---
title: QUIC Loss Detection and Congestion Control
abbrev: QUIC Loss Detection
number: 9002
docName: draft-ietf-quic-recovery-34
date: 2021-05
category: std
consensus: true
ipr: trust200902
area: Transport
workgroup: QUIC
keyword:
  - bbr
  - delay-sensitive congestion control
  - fec
  - loss-tolerant congestion control
  - next generation

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: 2021-05
    seriesinfo:
      RFC: 9000
      DOI: 10.17487/RFC9000
    author:
      -
        ins: J. Iyengar
        name: Jana Iyengar
        org: Fastly
        role: editor
      -
        ins: M. Thomson
        name: Martin Thomson
        org: Mozilla
        role: editor

  QUIC-TLS:
    title: "Using TLS to Secure QUIC"
    date: 2021-05
    seriesinfo:
      RFC: 9001
      DOI: 10.17487/RFC9001
    author:
      -
        ins: M. Thomson
        name: Martin Thomson
        org: Mozilla
        role: editor
      -
        ins: S. Turner
        name: Sean Turner
        org: sn3rd
        role: editor

  RFC8085:

informative:

  FACK:
    title: "Forward acknowledgement: Refining TCP Congestion Control"
    author:
      -
        initials: M.
        surname: Mathis
      -
        initials: J.
        surname: Mahdavi
    date: 1996-08
    refcontent: ACM SIGCOMM Computer Communication Review
    seriesinfo:
      DOI: 10.1145/248157.248181

  RETRANSMISSION:
    title: "Improving Round-Trip Time Estimates in Reliable Transport Protocols"
    author:
      -
        initials: P.
        surname: Karn
      -
        initials: C.
        surname: Partridge
    date: 1991-11
    refcontent: ACM Transactions on Computer Systems
    seriesinfo:
       DOI: 10.1145/118544.118549

  RFC3465:

--- abstract

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


--- middle

# Introduction


QUIC is a secure, general-purpose transport protocol, described in
{{QUIC-TRANSPORT}}. This document describes loss detection and congestion
control mechanisms for QUIC.

# 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-eliciting frames:

: All frames other than ACK, PADDING, and CONNECTION_CLOSE are considered
  ack-eliciting.

Ack-eliciting packets:

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

In-flight packets:

: Packets are considered in flight when they are ack-eliciting or contain a
  PADDING frame, and they have been sent but are not acknowledged, declared
  lost, or discarded along with old keys.

# 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 {{Section 12.3 of QUIC-TRANSPORT}}.  Packet numbers never repeat
within a packet number space for the lifetime of a connection.  Packet numbers
are sent in monotonically increasing order within a space, preventing ambiguity.
It is permitted for some packet numbers to never be used, leaving intentional
gaps.

This design obviates the need for disambiguating between transmissions and
retransmissions; this 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
  acknowledgment.

* Packets containing frames besides ACK or CONNECTION_CLOSE frames count toward
  congestion control limits and are considered to be in flight.

* 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. However, protocol differences
between QUIC and TCP contribute to algorithmic differences. These protocol
differences are briefly described 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 that the
acknowledgment of packets sent with one level of encryption will not
cause spurious retransmission of packets sent with a different
encryption level.  Congestion control and round-trip time (RTT)
measurement are unified across packet number spaces.

## Monotonically Increasing Packet Numbers

TCP conflates transmission order at the sender with delivery order at the
receiver, resulting in the retransmission ambiguity problem
{{RETRANSMISSION}}.  QUIC separates transmission order from delivery order:
packet numbers indicate transmission order, and delivery order is determined by
the stream offsets in 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 includes 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 nontrivial task, especially when TCP timestamps are
not available.

## Clearer Loss Epoch

QUIC starts a loss epoch when a packet is lost. The loss epoch ends when any
packet sent after the start of the epoch is acknowledged.  TCP waits for the gap
in the sequence number space to be filled, and so if a segment is lost multiple
times in a row, the loss epoch may not end for several round trips. Because both
should reduce their congestion windows only once per epoch, QUIC will do it once
for every round trip that experiences loss, while TCP may only do it once across
multiple round trips.

## No Reneging

QUIC ACK frames contain information similar to that in TCP Selective
Acknowledgments (SACKs) {{?RFC2018}}. However, QUIC does not allow a packet
acknowledgment to be reneged, greatly simplifying implementations on both sides
and reducing memory pressure on the sender.

## More ACK Ranges

QUIC supports many ACK ranges, as opposed to TCP's three SACK ranges.  In
high-loss environments, this speeds recovery, reduces spurious retransmits, and
ensures forward progress without relying on timeouts.

## Explicit Correction For Delayed Acknowledgments

QUIC endpoints measure the delay incurred between when a packet is received and
when the corresponding acknowledgment is sent, allowing a peer to maintain a
more accurate RTT estimate; see {{Section 13.2 of QUIC-TRANSPORT}}.

## Probe Timeout Replaces RTO and TLP

QUIC uses a probe timeout (PTO; see {{pto}}), with a timer based on TCP's
retransmission timeout (RTO) computation; see {{?RFC6298}}.  QUIC's PTO includes
the peer's maximum expected acknowledgment delay instead of using a fixed
minimum timeout.

Similar to the RACK-TLP loss detection algorithm for TCP {{?RFC8985}}, QUIC does
not collapse the congestion window when the PTO expires, since a single packet
loss at the tail does not indicate persistent congestion.  Instead, QUIC
collapses the congestion window when persistent congestion is declared; see
{{persistent-congestion}}. In doing this, QUIC avoids unnecessary congestion
window reductions, obviating the need for correcting mechanisms such as Forward
RTO-Recovery (F-RTO) {{?RFC5682}}. Since QUIC does not collapse the congestion
window on a PTO expiration, a QUIC sender is not limited from sending more
in-flight packets after a PTO expiration if it still has available congestion
window. This occurs when a sender is application limited and the PTO timer
expires. This is more aggressive than TCP's RTO mechanism when application
limited, but identical when not application limited.

QUIC allows probe packets to temporarily exceed the congestion window whenever
the timer expires.

## The Minimum Congestion Window Is Two Packets

TCP uses a minimum congestion window of one packet. However, loss of that single
packet means that the sender needs to wait for a PTO to recover ({{pto}}), which
can be much longer than an RTT.  Sending a single ack-eliciting packet also
increases the chances of incurring additional latency when a receiver delays its
acknowledgment.

QUIC therefore recommends that the minimum congestion window be two
packets. While this increases network load, it is considered safe since the
sender will still reduce its sending rate exponentially under persistent
congestion ({{pto}}).

## Handshake Packets Are Not Special

TCP treats the loss of SYN or SYN-ACK packet as persistent congestion and
reduces the congestion window to one packet; see {{?RFC5681}}. QUIC treats loss
of a packet containing handshake data the same as other losses.

# Estimating the Round-Trip Time {#compute-rtt}

At a high level, an endpoint measures the time from when a packet was sent to
when it is acknowledged as an RTT sample. The endpoint uses RTT samples and
peer-reported host delays (see {{Section 13.2 of QUIC-TRANSPORT}}) to generate a
statistical description of the network path's RTT. An endpoint computes the
following three values for each path: the minimum value over a period of time
(min_rtt), an exponentially weighted moving average (smoothed_rtt), and the mean
deviation (referred to as "variation" in the rest of this document) in the
observed RTT samples (rttvar).

## Generating RTT Samples {#latest-rtt}

An endpoint generates an RTT sample on receiving an ACK frame that meets the
following two conditions:

- the largest acknowledged packet number is newly acknowledged, and

- at least one of the newly acknowledged packets was ack-eliciting.

The RTT sample, latest_rtt, is generated as the time elapsed since the largest
acknowledged packet was sent:

~~~pseudocode
latest_rtt = ack_time - send_time_of_largest_acked
~~~

An RTT sample is generated using only the largest acknowledged packet in the
received ACK frame.  This is because a peer reports acknowledgment delays for
only the largest acknowledged packet in an ACK frame.  While the reported
acknowledgment delay is not used by the RTT sample measurement, it is used to
adjust the RTT sample in subsequent computations of smoothed_rtt and rttvar
({{smoothed-rtt}}).

To avoid generating multiple RTT samples for a single packet, an ACK frame
SHOULD NOT be used to update RTT estimates if it does not newly acknowledge the
largest acknowledged packet.

An RTT sample MUST NOT be generated on receiving an ACK frame that does not
newly acknowledge at least one ack-eliciting packet. A peer usually does not
send an ACK frame when only non-ack-eliciting packets are received. Therefore,
an ACK frame that contains acknowledgments for only non-ack-eliciting packets
could include an arbitrarily large ACK Delay value.  Ignoring
such ACK frames avoids complications in subsequent smoothed_rtt and rttvar
computations.

A sender might generate multiple RTT samples per RTT when multiple ACK frames
are received within an RTT.  As suggested in {{?RFC6298}}, doing so might result
in inadequate history in smoothed_rtt and rttvar.  Ensuring that RTT estimates
retain sufficient history is an open research question.

## Estimating min_rtt {#min-rtt}

min_rtt is the sender's estimate of the minimum RTT observed for a given network
path over a period of time. In this document, min_rtt is used by loss detection
to reject implausibly small RTT samples.

min_rtt MUST be set to the latest_rtt on the first RTT sample. min_rtt MUST be
set to the lesser of min_rtt and latest_rtt ({{latest-rtt}}) on all other
samples.

An endpoint uses only locally observed times in computing the min_rtt and does
not adjust for acknowledgment delays reported by the peer. Doing so allows the
endpoint to set a lower bound for the smoothed_rtt based entirely on what it
observes (see {{smoothed-rtt}}) and limits potential underestimation due to
erroneously reported delays by the peer.

The RTT for a network path may change over time. If a path's actual RTT
decreases, the min_rtt will adapt immediately on the first low sample.  If the
path's actual RTT increases, however, the min_rtt will not adapt to it, allowing
future RTT samples that are smaller than the new RTT to be included in
smoothed_rtt.

Endpoints SHOULD set the min_rtt to the newest RTT sample after persistent
congestion is established. This avoids repeatedly declaring persistent
congestion when the RTT increases. This also allows a connection to reset
its estimate of min_rtt and smoothed_rtt after a disruptive network event;
see {{smoothed-rtt}}.

Endpoints MAY reestablish the min_rtt at other times in the connection, such as
when traffic volume is low and an acknowledgment is received with a low
acknowledgment delay. Implementations SHOULD NOT refresh the min_rtt
value too often since the actual minimum RTT of the path is not
frequently observable.


## Estimating smoothed_rtt and rttvar {#smoothed-rtt}

smoothed_rtt is an exponentially weighted moving average of an endpoint's RTT
samples, and rttvar estimates the variation in the RTT samples using a mean
variation.

The calculation of smoothed_rtt uses RTT samples after adjusting them for
acknowledgment delays. These delays are decoded from the ACK Delay field of
ACK frames as described in {{Section 19.3 of QUIC-TRANSPORT}}.

The peer might report acknowledgment delays that are larger than the peer's
max_ack_delay during the handshake ({{Section 13.2.1 of QUIC-TRANSPORT}}). To
account for this, the endpoint SHOULD ignore max_ack_delay until the handshake
is confirmed, as defined in {{Section 4.1.2 of QUIC-TLS}}. When they occur,
these large acknowledgment delays are likely to be non-repeating and limited to
the handshake. The endpoint can therefore use them without limiting them to the
max_ack_delay, avoiding unnecessary inflation of the RTT estimate.

Note that a large acknowledgment delay can result in a substantially inflated
smoothed_rtt if there is an error either in the peer's reporting of the
acknowledgment delay or in the endpoint's min_rtt estimate.  Therefore, prior
to handshake confirmation, an endpoint MAY ignore RTT samples if adjusting
the RTT sample for acknowledgment delay causes the sample to be less than the
min_rtt.

After the handshake is confirmed, any acknowledgment delays reported by the
peer that are greater than the peer's max_ack_delay are attributed to
unintentional but potentially repeating delays, such as scheduler latency at the
peer or loss of previous acknowledgments.  Excess delays could also be due to
a noncompliant receiver.  Therefore, these extra delays are considered
effectively part of path delay and incorporated into the RTT estimate.

Therefore, when adjusting an RTT sample using peer-reported acknowledgment
delays, an endpoint:

- MAY ignore the acknowledgment delay for Initial packets, since these
  acknowledgments are not delayed by the peer ({{Section 13.2.1 of
  QUIC-TRANSPORT}});

- SHOULD ignore the peer's max_ack_delay until the handshake is confirmed;

- MUST use the lesser of the acknowledgment delay and the peer's max_ack_delay
  after the handshake is confirmed; and

- MUST NOT subtract the acknowledgment delay from the RTT sample if the
  resulting value is smaller than the min_rtt.  This limits the underestimation
  of the smoothed_rtt due to a misreporting peer.

Additionally, an endpoint might postpone the processing of acknowledgments when
the corresponding decryption keys are not immediately available. For example, a
client might receive an acknowledgment for a 0-RTT packet that it cannot
decrypt because 1-RTT packet protection keys are not yet available to it. In
such cases, an endpoint SHOULD subtract such local delays from its RTT sample
until the handshake is confirmed.

Similar to {{?RFC6298}}, smoothed_rtt and rttvar are computed as follows.

An endpoint initializes the RTT estimator during connection establishment and
when the estimator is reset during connection migration; see {{Section 9.4 of
QUIC-TRANSPORT}}. Before any RTT samples are available for a new path or when
the estimator is reset, the estimator is initialized using the initial RTT; see
{{pto-handshake}}.

smoothed_rtt and rttvar are initialized as follows, where kInitialRtt contains
the initial RTT value:

~~~pseudocode
smoothed_rtt = kInitialRtt
rttvar = kInitialRtt / 2
~~~

RTT samples for the network path are recorded in latest_rtt; see
{{latest-rtt}}. On the first RTT sample after initialization, the estimator is
reset using that sample. This ensures that the estimator retains no history of
past samples.  Packets sent on other paths do not contribute RTT samples to the
current path, as described in {{Section 9.4 of QUIC-TRANSPORT}}.

On the first RTT sample after initialization, smoothed_rtt and rttvar are set as
follows:

~~~pseudocode
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
~~~

On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:

~~~pseudocode
ack_delay = decoded acknowledgment delay from ACK frame
if (handshake confirmed):
  ack_delay = min(ack_delay, max_ack_delay)
adjusted_rtt = latest_rtt
if (latest_rtt >= min_rtt + ack_delay):
  adjusted_rtt = latest_rtt - ack_delay
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
rttvar_sample = abs(smoothed_rtt - adjusted_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
~~~

# Loss Detection {#loss-detection}

QUIC senders use acknowledgments to detect lost packets and a PTO to ensure
acknowledgments are received; see {{pto}}. This section provides a description
of these algorithms.

If a packet is lost, the QUIC transport needs to recover from that loss, such
as by retransmitting the data, sending an updated frame, or discarding the
frame.  For more information, see {{Section 13.3 of QUIC-TRANSPORT}}.

Loss detection is separate per packet number space, unlike RTT measurement and
congestion control, because RTT and congestion control are properties of the
path, whereas loss detection also relies upon key availability.

## Acknowledgment-Based Detection {#ack-loss-detection}

Acknowledgment-based loss detection implements the spirit of TCP's Fast
Retransmit {{?RFC5681}}, Early Retransmit {{?RFC5827}}, Forward Acknowledgment
{{FACK}}, SACK loss recovery {{?RFC6675}}, and RACK-TLP {{?RFC8985}}. This
section provides an overview of how these algorithms are implemented in QUIC.

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

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

* The packet was sent kPacketThreshold packets before an acknowledged packet
  ({{packet-threshold}}), or it was sent long enough in the past
  ({{time-threshold}}).

The acknowledgment indicates that a packet sent later was delivered, and 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 can detect spurious
retransmissions and increase the packet or time reordering threshold to
reduce future spurious retransmissions and loss events. Implementations with
adaptive time thresholds MAY choose to start with smaller initial reordering
thresholds to minimize recovery latency.

### Packet Threshold {#packet-threshold}

The RECOMMENDED initial value for the packet reordering threshold
(kPacketThreshold) is 3, based on best practices for TCP loss detection
{{?RFC5681}} {{?RFC6675}}.  In order to remain similar to TCP,
implementations SHOULD NOT use a packet threshold less than 3; see {{?RFC5681}}.

Some networks may exhibit higher degrees of packet reordering, causing a sender
to detect spurious losses. Additionally, packet reordering could be more common
with QUIC than TCP because network elements that could observe and reorder TCP
packets cannot do that for QUIC and also because QUIC packet numbers are
encrypted.  Algorithms that increase the reordering threshold after spuriously
detecting losses, such as RACK {{?RFC8985}}, have proven to be useful in TCP and
are expected to be at least as useful in QUIC.

### Time Threshold {#time-threshold}

Once a later packet within the same packet number space has been acknowledged,
an endpoint SHOULD declare an earlier packet lost if it was sent a threshold
amount of time in the past. To avoid declaring packets as lost too early, this
time threshold MUST be set to at least the local timer granularity, as
indicated by the kGranularity constant.  The time threshold is:

~~~pseudocode
max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity)
~~~

If packets sent prior to the largest acknowledged packet cannot yet be declared
lost, then a timer SHOULD be set for the remaining time.

Using max(smoothed_rtt, latest_rtt) protects from the two following cases:

* the latest RTT sample is lower than the smoothed RTT, perhaps due to
  reordering where the acknowledgment encountered a shorter path;

* the latest RTT sample is higher than the smoothed RTT, perhaps due to a
  sustained increase in the actual RTT, but the smoothed RTT has not yet caught
  up.

The RECOMMENDED time threshold (kTimeThreshold), expressed as an RTT multiplier,
is 9/8. The RECOMMENDED value of the timer granularity (kGranularity) is 1
millisecond.

<aside markdown="block">
Note: TCP's RACK {{?RFC8985}} specifies a slightly larger threshold, equivalent
to 5/4, for a similar purpose. Experience with QUIC shows that 9/8 works well.
</aside>

Implementations MAY experiment with absolute thresholds, thresholds from
previous connections, adaptive thresholds, or the including of RTT variation.
Smaller thresholds reduce reordering resilience and increase spurious
retransmissions, and larger thresholds increase loss detection delay.


## Probe Timeout {#pto}

A Probe Timeout (PTO) triggers the sending of one or two probe datagrams when
ack-eliciting packets are not acknowledged within the expected period of
time or the server may not have validated the client's address.  A PTO enables
a connection to recover from loss of tail packets or acknowledgments.

As with loss detection, the PTO is per packet number space. That is, a
PTO value is computed per packet number space.

A PTO timer expiration event does not indicate packet loss and MUST NOT cause
prior unacknowledged packets to be marked as lost. When an acknowledgment is
received that newly acknowledges packets, loss detection proceeds as dictated
by the packet and time threshold mechanisms; see {{ack-loss-detection}}.

The PTO algorithm used in QUIC implements the reliability functions of Tail Loss
Probe {{?RFC8985}}, RTO {{?RFC5681}}, and F-RTO algorithms for TCP
{{?RFC5682}}. The timeout computation is based on TCP's RTO period {{?RFC6298}}.

### Computing PTO

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

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

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

When the PTO is armed for Initial or Handshake packet number spaces, the
max_ack_delay in the PTO period computation is set to 0, since the peer is
expected to not delay these packets intentionally; see {{Section 13.2.1 of
QUIC-TRANSPORT}}.

The PTO period MUST be at least kGranularity to avoid the timer expiring
immediately.

When ack-eliciting packets in multiple packet number spaces are in flight, the
timer MUST be set to the earlier value of the Initial and Handshake packet
number spaces.

An endpoint MUST NOT set its PTO timer for the Application Data packet number
space until the handshake is confirmed. Doing so prevents the endpoint from
retransmitting information in packets when either the peer does not yet have the
keys to process them or the endpoint does not yet have the keys to process their
acknowledgments. For example, this can happen when a client sends 0-RTT packets
to the server; it does so without knowing whether the server will be able to
decrypt them. Similarly, this can happen when a server sends 1-RTT packets
before confirming that the client has verified the server's certificate and can
therefore read these 1-RTT packets.

A sender SHOULD restart its PTO timer every time an ack-eliciting packet is
sent or acknowledged, or when Initial or Handshake keys are discarded
({{Section 4.9 of QUIC-TLS}}). This ensures the PTO is always set based on the
latest estimate of the RTT and for the correct packet across packet
number spaces.

When a PTO timer expires, the PTO backoff MUST be increased, resulting in the
PTO period being set to twice its current value. The PTO backoff factor is reset
when an acknowledgment is received, except in the following case. A server
might take longer to respond to packets during the handshake than otherwise.  To
protect such a server from repeated client probes, the PTO backoff is not reset
at a client that is not yet certain that the server has finished validating the
client's address. That is, a client does not reset the PTO backoff factor on
receiving acknowledgments in Initial packets.

This exponential reduction in the sender's rate is important because consecutive
PTOs might be caused by loss of packets or acknowledgments due to severe
congestion.  Even when there are ack-eliciting packets in flight in multiple
packet number spaces, the exponential increase in PTO occurs across all spaces
to prevent excess load on the network.  For example, a timeout in the Initial
packet number space doubles the length of the timeout in the Handshake packet
number space.

The total length of time over which consecutive PTOs expire is limited by the
idle timeout.

The PTO timer MUST NOT be set if a timer is set for time threshold
loss detection; see {{time-threshold}}.  A timer that is set for time
threshold loss detection will expire earlier than the PTO timer
in most cases and is less likely to spuriously retransmit data.

### Handshakes and New Paths {#pto-handshake}

Resumed connections over the same network MAY use the previous connection's
final smoothed RTT value as the resumed connection's initial RTT.  When no
previous RTT is available, the initial RTT SHOULD be set to 333 milliseconds.
This results in handshakes starting with a PTO of 1 second, as recommended for
TCP's initial RTO; see {{Section 2 of RFC6298}}.

A connection MAY use the delay between sending a PATH_CHALLENGE and receiving a
PATH_RESPONSE to set the initial RTT (see kInitialRtt in
{{constants-of-interest}}) for a new path, but the delay SHOULD NOT be
considered an RTT sample.

When the Initial keys and Handshake keys are discarded (see
{{discarding-packets}}), any Initial packets and Handshake packets can
no longer be acknowledged, so they are removed from bytes in
flight. When Initial or Handshake keys are discarded, the PTO and loss
detection timers MUST be reset, because discarding keys indicates
forward progress and the loss detection timer might have been set for
a now-discarded packet number space.

#### Before Address Validation

Until the server has validated the client's address on the path, the amount of
data it can send is limited to three times the amount of data received,
as specified in {{Section 8.1 of QUIC-TRANSPORT}}. If no additional data can be
sent, the server's PTO timer MUST NOT be armed until datagrams have been
received from the client because packets sent on PTO count against the
anti-amplification limit.

When the server receives a datagram from the client, the amplification limit is
increased and the server resets the PTO timer.  If the PTO timer is then set to
a time in the past, it is executed immediately. Doing so avoids sending new
1-RTT packets prior to packets critical to the completion of the handshake.
In particular, this can happen when 0-RTT is accepted but the server fails to
validate the client's address.

Since the server could be blocked until more datagrams are received from the
client, it is the client's responsibility to send packets to unblock the server
until it is certain that the server has finished its address validation (see
{{Section 8 of QUIC-TRANSPORT}}). That is, the client MUST set the PTO timer
if the client has not received an acknowledgment for any of its Handshake
packets and the handshake is not confirmed (see {{Section 4.1.2 of QUIC-TLS}}),
even if there are no packets in flight. When the PTO fires, the client MUST
send a Handshake packet if it has Handshake keys, otherwise it MUST send an
Initial packet in a UDP datagram with a payload of at least 1200 bytes.

### Speeding up Handshake Completion

When a server receives an Initial packet containing duplicate CRYPTO data,
it can assume the client did not receive all of the server's CRYPTO data sent
in Initial packets, or the client's estimated RTT is too small. When a
client receives Handshake or 1-RTT packets prior to obtaining Handshake keys,
it may assume some or all of the server's Initial packets were lost.

To speed up handshake completion under these conditions, an endpoint MAY, for a
limited number of times per connection, send a packet containing
unacknowledged CRYPTO data earlier than the PTO expiry, subject to the address
validation limits in {{Section 8.1 of QUIC-TRANSPORT}}. Doing so at most once
for each connection is adequate to quickly recover from a single packet loss.
An endpoint that always retransmits packets in response to receiving packets
that it cannot process risks creating an infinite exchange of packets.

Endpoints can also use coalesced packets (see {{Section 12.2 of
QUIC-TRANSPORT}}) to ensure that each datagram elicits at least one
acknowledgment. For example, a client can coalesce an Initial packet containing
PING and PADDING frames with a 0-RTT data packet, and a server can coalesce an
Initial packet containing a PING frame with one or more packets in its first
flight.

### Sending Probe Packets

When a PTO timer expires, a sender MUST send at least one ack-eliciting packet
in the packet number space as a probe.  An endpoint MAY send up to two
full-sized datagrams containing ack-eliciting packets to avoid an expensive
consecutive PTO expiration due to a single lost datagram or to transmit data
from multiple packet number spaces. All probe packets sent on a PTO MUST be
ack-eliciting.

In addition to sending data in the packet number space for which the timer
expired, the sender SHOULD send ack-eliciting packets from other packet number
spaces with in-flight data, coalescing packets if possible.  This is
particularly valuable when the server has both Initial and Handshake data in
flight or when the client has both Handshake and Application Data in flight
because the peer might only have receive keys for one of the two packet number
spaces.

If the sender wants to elicit a faster acknowledgment on PTO, it can skip a
packet number to eliminate the acknowledgment delay.

An endpoint SHOULD include new data in packets that are sent on PTO expiration.
Previously sent data MAY be sent if no new data can be sent. Implementations
MAY use alternative strategies for determining the content of probe packets,
including sending new or retransmitted data based on the application's
priorities.

It is possible the sender has no new or previously sent data to send.
As an example, consider the following sequence of events: new application data
is sent in a STREAM frame, deemed lost, then retransmitted in a new packet,
and then the original transmission is acknowledged.  When there is no data to
send, the sender SHOULD send a PING or other ack-eliciting frame in a single
packet, rearming the PTO timer.

Alternatively, instead of sending an ack-eliciting packet, the sender MAY mark
any packets still in flight as lost.  Doing so avoids sending an additional
packet but increases the risk that loss is declared too aggressively, resulting
in an unnecessary rate reduction by the congestion controller.

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

When the PTO timer expires multiple times and new data cannot be sent,
implementations must choose between sending the same payload every time
or sending different payloads.  Sending the same payload may be simpler
and ensures the highest priority frames arrive first.  Sending different
payloads each time reduces the chances of spurious retransmission.


## Handling Retry Packets

A Retry packet causes a client to send another Initial packet, effectively
restarting the connection process.  A Retry packet indicates that the Initial
packet was received but not processed.  A Retry packet cannot be treated as an
acknowledgment because it does not indicate that a packet was processed or
specify the packet number.

Clients that receive a Retry packet reset congestion control and loss recovery
state, including resetting any pending timers.  Other connection state, in
particular cryptographic handshake messages, is retained; see
{{Section 17.2.5 of QUIC-TRANSPORT}}.

The client MAY compute an RTT estimate to the server as the time period from
when the first Initial packet was sent to when a Retry or a Version Negotiation
packet is received.  The client MAY use this value in place of its default for
the initial RTT estimate.

## Discarding Keys and Packet State {#discarding-packets}

When Initial and Handshake packet protection keys are discarded
(see {{Section 4.9 of QUIC-TLS}}), all packets that were sent with those keys
can no longer be acknowledged because their acknowledgments cannot be processed.
The sender MUST discard all recovery state associated with those packets
and MUST remove them from the count of bytes in flight.

Endpoints stop sending and receiving Initial packets once they start exchanging
Handshake packets; see {{Section 17.2.2.1 of QUIC-TRANSPORT}}. At this point,
recovery state for all in-flight Initial packets is discarded.

When 0-RTT is rejected, recovery state for all in-flight 0-RTT packets is
discarded.

If a server accepts 0-RTT, but does not buffer 0-RTT packets that arrive
before Initial packets, early 0-RTT packets will be declared lost, but that
is expected to be infrequent.

It is expected that keys are discarded at some time after the packets
encrypted with them are either acknowledged or declared lost. However,
Initial and Handshake secrets are discarded as soon as Handshake and
1-RTT keys are proven to be available to both client and server; see
{{Section 4.9.1 of QUIC-TLS}}.

# Congestion Control {#congestion-control}

This document specifies a sender-side congestion controller for QUIC similar to
TCP NewReno {{?RFC6582}}.

The signals QUIC provides for congestion control are generic and are designed to
support different sender-side algorithms. A sender can unilaterally choose a
different algorithm to use, such as CUBIC {{?RFC8312}}.

If a sender uses a different controller than that specified in this document,
the chosen controller MUST conform to the congestion control guidelines
specified in {{Section 3.1 of RFC8085}}.

Similar to TCP, packets containing only ACK frames do not count toward bytes
in flight and are not congestion controlled.  Unlike TCP, QUIC can detect the
loss of these packets and MAY use that information to adjust the congestion
controller or the rate of ACK-only packets being sent, but this document does
not describe a mechanism for doing so.

The congestion controller is per path, so packets sent on other paths do not
alter the current path's congestion controller, as described in
{{Section 9.4 of QUIC-TRANSPORT}}.

The algorithm in this document specifies and uses the controller's congestion
window in bytes.

An endpoint MUST NOT send a packet if it would cause bytes_in_flight (see
{{vars-of-interest}}) to be larger than the congestion window, unless the packet
is sent on a PTO timer expiration (see {{pto}}) or when entering recovery
(see {{recovery-period}}).

## Explicit Congestion Notification {#congestion-ecn}

If a path has been validated to support Explicit Congestion Notification (ECN)
{{!RFC3168}} {{?RFC8311}}, QUIC treats a Congestion Experienced (CE) codepoint
in the IP header as a signal of congestion. This document specifies an
endpoint's response when the peer-reported ECN-CE count increases; see {{Section
13.4.2 of QUIC-TRANSPORT}}.

## Initial and Minimum Congestion Window {#initial-cwnd}

QUIC begins every connection in slow start with the congestion window set to an
initial value.  Endpoints SHOULD use an initial congestion window of ten times
the maximum datagram size (max_datagram_size), while limiting the window to the
larger of 14,720 bytes or twice the maximum datagram size. This follows the
analysis and recommendations in {{?RFC6928}}, increasing the byte limit to
account for the smaller 8-byte overhead of UDP compared to the 20-byte overhead
for TCP.

If the maximum datagram size changes during the connection, the initial
congestion window SHOULD be recalculated with the new size.  If the maximum
datagram size is decreased in order to complete the handshake, the
congestion window SHOULD be set to the new initial congestion window.

Prior to validating the client's address, the server can be further limited by
the anti-amplification limit as specified in {{Section 8.1 of QUIC-TRANSPORT}}.
Though the anti-amplification limit can prevent the congestion window from
being fully utilized and therefore slow down the increase in congestion window,
it does not directly affect the congestion window.

The minimum congestion window is the smallest value the congestion window can
attain in response to loss, an increase in the peer-reported ECN-CE count,
or persistent congestion.  The RECOMMENDED value is 2 * max_datagram_size.

## Congestion Control States

The NewReno congestion controller described in this document has three
distinct states, as shown in {{fig-cc-fsm}}.

~~~
                 New path or      +------------+
            persistent congestion |   Slow     |
        (O)---------------------->|   Start    |
                                  +------------+
                                        |
                                Loss or |
                        ECN-CE increase |
                                        v
 +------------+     Loss or       +------------+
 | Congestion |  ECN-CE increase  |  Recovery  |
 | Avoidance  |------------------>|   Period   |
 +------------+                   +------------+
           ^                            |
           |                            |
           +----------------------------+
              Acknowledgment of packet
                sent during recovery
~~~
{: #fig-cc-fsm title="Congestion Control States and Transitions"}

These states and the transitions between them are described in subsequent
sections.

### Slow Start

A NewReno sender is in slow start any time the congestion window is below the
slow start threshold. A sender begins in slow start because the slow start
threshold is initialized to an infinite value.

While a sender is in slow start, the congestion window increases by the number
of bytes acknowledged when each acknowledgment is processed. This results in
exponential growth of the congestion window.

The sender MUST exit slow start and enter a recovery period when a packet is
lost or when the ECN-CE count reported by its peer increases.

A sender reenters slow start any time the congestion window is less than the
slow start threshold, which only occurs after persistent congestion is
declared.

### Recovery {#recovery-period}

A NewReno sender enters a recovery period when it detects the loss of a packet
or when the ECN-CE count reported by its peer increases. A sender that is
already in a recovery period stays in it and does not reenter it.

On entering a recovery period, a sender MUST set the slow start threshold to
half the value of the congestion window when loss is detected. The congestion
window MUST be set to the reduced value of the slow start threshold before
exiting the recovery period.

Implementations MAY reduce the congestion window immediately upon entering a
recovery period or use other mechanisms, such as Proportional Rate Reduction
{{?PRR=RFC6937}}, to reduce the congestion window more gradually. If the
congestion window is reduced immediately, a single packet can be sent prior to
reduction. This speeds up loss recovery if the data in the lost packet is
retransmitted and is similar to TCP as described in {{Section 5 of RFC6675}}.

The recovery period aims to limit congestion window reduction to once per round
trip. Therefore, during a recovery period, the congestion window does not change
in response to new losses or increases in the ECN-CE count.

A recovery period ends and the sender enters congestion avoidance when a packet
sent during the recovery period is acknowledged. This is slightly different
from TCP's definition of recovery, which ends when the lost segment that
started recovery is acknowledged {{?RFC5681}}.

### Congestion Avoidance

A NewReno sender is in congestion avoidance any time the congestion window is
at or above the slow start threshold and not in a recovery period.

A sender in congestion avoidance uses an Additive Increase Multiplicative
Decrease (AIMD) approach that MUST limit the increase to the congestion window
to at most one maximum datagram size for each congestion window that is
acknowledged.

The sender exits congestion avoidance and enters a recovery period when a
packet is lost or when the ECN-CE count reported by its peer increases.

## Ignoring Loss of Undecryptable Packets

During the handshake, some packet protection keys might not be available when
a packet arrives, and the receiver can choose to drop the packet. In particular,
Handshake and 0-RTT packets cannot be processed until the Initial packets
arrive, and 1-RTT packets cannot be processed until the handshake completes.
Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might
have arrived before the peer had packet protection keys to process those
packets. Endpoints MUST NOT ignore the loss of packets that were sent after
the earliest acknowledged packet in a given packet number space.

## Probe Timeout

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

## Persistent Congestion {#persistent-congestion}

When a sender establishes loss of all packets sent over a long enough duration,
the network is considered to be experiencing persistent congestion.

### Duration {#pc-duration}

The persistent congestion duration is computed as follows:

~~~pseudocode
(smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay) *
    kPersistentCongestionThreshold
~~~

Unlike the PTO computation in {{pto}}, this duration includes the max_ack_delay
irrespective of the packet number spaces in which losses are established.

This duration allows a sender to send as many packets before establishing
persistent congestion, including some in response to PTO expiration, as TCP does
with Tail Loss Probes {{?RFC8985}} and an RTO {{?RFC5681}}.

Larger values of kPersistentCongestionThreshold cause the sender to become less
responsive to persistent congestion in the network, which can result in
aggressive sending into a congested network. Too small a value can result in a
sender declaring persistent congestion unnecessarily, resulting in reduced
throughput for the sender.

The RECOMMENDED value for kPersistentCongestionThreshold is 3, which results in
behavior that is approximately equivalent to a TCP sender declaring an RTO after
two TLPs.

This design does not use consecutive PTO events to establish persistent
congestion, since application patterns impact PTO expiration. For example, a
sender that sends small amounts of data with silence periods between them
restarts the PTO timer every time it sends, potentially preventing the PTO timer
from expiring for a long period of time, even when no acknowledgments are being
received. The use of a duration enables a sender to establish persistent
congestion without depending on PTO expiration.

### Establishing Persistent Congestion

A sender establishes persistent congestion after the receipt of an
acknowledgment if two packets that are ack-eliciting are declared lost, and:

* across all packet number spaces, none of the packets sent between the send
  times of these two packets are acknowledged;

* the duration between the send times of these two packets exceeds the
  persistent congestion duration ({{pc-duration}}); and

* a prior RTT sample existed when these two packets were sent.

These two packets MUST be ack-eliciting, since a receiver is required to
acknowledge only ack-eliciting packets within its maximum acknowledgment delay;
see {{Section 13.2 of QUIC-TRANSPORT}}.

The persistent congestion period SHOULD NOT start until there is at least one
RTT sample. Before the first RTT sample, a sender arms its PTO timer based on
the initial RTT ({{pto-handshake}}), which could be substantially larger than
the actual RTT. Requiring a prior RTT sample prevents a sender from establishing
persistent congestion with potentially too few probes.

Since network congestion is not affected by packet number spaces, persistent
congestion SHOULD consider packets sent across packet number spaces. A sender
that does not have state for all packet number spaces or an implementation that
cannot compare send times across packet number spaces MAY use state for just the
packet number space that was acknowledged. This might result in erroneously
declaring persistent congestion, but it will not lead to a failure to detect
persistent congestion.

When persistent congestion is declared, the sender's congestion window MUST be
reduced to the minimum congestion window (kMinimumWindow), similar to a TCP
sender's response on an RTO {{RFC5681}}.

### Example

The following example illustrates how a sender might establish persistent
congestion. Assume:

~~~pseudocode
smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay = 2
kPersistentCongestionThreshold = 3
~~~

Consider the following sequence of events:

| Time   |              Action               |
|:-------|:----------------------------------|
| t=0    | Send packet #1 (application data) |
| t=1    | Send packet #2 (application data) |
| t=1.2  | Receive acknowledgment of #1      |
| t=2    | Send packet #3 (application data) |
| t=3    | Send packet #4 (application data) |
| t=4    | Send packet #5 (application data) |
| t=5    | Send packet #6 (application data) |
| t=6    | Send packet #7 (application data) |
| t=8    | Send packet #8 (PTO 1)            |
| t=12   | Send packet #9 (PTO 2)            |
| t=12.2 | Receive acknowledgment of #9      |

Packets 2 through 8 are declared lost when the acknowledgment for packet 9 is
received at `t = 12.2`.

The congestion period is calculated as the time between the oldest and newest
lost packets: `8 - 1 = 7`.  The persistent congestion duration is `2 * 3 = 6`.
Because the threshold was reached and because none of the packets between the
oldest and the newest lost packets were acknowledged, the network is considered
to have experienced persistent congestion.

While this example shows PTO expiration, they are not required for persistent
congestion to be established.


## Pacing {#pacing}

A sender SHOULD pace sending of all in-flight packets based on input from the
congestion controller.

Sending multiple packets into the network without any delay between them creates
a packet burst that might cause short-term congestion and losses. Senders MUST
either use pacing or limit such bursts. Senders SHOULD limit bursts to the
initial congestion window; see {{initial-cwnd}}. A sender with knowledge that
the network path to the receiver can absorb larger bursts MAY use a higher
limit.

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. To avoid
delaying their delivery to the peer, packets containing only ACK frames SHOULD
therefore not be paced.

Endpoints can implement pacing as they choose. A perfectly paced sender spreads
packets exactly evenly over time. For a window-based congestion controller, such
as the one in this document, that rate can be computed by averaging the
congestion window over the RTT. Expressed as a rate in units of
bytes per time, where congestion_window is in bytes:

~~~pseudocode
rate = N * congestion_window / smoothed_rtt
~~~

Or expressed as an inter-packet interval in units of time:

~~~pseudocode
interval = ( smoothed_rtt * packet_size / congestion_window ) / N
~~~

Using a value for `N` that is small, but at least 1 (for example, 1.25) ensures
that variations in RTT do not result in underutilization of the
congestion window.

Practical considerations, such as packetization, scheduling delays, and
computational efficiency, can cause a sender to deviate from this rate over time
periods that are much shorter than an RTT.

One possible implementation strategy for pacing uses a leaky bucket algorithm,
where the capacity of the "bucket" is limited to the maximum burst size and the
rate the "bucket" fills is determined by the above function.

## Underutilizing the Congestion Window

When bytes in flight is smaller than the congestion window and sending is not
pacing limited, the congestion window is underutilized. This can happen due to
insufficient application data or flow control limits. When this occurs,
the congestion window SHOULD NOT be increased in either slow start or
congestion avoidance.

A sender that paces packets (see {{pacing}}) might delay sending packets
and not fully utilize the congestion window due to this delay. A sender
SHOULD NOT consider itself application limited if it would have fully
utilized the congestion window without pacing delay.

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


# Security Considerations

## Loss and Congestion Signals

Loss detection and congestion control fundamentally involve the consumption of
signals, such as delay, loss, and ECN markings, from unauthenticated
entities. An attacker can cause endpoints to reduce their sending rate by
manipulating these signals: by dropping packets, by altering path delay
strategically, or by changing ECN codepoints.

## Traffic Analysis

Packets that carry only ACK frames can be heuristically identified by observing
packet size.  Acknowledgment patterns may expose information about link
characteristics or application behavior.  To reduce leaked information,
endpoints can bundle acknowledgments with other frames, or they can use PADDING
frames at a potential cost to performance.

## 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 can detect suppression of reports by marking occasional packets that it
sends with an ECN-CE marking. If a packet sent with an ECN-CE marking is not
reported as having been CE marked when the packet is acknowledged, then the
sender can disable ECN for that path by not setting ECN-Capable Transport (ECT)
codepoints in subsequent packets sent on that path {{!RFC3168}}.

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. Congestion controllers
respond to reports of ECN-CE by reducing their rate, but the response may vary.
Markings can be treated as equivalent to loss {{!RFC3168}}, but other
responses can be specified, such as {{?RFC8511}} or {{?RFC8311}}.


--- back

# Loss Recovery Pseudocode

We now describe an example implementation of the loss detection mechanisms
described in {{loss-detection}}.

The pseudocode segments in this section are licensed as Code Components; see the
copyright notice.

## 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, the endpoint can still maintain state for it
for an amount of time to allow for packet reordering; see {{Section 13.3 of
QUIC-TRANSPORT}}. This enables a sender to detect 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 acknowledgment will be received,
  though the peer could delay sending the ACK frame containing it
  by up to the max_ack_delay.

in_flight:
: A Boolean that indicates whether the packet counts toward bytes in
  flight.

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.


## Constants of Interest {#constants-of-interest}

Constants used in loss recovery are based on a combination of RFCs, papers, and
common practice.

kPacketThreshold:
: Maximum reordering in packets before packet threshold loss detection
  considers a packet lost. The value recommended in {{packet-threshold}} is 3.

kTimeThreshold:

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

kGranularity:

: Timer granularity. This is a system-dependent value, and {{time-threshold}}
  recommends a value of 1 ms.

kInitialRtt:
: The RTT used before an RTT sample is taken. The value recommended in
{{pto-handshake}} is 333 ms.

kPacketNumberSpace:
: An enum to enumerate the three packet number spaces:

~~~
enum kPacketNumberSpace {
  Initial,
  Handshake,
  ApplicationData,
}
~~~

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

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

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

smoothed_rtt:
: The smoothed RTT of the connection, computed as described in
  {{smoothed-rtt}}.

rttvar:
: The RTT variation, computed as described in {{smoothed-rtt}}.

min_rtt:
: The minimum RTT seen over a period of time, ignoring acknowledgment delay, as
  described in {{min-rtt}}.

first_rtt_sample:
: The time that the first RTT sample was obtained.

max_ack_delay:
: The maximum amount of time by which the receiver intends to delay
  acknowledgments for packets in the Application Data packet number
  space, as defined by the eponymous transport parameter ({{Section 18.2
  of QUIC-TRANSPORT}}). Note that the actual ack_delay in a received
  ACK frame may be larger due to late timers, reordering, or loss.

loss_detection_timer:
: Multi-modal timer used for loss detection.

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

time_of_last_ack_eliciting_packet\[kPacketNumberSpace]:
: The time the most recent ack-eliciting packet was sent.

largest_acked_packet\[kPacketNumberSpace]:
: The largest packet number acknowledged in the packet number space so far.

loss_time\[kPacketNumberSpace]:
: The time at which the next packet in that packet number space can be
  considered lost based on exceeding the reordering window in time.

sent_packets\[kPacketNumberSpace]:
: An association of packet numbers in a packet number space to information
  about them.  Described in detail above in {{tracking-sent-packets}}.


## Initialization

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

~~~pseudocode
loss_detection_timer.reset()
pto_count = 0
latest_rtt = 0
smoothed_rtt = kInitialRtt
rttvar = kInitialRtt / 2
min_rtt = 0
first_rtt_sample = 0
for pn_space in [ Initial, Handshake, ApplicationData ]:
  largest_acked_packet[pn_space] = infinite
  time_of_last_ack_eliciting_packet[pn_space] = 0
  loss_time[pn_space] = 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:

~~~pseudocode
OnPacketSent(packet_number, pn_space, ack_eliciting,
             in_flight, sent_bytes):
  sent_packets[pn_space][packet_number].packet_number =
                                           packet_number
  sent_packets[pn_space][packet_number].time_sent = now()
  sent_packets[pn_space][packet_number].ack_eliciting =
                                           ack_eliciting
  sent_packets[pn_space][packet_number].in_flight = in_flight
  sent_packets[pn_space][packet_number].sent_bytes = sent_bytes
  if (in_flight):
    if (ack_eliciting):
      time_of_last_ack_eliciting_packet[pn_space] = now()
    OnPacketSentCC(sent_bytes)
    SetLossDetectionTimer()
~~~

## On Receiving a Datagram

When a server is blocked by anti-amplification limits, receiving
a datagram unblocks it, even if none of the packets in the
datagram are successfully processed. In such a case, the PTO
timer will need to be rearmed.

Pseudocode for OnDatagramReceived follows:

~~~pseudocode
OnDatagramReceived(datagram):
  // If this datagram unblocks the server, arm the
  // PTO timer to avoid deadlock.
  if (server was at anti-amplification limit):
    SetLossDetectionTimer()
    if loss_detection_timer.timeout < now():
      // Execute PTO if it would have expired
      // while the amplification limit applied.
      OnLossDetectionTimeout()
~~~

## On Receiving an Acknowledgment

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

Pseudocode for OnAckReceived and UpdateRtt follow:

~~~pseudocode
IncludesAckEliciting(packets):
  for packet in packets:
    if (packet.ack_eliciting):
      return true
  return false

OnAckReceived(ack, pn_space):
  if (largest_acked_packet[pn_space] == infinite):
    largest_acked_packet[pn_space] = ack.largest_acked
  else:
    largest_acked_packet[pn_space] =
        max(largest_acked_packet[pn_space], ack.largest_acked)

  // DetectAndRemoveAckedPackets finds packets that are newly
  // acknowledged and removes them from sent_packets.
  newly_acked_packets =
      DetectAndRemoveAckedPackets(ack, pn_space)
  // Nothing to do if there are no newly acked packets.
  if (newly_acked_packets.empty()):
    return

  // Update the RTT if the largest acknowledged is newly acked
  // and at least one ack-eliciting was newly acked.
  if (newly_acked_packets.largest().packet_number ==
          ack.largest_acked &&
      IncludesAckEliciting(newly_acked_packets)):
    latest_rtt =
      now() - newly_acked_packets.largest().time_sent
    UpdateRtt(ack.ack_delay)

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

  lost_packets = DetectAndRemoveLostPackets(pn_space)
  if (!lost_packets.empty()):
    OnPacketsLost(lost_packets)
  OnPacketsAcked(newly_acked_packets)

  // Reset pto_count unless the client is unsure if
  // the server has validated the client's address.
  if (PeerCompletedAddressValidation()):
    pto_count = 0
  SetLossDetectionTimer()


UpdateRtt(ack_delay):
  if (first_rtt_sample == 0):
    min_rtt = latest_rtt
    smoothed_rtt = latest_rtt
    rttvar = latest_rtt / 2
    first_rtt_sample = now()
    return

  // min_rtt ignores acknowledgment delay.
  min_rtt = min(min_rtt, latest_rtt)
  // Limit ack_delay by max_ack_delay after handshake
  // confirmation.
  if (handshake confirmed):
    ack_delay = min(ack_delay, max_ack_delay)

  // Adjust for acknowledgment delay if plausible.
  adjusted_rtt = latest_rtt
  if (latest_rtt >= min_rtt + ack_delay):
    adjusted_rtt = latest_rtt - ack_delay

  rttvar = 3/4 * rttvar + 1/4 * abs(smoothed_rtt - adjusted_rtt)
  smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * adjusted_rtt
~~~

## 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 fire immediately.

Pseudocode for SetLossDetectionTimer follows (where the "^" operator represents
exponentiation):

~~~pseudocode
GetLossTimeAndSpace():
  time = loss_time[Initial]
  space = Initial
  for pn_space in [ Handshake, ApplicationData ]:
    if (time == 0 || loss_time[pn_space] < time):
      time = loss_time[pn_space];
      space = pn_space
  return time, space

GetPtoTimeAndSpace():
  duration = (smoothed_rtt + max(4 * rttvar, kGranularity))
      * (2 ^ pto_count)
  // Anti-deadlock PTO starts from the current time
  if (no ack-eliciting packets in flight):
    assert(!PeerCompletedAddressValidation())
    if (has handshake keys):
      return (now() + duration), Handshake
    else:
      return (now() + duration), Initial
  pto_timeout = infinite
  pto_space = Initial
  for space in [ Initial, Handshake, ApplicationData ]:
    if (no ack-eliciting packets in flight in space):
        continue;
    if (space == ApplicationData):
      // Skip Application Data until handshake confirmed.
      if (handshake is not confirmed):
        return pto_timeout, pto_space
      // Include max_ack_delay and backoff for Application Data.
      duration += max_ack_delay * (2 ^ pto_count)

    t = time_of_last_ack_eliciting_packet[space] + duration
    if (t < pto_timeout):
      pto_timeout = t
      pto_space = space
  return pto_timeout, pto_space

PeerCompletedAddressValidation():
  // Assume clients validate the server's address implicitly.
  if (endpoint is server):
    return true
  // Servers complete address validation when a
  // protected packet is received.
  return has received Handshake ACK ||
       handshake confirmed

SetLossDetectionTimer():
  earliest_loss_time, _ = GetLossTimeAndSpace()
  if (earliest_loss_time != 0):
    // Time threshold loss detection.
    loss_detection_timer.update(earliest_loss_time)
    return

  if (server is at anti-amplification limit):
    // The server's timer is not set if nothing can be sent.
    loss_detection_timer.cancel()
    return

  if (no ack-eliciting packets in flight &&
      PeerCompletedAddressValidation()):
    // There is nothing to detect lost, so no timer is set.
    // However, the client needs to arm the timer if the
    // server might be blocked by the anti-amplification limit.
    loss_detection_timer.cancel()
    return

  timeout, _ = GetPtoTimeAndSpace()
  loss_detection_timer.update(timeout)
~~~


## On Timeout

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

Pseudocode for OnLossDetectionTimeout follows:

~~~pseudocode
OnLossDetectionTimeout():
  earliest_loss_time, pn_space = GetLossTimeAndSpace()
  if (earliest_loss_time != 0):
    // Time threshold loss Detection
    lost_packets = DetectAndRemoveLostPackets(pn_space)
    assert(!lost_packets.empty())
    OnPacketsLost(lost_packets)
    SetLossDetectionTimer()
    return

  if (no ack-eliciting packets in flight):
    assert(!PeerCompletedAddressValidation())
    // Client sends an anti-deadlock packet: Initial is padded
    // to earn more anti-amplification credit,
    // a Handshake packet proves address ownership.
    if (has Handshake keys):
      SendOneAckElicitingHandshakePacket()
    else:
      SendOneAckElicitingPaddedInitialPacket()
  else:
    // PTO. Send new data if available, else retransmit old data.
    // If neither is available, send a single PING frame.
    _, pn_space = GetPtoTimeAndSpace()
    SendOneOrTwoAckElicitingPackets(pn_space)

  pto_count++
  SetLossDetectionTimer()
~~~


## Detecting Lost Packets

DetectAndRemoveLostPackets is called every time an ACK is received or the time
threshold loss detection timer expires. This function operates on the
sent_packets for that packet number space and returns a list of packets newly
detected as lost.

Pseudocode for DetectAndRemoveLostPackets follows:

~~~pseudocode
DetectAndRemoveLostPackets(pn_space):
  assert(largest_acked_packet[pn_space] != infinite)
  loss_time[pn_space] = 0
  lost_packets = []
  loss_delay = kTimeThreshold * max(latest_rtt, smoothed_rtt)

  // Minimum time of kGranularity before packets are deemed lost.
  loss_delay = max(loss_delay, kGranularity)

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

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

    // Mark packet as lost, or set time when it should be marked.
    // Note: The use of kPacketThreshold here assumes that there
    // were no sender-induced gaps in the packet number space.
    if (unacked.time_sent <= lost_send_time ||
        largest_acked_packet[pn_space] >=
          unacked.packet_number + kPacketThreshold):
      sent_packets[pn_space].remove(unacked.packet_number)
      lost_packets.insert(unacked)
    else:
      if (loss_time[pn_space] == 0):
        loss_time[pn_space] = unacked.time_sent + loss_delay
      else:
        loss_time[pn_space] = min(loss_time[pn_space],
                                  unacked.time_sent + loss_delay)
  return lost_packets
~~~


## Upon Dropping Initial or Handshake Keys

When Initial or Handshake keys are discarded, packets from the space
are discarded and loss detection state is updated.

Pseudocode for OnPacketNumberSpaceDiscarded follows:

~~~pseudocode
OnPacketNumberSpaceDiscarded(pn_space):
  assert(pn_space != ApplicationData)
  RemoveFromBytesInFlight(sent_packets[pn_space])
  sent_packets[pn_space].clear()
  // Reset the loss detection and PTO timer
  time_of_last_ack_eliciting_packet[pn_space] = 0
  loss_time[pn_space] = 0
  pto_count = 0
  SetLossDetectionTimer()
~~~


# Congestion Control Pseudocode

We now describe an example implementation of the congestion controller described
in {{congestion-control}}.

The pseudocode segments in this section are licensed as Code Components; see the
copyright notice.

## Constants of Interest {#cc-consts-of-interest}

Constants used in congestion control are based on a combination of RFCs, papers,
and common practice.

kInitialWindow:
: Default limit on the initial bytes in flight as described in {{initial-cwnd}}.

kMinimumWindow:
: Minimum congestion window in bytes as described in {{initial-cwnd}}.

kLossReductionFactor:
: Scaling factor applied to reduce the congestion window when a new loss event
  is detected. {{congestion-control}} recommends a value of 0.5.

kPersistentCongestionThreshold:
: Period of time for persistent congestion to be established, specified as a PTO
  multiplier. {{persistent-congestion}} recommends a value of 3.


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

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

max_datagram_size:
: The sender's current maximum payload size. This does not include UDP or IP
  overhead.  The max datagram size is used for congestion window
  computations. An endpoint sets the value of this variable based on its Path
  Maximum Transmission Unit (PMTU; see {{Section 14.2 of QUIC-TRANSPORT}}), with
  a minimum value of 1200 bytes.

ecn_ce_counters\[kPacketNumberSpace]:
: The highest value reported for the ECN-CE counter in the packet number space
  by the peer in an ACK frame. This value 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 acknowledged or declared
  lost. The size does not include IP or UDP overhead, but does include the QUIC
  header and Authenticated Encryption with Associated Data (AEAD) overhead.
  Packets only containing ACK frames do not count toward bytes_in_flight to
  ensure congestion control does not impede congestion feedback.

congestion_window:
: Maximum number of bytes allowed to be in flight.

congestion_recovery_start_time:
: The time the current recovery period started due to the detection of loss
  or ECN. When a packet sent after this time is acknowledged, QUIC exits
  congestion recovery.

ssthresh:
: Slow start threshold in bytes.  When the congestion window is below ssthresh,
  the mode is slow start and the window grows by the number of bytes
  acknowledged.

The congestion control pseudocode also accesses some of the variables from the
loss recovery pseudocode.

## Initialization

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

~~~pseudocode
congestion_window = kInitialWindow
bytes_in_flight = 0
congestion_recovery_start_time = 0
ssthresh = infinite
for pn_space in [ Initial, Handshake, ApplicationData ]:
  ecn_ce_counters[pn_space] = 0
~~~


## On Packet Sent

Whenever a packet is sent and it contains non-ACK frames, the packet
increases bytes_in_flight.

~~~pseudocode
OnPacketSentCC(sent_bytes):
  bytes_in_flight += sent_bytes
~~~


## On Packet Acknowledgment

This is invoked from loss detection's OnAckReceived and is supplied with the
newly acked_packets from sent_packets.

In congestion avoidance, implementers that use an integer representation
for congestion_window should be careful with division and can use
the alternative approach suggested in {{Section 2.1 of RFC3465}}.

~~~pseudocode
InCongestionRecovery(sent_time):
  return sent_time <= congestion_recovery_start_time

OnPacketsAcked(acked_packets):
  for acked_packet in acked_packets:
    OnPacketAcked(acked_packet)

OnPacketAcked(acked_packet):
  if (!acked_packet.in_flight):
    return;
  // Remove from bytes_in_flight.
  bytes_in_flight -= acked_packet.sent_bytes
  // Do not increase congestion_window if application
  // limited or flow control limited.
  if (IsAppOrFlowControlLimited())
    return
  // Do not increase congestion window in recovery period.
  if (InCongestionRecovery(acked_packet.time_sent)):
    return
  if (congestion_window < ssthresh):
    // Slow start.
    congestion_window += acked_packet.sent_bytes
  else:
    // Congestion avoidance.
    congestion_window +=
      max_datagram_size * acked_packet.sent_bytes
      / congestion_window
~~~


## On New Congestion Event

This is invoked from ProcessECN and OnPacketsLost when a new congestion event is
detected. If not already in recovery, this starts a recovery period and
reduces the slow start threshold and congestion window immediately.

~~~pseudocode
OnCongestionEvent(sent_time):
  // No reaction if already in a recovery period.
  if (InCongestionRecovery(sent_time)):
    return

  // Enter recovery period.
  congestion_recovery_start_time = now()
  ssthresh = congestion_window * kLossReductionFactor
  congestion_window = max(ssthresh, kMinimumWindow)
  // A packet can be sent to speed up loss recovery.
  MaybeSendOnePacket()
~~~


## Process ECN Information

This is invoked when an ACK frame with an ECN section is received from the peer.

~~~pseudocode
ProcessECN(ack, pn_space):
  // If the ECN-CE counter reported by the peer has increased,
  // this could be a new congestion event.
  if (ack.ce_counter > ecn_ce_counters[pn_space]):
    ecn_ce_counters[pn_space] = ack.ce_counter
    sent_time = sent_packets[ack.largest_acked].time_sent
    OnCongestionEvent(sent_time)
~~~


## On Packets Lost

This is invoked when DetectAndRemoveLostPackets deems packets lost.

~~~pseudocode
OnPacketsLost(lost_packets):
  sent_time_of_last_loss = 0
  // Remove lost packets from bytes_in_flight.
  for lost_packet in lost_packets:
    if lost_packet.in_flight:
      bytes_in_flight -= lost_packet.sent_bytes
      sent_time_of_last_loss =
        max(sent_time_of_last_loss, lost_packet.time_sent)
  // Congestion event if in-flight packets were lost
  if (sent_time_of_last_loss != 0):
    OnCongestionEvent(sent_time_of_last_loss)

  // Reset the congestion window if the loss of these
  // packets indicates persistent congestion.
  // Only consider packets sent after getting an RTT sample.
  if (first_rtt_sample == 0):
    return
  pc_lost = []
  for lost in lost_packets:
    if lost.time_sent > first_rtt_sample:
      pc_lost.insert(lost)
  if (InPersistentCongestion(pc_lost)):
    congestion_window = kMinimumWindow
    congestion_recovery_start_time = 0
~~~


## Removing Discarded Packets from Bytes in Flight

When Initial or Handshake keys are discarded, packets sent in that space no
longer count toward bytes in flight.

Pseudocode for RemoveFromBytesInFlight follows:

~~~pseudocode
RemoveFromBytesInFlight(discarded_packets):
  // Remove any unacknowledged packets from flight.
  foreach packet in discarded_packets:
    if packet.in_flight
      bytes_in_flight -= size
~~~


# Contributors
{: numbered="false"}

The IETF QUIC Working Group received an enormous amount of support from many
people. The following people provided substantive contributions to this
document:

<ul spacing="compact">
<li><t><contact fullname="Alessandro Ghedini"/></t></li>
<li><t><contact fullname="Benjamin Saunders"/></t></li>
<li><t><contact fullname="Gorry Fairhurst"/></t></li>
<li><t><contact asciiFullname="Kazu Yamamoto" fullname="山本和彦"/></t></li>
<li><t><contact asciiFullname="Kazuho Oku" fullname="奥 一穂"/></t></li>
<li><t><contact fullname="Lars Eggert"/></t></li>
<li><t><contact fullname="Magnus Westerlund"/></t></li>
<li><t><contact fullname="Marten Seemann"/></t></li>
<li><t><contact fullname="Martin Duke"/></t></li>
<li><t><contact fullname="Martin Thomson"/></t></li>
<li><t><contact fullname="Mirja Kühlewind"/></t></li>
<li><t><contact fullname="Nick Banks"/></t></li>
<li><t><contact fullname="Praveen Balasubramanian"/></t></li>
</ul>