpoint uses the following rules to mark packets as lost when an
acknowledgement is received that newly acknowledges packets.

When an acknowledgement is received that newly acknowledges packets, loss
detection proceeds as dictated by packet and time threshold mechanisms, see
{{ack-loss-detection}}.


## Tracking Sent Packets {#tracking-sent-packets}

To correctly implement congestion control, a QUIC sender tracks every
ack-eliciting packet until the packet is acknowledged or lost.
It is expected that implementations will be able to access this information by
packet number and crypto context and store the per-packet fields
({{sent-packets-fields}}) for loss recovery and congestion control.

After a packet is declared lost, it SHOULD be tracked for an amount of time
comparable to the maximum expected packet reordering, such as 1 RTT.  This
allows for detection of spurious retransmissions.

Sent packets are tracked for each packet number space, and ACK
processing only applies to a single space.

### Sent Packet Fields {#sent-packets-fields}

packet_number:
: The packet number of the sent packet.

ack_eliciting:
: A boolean that indicates whether a packet is ack-eliciting.
  If true, it is expected that an acknowledgement will be received,
  though the peer could delay sending the ACK frame containing it
  by up to the MaxAckDelay.

in_flight:
: A boolean that indicates whether the packet counts towards bytes in
  flight.

is_crypto_packet:
: A boolean that indicates whether the packet contains
  cryptographic handshake messages critical to the completion of the QUIC
  handshake. In this version of QUIC, this includes any packet with the long
  header that includes a CRYPTO frame.

sent_bytes:
: The number of bytes sent in the packet, not including UDP or IP
  overhead, but including QUIC framing overhead.

time_sent:
: The time the packet was sent.


## Pseudocode

### Constants of interest {#ld-consts-of-interest}

Constants used in loss recovery are based on a combination of RFCs, papers, and
common practice.  Some may need to be changed or negotiated in order to better
suit a variety of environments.

kPacketThreshold:
: Maximum reordering in packets before packet threshold loss detection
  considers a packet lost. The RECOMMENDED value is 3.

kTimeThreshold:

: Maximum reordering in time before time threshold loss detection
  considers a packet lost. Specified as an RTT multiplier. The RECOMMENDED
  value is 9/8.

kGranularity:

: Timer granularity. This is a system-dependent value.  However, implementations
  SHOULD use a value no smaller than 1ms.

kInitialRtt:
: The RTT used before an RTT sample is taken. The RECOMMENDED value is 100ms.

### Variables of interest {#ld-vars-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.

crypto_count:
: The number of times all unacknowledged CRYPTO data has been
  retransmitted without receiving an ack.

pto_count:
: The number of times a PTO has been sent without receiving an ack.

time_of_last_sent_ack_eliciting_packet:
: The time the most recent ack-eliciting packet was sent.

time_of_last_sent_crypto_packet:
: The time the most recent crypto packet was sent.

largest_sent_packet:
: The packet number of the most recently sent packet.

largest_acked_packet:
: The largest packet number acknowledged in the packet number space so far.

latest_rtt:
: The most recent RTT measurement made when receiving an ack for
  a previously unacked packet.

smoothed_rtt:
: The smoothed RTT of the connection, computed as described in
  {{?RFC6298}}

rttvar:
: The RTT variance, computed as described in {{?RFC6298}}

min_rtt:
: The minimum RTT seen in the connection, ignoring ack delay.

max_ack_delay:
: The maximum amount of time by which the receiver intends to delay
  acknowledgments, in milliseconds.  The actual ack_delay in a
  received ACK frame may be larger due to late timers, reordering,
  or lost ACKs.

loss_time:
: 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.  Described
  in detail above in {{tracking-sent-packets}}.

### Initialization

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

~~~
   loss_detection_timer.reset()
   crypto_count = 0
   pto_count = 0
   loss_time = 0
   smoothed_rtt = 0
   rttvar = 0
   min_rtt = infinite
   time_of_last_sent_ack_eliciting_packet = 0
   time_of_last_sent_crypto_packet = 0
   largest_sent_packet = 0
   largest_acked_packet = 0
~~~

### On Sending a Packet

After a packet is sent, information about the packet is stored.  The parameters
to OnPacketSent are described in detail above in {{sent-packets-fields}}.

Pseudocode for OnPacketSent follows:

~~~
 OnPacketSent(packet_number, ack_eliciting, in_flight,
              is_crypto_packet, sent_bytes):
   largest_sent_packet = packet_number
   sent_packets[packet_number].packet_number = packet_number
   sent_packets[packet_number].time_sent = now
   sent_packets[packet_number].ack_eliciting = ack_eliciting
   sent_packets[packet_number].in_flight = in_flight
   if (ack_eliciting):
     if (is_crypto_packet):
       time_of_last_sent_crypto_packet = now
     time_of_last_sent_ack_eliciting_packet = now
     OnPacketSentCC(sent_bytes)
     sent_packets[packet_number].size = sent_bytes
     SetLossDetectionTimer()
~~~

### On Receiving an Acknowledgment

When an ACK frame is received, it may newly acknowledge any number of packets.

Pseudocode for OnAckReceived and UpdateRtt follow:

~~~
  OnAckReceived(ack):
    largest_acked_packet = max(largest_acked_packet,
                               ack.largest_acked)

    // If the largest acknowledged is newly acked and
    // ack-eliciting, update the RTT.
    if (sent_packets[ack.largest_acked] &&
        sent_packets[ack.largest_acked].ack_eliciting):
      latest_rtt =
        now - sent_packets[ack.largest_acked].time_sent
      UpdateRtt(latest_rtt, ack.ack_delay)

    // Process ECN information if present.
    if (ACK frame contains ECN information):
       ProcessECN(ack)

    // Find all newly acked packets in this ACK frame
    newly_acked_packets = DetermineNewlyAckedPackets(ack)
    if (newly_acked_packets.empty()):
      return

    for acked_packet in newly_acked_packets:
      OnPacketAcked(acked_packet.packet_number)

    crypto_count = 0
    pto_count = 0

    DetectLostPackets()
    SetLossDetectionTimer()


  UpdateRtt(latest_rtt, ack_delay):
    // min_rtt ignores ack delay.
    min_rtt = min(min_rtt, latest_rtt)
    // Limit ack_delay by max_ack_delay
    ack_delay = min(ack_delay, max_ack_delay)
    // Adjust for ack delay if 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
~~~

### On Packet Acknowledgment

When a packet is acknowledged for the first time, the following OnPacketAcked
function is called.  Note that a single ACK frame may newly acknowledge several
packets. OnPacketAcked must be called once for each of these newly acknowledged
packets.

OnPacketAcked takes one parameter, acked_packet, which is the struct detailed in
{{sent-packets-fields}}.

Pseudocode for OnPacketAcked follows:

~~~
   OnPacketAcked(acked_packet):
     if (acked_packet.ack_eliciting):
       OnPacketAckedCC(acked_packet)
     sent_packets.remove(acked_packet.packet_number)
~~~

### Setting the Loss Detection Timer

QUIC loss detection uses a single timer for all timeout loss detection.  The
duration of the timer is based on the timer's mode, which is set in the packet
and timer events further below.  The function SetLossDetectionTimer defined
below shows how the single timer is set.

This algorithm may result in the timer being set in the past, particularly if
timers wake up late. Timers set in the past SHOULD fire immediately.

Pseudocode for SetLossDetectionTimer follows:

~~~
 SetLossDetectionTimer():
    // Don't arm timer if there are no ack-eliciting packets
    // in flight.
    if (no ack-eliciting packets in flight):
      loss_detection_timer.cancel()
      return

    if (crypto packets are in flight):
      // Crypto retransmission timer.
      if (smoothed_rtt == 0):
        timeout = 2 * kInitialRtt
      else:
        timeout = 2 * smoothed_rtt
      timeout = max(timeout, kGranularity)
      timeout = timeout * (2 ^ crypto_count)
      loss_detection_timer.set(
        time_of_last_sent_crypto_packet + timeout)
      return
    if (loss_time != 0):
      // Time threshold loss detection.
      loss_detection_timer.set(loss_time)
      return

    // Calculate PTO duration
    timeout =
      smoothed_rtt + 4 * rttvar + max_ack_delay
    timeout = max(timeout, kGranularity)
    timeout = timeout * (2 ^ pto_count)

    loss_detection_timer.set(
      time_of_last_sent_ack_eliciting_packet + timeout)
~~~

### On Timeout

When the loss detection timer expires, the timer's mode determines the action
to be performed.

Pseudocode for OnLossDetectionTimeout follows:

~~~
   OnLossDetectionTimeout():
     if (crypto packets are in flight):
       // Crypto retransmission timeout.
       RetransmitUnackedCryptoData()
       crypto_count++
     else if (loss_time != 0):
       // Time threshold loss Detection
       DetectLostPackets()
     else:
       // PTO
       SendTwoPackets()
       pto_count++

     SetLossDetectionTimer()
~~~

### Detecting Lost Packets

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 acknowledged packet
is supplied.

Pseudocode for DetectLostPackets follows:

~~~
DetectLostPackets():
  loss_time = 0
  lost_packets = {}
  loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)

  // Packets sent before this time are deemed lost.
  lost_send_time = now() - loss_delay

  // Packets with packet numbers before this are deemed lost.
  lost_pn = largest_acked_packet - kPacketThreshold

  foreach unacked in sent_packets:
    if (unacked.packet_number > largest_acked_packet):
      continue

    // Mark packet as lost, or set time when it should be marked.
    if (unacked.time_sent <= lost_send_time ||
        unacked.packet_number <= lost_pn):
      sent_packets.remove(unacked.packet_number)
      if (unacked.in_flight):
        lost_packets.insert(unacked)
    else if (loss_time == 0):
      loss_time = unacked.time_sent + loss_delay
    else:
      loss_time = min(loss_time, unacked.time_sent + loss_delay)

  // Inform the congestion controller of lost packets and
  // let it decide whether to retransmit immediately.
  if (!lost_packets.empty()):
    OnPacketsLost(lost_packets)
~~~

## 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}}.  NewReno is a
congestion window based congestion control.  QUIC specifies the congestion
window in bytes rather than packets due to finer control and the ease of
appropriate byte counting {{?RFC3465}}.

QUIC hosts MUST NOT send packets if they would increase bytes_in_flight (defined
in {{vars-of-interest}}) beyond the available congestion window, unless the
packet is a probe packet sent after a PTO timer expires, as described in
{{pto}}.

Implementations MAY use other congestion control algorithms, such as
Cubic {{?RFC8312}}, and endpoints MAY use different algorithms from one another.
The signals QUIC provides for congestion control are generic and are designed
to support different algorithms.

## Explicit Congestion Notification {#congestion-ecn}

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.

## Slow Start

QUIC begins every connection in slow start and exits slow start upon loss or
upon increase in the ECN-CE counter. QUIC re-enters slow start anytime the
congestion window is less than ssthresh, which typically only occurs after an
PTO. While in slow start, QUIC increases the congestion window by the number of
bytes acknowledged when each acknowledgment is processed.


## 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.

## Recovery Period

Recovery is a period of time beginning with detection of a lost packet or an
increase in the ECN-CE counter. Because QUIC does not retransmit packets,
it defines the end of recovery as a packet sent after the start of recovery
being acknowledged. This is slightly different from TCP's definition of
recovery, which ends when the lost packet that started recovery is acknowledged.

The recovery period limits congestion window reduction to once per round trip.
During recovery, the congestion window remains unchanged irrespective of new
losses or increases in the ECN-CE counter.


## Probe Timeout

Probe packets MUST NOT be blocked by the congestion controller.  A sender MUST
however count these packets as being additionally in flight, since these packets
adds network load without establishing packet loss.  Note that sending probe
packets might cause the sender's bytes in flight to exceed the congestion window
until an acknowledgement is received that establishes loss or delivery of
packets.

If a threshold number of consecutive PTOs have occurred (pto_count is more than
kPersistentCongestionThreshold, see {{cc-consts-of-interest}}), the network is
considered to be experiencing persistent congestion, and the sender's congestion
window MUST be reduced to the minimum congestion window.

## 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).


## Sending data after an idle period

A sender becomes idle if it ceases to send data and has no bytes in flight.  A
sender's congestion window MUST not increase while it is idle.

When sending data after becoming idle, a sender MUST reset its congestion window
to the initial congestion window (see Section 4.1 of {{?RFC5681}}), unless it
paces the sending of packets. A sender MAY retain its congestion window if it
paces the sending of any packets in excess of the initial congestion window.

A sender MAY implement alternate mechanisms to update its congestion window
after idle periods, such as those proposed for TCP in {{?RFC7661}}.

## Discarding Packet Number Space State

When keys for an packet number space are discarded, any packets sent with those
keys are removed from the count of bytes in flight.  No loss events will occur
any in-flight packets from that space, as a result of discarding loss recovery
state (see {{discard-initial}}).  Note that it is expected that keys are
discarded after those packets would be declared lost, but Initial secrets are
destroyed earlier.

## Pseudocode

### Constants of interest {#cc-consts-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 data in flight, 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.

kPersistentCongestionThreshold:
: Number of consecutive PTOs after which network is considered to be
  experiencing persistent congestion.  The rationale for this threshold is to
  enable a sender to use initial PTOs for aggressive probing, similar to Tail
  Loss Probe (TLP) in TCP {{TLP}} {{RACK}}.  Once the number of consecutive PTOs
  reaches this threshold - that is, persistent congestion is established - the
  sender responds by collapsing its congestion window to kMinimumWindow, similar
  to a Retransmission Timeout (RTO) in TCP {{RFC5681}}.  The RECOMMENDED value
  for kPersistentCongestionThreshold is 2, which is equivalent to having two
  TLPs before an RTO in TCP.

### Variables of interest {#vars-of-interest}

Variables required to implement the congestion control mechanisms
are described in this section.

ecn_ce_counter:
: The highest value reported for the ECN-CE counter by the peer in an ACK
  frame. This variable is used to detect increases in the reported ECN-CE
  counter.

bytes_in_flight:
: The sum of the size in bytes of all sent packets that contain at least one
  ack-eliciting or PADDING frame, and have not been acked or declared
  lost. The size does not include IP or UDP overhead, but does include the QUIC
  header and AEAD overhead.  Packets only containing ACK frames do not count
  towards bytes_in_flight to ensure congestion control does not impede
  congestion feedback.

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

recovery_start_time:
: The time when QUIC first detects a loss, causing it to enter recovery.
  When a packet sent after this time 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.

### Initialization

At the beginning of the connection, initialize the congestion control
variables as follows:

~~~
   congestion_window = kInitialWindow
   bytes_in_flight = 0
   recovery_start_time = 0
   ssthresh = infinite
   ecn_ce_counter = 0
~~~

### 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
~~~

### On Packet Acknowledgement

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

~~~
   InRecovery(sent_time):
     return sent_time <= recovery_start_time

   OnPacketAckedCC(acked_packet):
     // Remove from bytes_in_flight.
     bytes_in_flight -= acked_packet.size
     if (InRecovery(acked_packet.time_sent)):
       // Do not increase congestion window in recovery period.
       return
     if (congestion_window < ssthresh):
       // Slow start.
       congestion_window += acked_packet.size
     else:
       // Congestion avoidance.
       congestion_window += kMaxDatagramSize * acked_packet.size
           / congestion_window
~~~

### On New Congestion Event

Invoked from ProcessECN and OnPacketsLost when a new congestion event is
detected. May start a new recovery period and reduces the congestion
window.

~~~
   CongestionEvent(sent_time):
     // Start a new congestion event if the sent time is larger
     // than the start time of the previous recovery epoch.
     if (!InRecovery(sent_time)):
       recovery_start_time = Now()
       congestion_window *= kLossReductionFactor
       congestion_window = max(congestion_window, kMinimumWindow)
       ssthresh = congestion_window
       // Collapse congestion window if persistent congestion
       if (pto_count > kPersistentCongestionThreshold):
         congestion_window = kMinimumWindow
~~~

### 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 was sent after the start of the previous
       // recovery epoch.
       CongestionEvent(sent_packets[ack.largest_acked].time_sent)
~~~


### 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.size
     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.time_sent)
~~~


# Security Considerations

## 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.

## 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.

## Misreporting ECN Markings

A receiver can misreport ECN markings to alter the congestion response of a
sender.  Suppressing reports of ECN-CE markings could cause a sender to
increase their send rate.  This increase could result in congestion and loss.

A sender MAY attempt to detect suppression of reports by marking occasional
packets that they send with ECN-CE.  If a packet marked with ECN-CE is not
reported as having been marked when the packet is acknowledged, the sender
SHOULD then disable ECN for that path.

Reporting additional ECN-CE markings will cause a sender to reduce their sending
rate, which is similar in effect to advertising reduced connection flow control
limits and so no advantage is gained by doing so.

Endpoints choose the congestion controller that they use.  Though congestion
controllers generally treat reports of ECN-CE markings as equivalent to loss
[RFC8311], the exact response for each controller could be different.  Failure
to correctly respond to information about ECN markings is therefore difficult to
detect.


# IANA Considerations

This document has no IANA actions.  Yet.


--- back

# Change Log

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

Issue and pull request numbers are listed with a leading octothorp.

## Since draft-ietf-quic-recovery-16

- Unify TLP and RTO into a single PTO; eliminate min RTO, min TLP and min crypto
  timeouts; eliminate timeout validation (#2114, #2166, #2168, #1017)
- Redefine how congestion avoidance in terms of when the period starts (#1928,
  #1930)
- Document what needs to be tracked for packets that are in flight (#765, #1724,
  #1939)
- Integrate both time and packet thresholds into loss detection (#1969, #1212,
  #934, #1974)
- Reduce congestion window after idle, unless pacing is used (#2007, #2023)
- Disable RTT calculation for packets that don't elicit acknowledgment (#2060,
  #2078)
- Limit ack_delay by max_ack_delay (#2060, #2099)
- Initial keys are discarded once Handshake are avaialble (#1951, #2045)
- Reorder ECN and loss detection in pseudocode (#2142)
- Only cancel loss detection timer if ack-eliciting packets are in flight
  (#2093, #2117)


## Since draft-ietf-quic-recovery-14

- Used max_ack_delay from transport params (#1796, #1782)
- Merge ACK and ACK_ECN (#1783)

## Since draft-ietf-quic-recovery-13

- Corrected the lack of ssthresh reduction in CongestionEvent pseudocode (#1598)
- Considerations for ECN spoofing (#1426, #1626)
- Clarifications for PADDING and congestion control (#837, #838, #1517, #1531,
  #1540)
- Reduce early retransmission timer to RTT/8 (#945, #1581)
- Packets are declared lost after an RTO is verified (#935, #1582)


## Since draft-ietf-quic-recovery-12

- Changes to manage separate packet number spaces and encryption levels (#1190,
  #1242, #1413, #1450)
- Added ECN feedback mechanisms and handling; new ACK_ECN frame (#804, #805,
  #1372)


## Since draft-ietf-quic-recovery-11

No significant changes.

## Since draft-ietf-quic-recovery-10

- Improved text on ack generation (#1139, #1159)
- Make references to TCP recovery mechanisms informational (#1195)
- Define time_of_last_sent_handshake_packet (#1171)
- Added signal from TLS the data it includes needs to be sent in a Retry packet
  (#1061, #1199)
- Minimum RTT (min_rtt) is initialized with an infinite value (#1169)

## Since draft-ietf-quic-recovery-09

No significant changes.

## Since draft-ietf-quic-recovery-08

- Clarified pacing and RTO (#967, #977)

## Since draft-ietf-quic-recovery-07

- Include Ack Delay in RTO(and TLP) computations (#981)
- Ack Delay in SRTT computation (#961)
- Default RTT and Slow Start (#590)
- Many editorial fixes.

## Since draft-ietf-quic-recovery-06

No significant changes.

## Since draft-ietf-quic-recovery-05

- Add more congestion control text (#776)

## 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


# Acknowledgments
{:numbered="false"}