QUIC Loss Detection and Congestion ControlFastlyjri.ietf@gmail.comGoogleianswett@google.com
Transport
QUICThis document describes loss detection and congestion control mechanisms for
QUIC.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.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 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.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
when, and only when, they appear in all capitals, as shown here.Definitions of terms that are used in this document:
ACK frames refer to both ACK and ACK_ECN frames in this
document.
Any packet containing only an ACK or ACK_ECN frame.
Packets are considered in-flight when they have been sent
and neither acknowledged nor declared lost, and they are not
ACK-only.
All frames besides ACK, ACK_ECN, or PADDING are considered
retransmittable.
Packets that contain retransmittable frames elicit an ACK from
the receiver and are called retransmittable packets.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 . 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 only ACK,
ACK_ECN, and PADDING frames are not acknowledged immediately.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 and ACK_ECN 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.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.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.TCP conflates transmission sequence number at the sender with delivery sequence
number at the receiver, which results in retransmissions of the same data
carrying the same sequence number, and consequently to problems caused by
“retransmission ambiguity”. QUIC separates the two: QUIC uses a packet number
for transmissions, and any application data is sent in one or more streams,
with delivery order determined by stream offsets encoded within STREAM frames.QUIC’s packet number is strictly increasing, and directly encodes transmission
order. A higher QUIC packet number signifies that the packet was sent later,
and a lower QUIC packet number signifies that the packet was sent earlier. When
a packet containing frames is deemed lost, QUIC rebundles necessary frames in a
new packet with a new packet number, removing ambiguity about which packet is
acknowledged when an ACK is received. Consequently, more accurate RTT
measurements can be made, spurious retransmissions are trivially detected, and
mechanisms such as Fast Retransmit can be applied universally, based only on
packet number.This design point significantly simplifies loss detection mechanisms for QUIC.
Most TCP mechanisms implicitly attempt to infer transmission ordering based on
TCP sequence numbers - a non-trivial task, especially when TCP timestamps are
not available.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.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.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.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.RTT is calculated when an ACK frame arrives by computing the difference between
the current time and the time the largest newly acked packet was sent. If no
packets are newly acknowledged, RTT cannot be calculated. When RTT is
calculated, the ack delay field from the ACK frame SHOULD be subtracted from the
RTT as long as the result is larger than the Min RTT. If the result is smaller
than the min_rtt, the RTT should be used, but the ack delay field should be
ignored.Like TCP, QUIC calculates both smoothed RTT and RTT variance similar to those
specified in .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.QUIC is able to explicitly model delay at the receiver via the ack delay
field in the ACK frame. Therefore, QUIC diverges from TCP by calculating a
MaxAckDelay dynamically, instead of assuming a constant delayed ack timeout
for all connections.MaxAckDelay is the maximum ack delay supplied in an all incoming ACK frames.
MaxAckDelay excludes ack delays that aren’t included in an RTT sample because
they’re too large or the largest acknowledged has already been acknowledged.
MaxAckDelay also excludes ack delays where the largest acknowledged references
an ACK-only packet.Ack-based loss detection implements the spirit of TCP’s Fast Retransmit
, Early Retransmit , FACK, and SACK loss recovery
. This section provides an overview of how these algorithms are
implemented in QUIC.An unacknowledged packet is marked as lost when an acknowledgment is received
for a packet that was sent a threshold number of packets (kReorderingThreshold)
and/or a threshold amount of time after the unacknowledged packet. Receipt of
the acknowledgement indicates that a later packet was received, while the
reordering threshold provides some tolerance for reordering of packets in the
network.The RECOMMENDED initial value for kReorderingThreshold is 3, based on
TCP loss recovery . Some networks may exhibit higher
degrees of reordering, causing a sender to detect spurious losses. Spuriously
declaring packets lost leads to unnecessary retransmissions and may result in
degraded performance due to the actions of the congestion controller upon
detecting loss. Implementers MAY use algorithms developed for TCP, such as
TCP-NCR , to improve QUIC’s reordering resilience.QUIC implementations can use time-based loss detection to handle reordering
based on time elapsed since the packet was sent. This may be used either as
a replacement for a packet reordering threshold or in addition to it.
The RECOMMENDED time threshold, expressed as a fraction of the
round-trip time (kTimeReorderingFraction), is 1/8.Unacknowledged packets close to the tail may have fewer than
kReorderingThreshold retransmittable packets sent after them. Loss of such
packets cannot be detected via Fast Retransmit. To enable ack-based loss
detection of such packets, receipt of an acknowledgment for the last outstanding
retransmittable packet triggers the Early Retransmit process, as follows.If there are unacknowledged in-flight packets still pending, they should
be marked as lost. To compensate for the reduced reordering resilience, the
sender SHOULD set a timer for a small period of time. If the unacknowledged
in-flight packets are not acknowledged during this time, then these
packets MUST be marked as lost.An endpoint SHOULD set the timer such that a packet is marked as lost no earlier
than 1.125 * max(SRTT, latest_RTT) since when it was sent.Using max(SRTT, latest_RTT) protects from the two following cases:the latest RTT sample is lower than the SRTT, perhaps due to reordering where
packet whose ack triggered the Early Retransit process encountered a shorter
path;the latest RTT sample is higher than the SRTT, perhaps due to a sustained
increase in the actual RTT, but the smoothed SRTT has not yet caught up.The 1.125 multiplier increases reordering resilience. Implementers MAY
experiment with using other multipliers, bearing in mind that a lower multiplier
reduces reordering resilience and increases spurious retransmissions, and a
higher multipler increases loss recovery delay.This mechanism is based on Early Retransmit for TCP . However,
does not include the timer described above. Early Retransmit is
prone to spurious retransmissions due to its reduced reordering resilence
without the timer. This observation led Linux TCP implementers to implement a
timer for TCP as well, and this document incorporates this advancement.Timer-based loss detection recovers from losses that cannot be handled by
ack-based loss detection. It uses a single timer which switches between
a handshake retransmission timer, a Tail Loss Probe timer and
Retransmission Timeout mechanisms.Data in CRYPTO frames is critical to QUIC transport and crypto negotiation, so a
more aggressive timeout is used to retransmit it. Below, the term “handshake
packet” is used to refer to packets containing CRYPTO frames, not packets with
the specific long header packet type Handshake.The initial handshake timeout SHOULD be set to twice the initial RTT.At the beginning, there are no prior RTT samples within a connection. Resumed
connections over the same network SHOULD use the previous connection’s final
smoothed RTT value as the resumed connection’s initial RTT.If no previous RTT is available, or if the network changes, the initial RTT
SHOULD be set to 100ms.When CRYPTO frames are sent, the sender SHOULD set a timer for the handshake
timeout period. Upon timeout, the sender MUST retransmit all unacknowledged
CRYPTO data by calling RetransmitAllUnackedHandshakeData(). On each
consecutive expiration of the handshake timer without receiving an
acknowledgement for a new packet, the sender SHOULD double the handshake timeout
and set a timer for this period.When CRYPTO frames are outstanding, the TLP and RTO timers are not active unless
the CRYPTO frames were sent at 1RTT encryption.When an acknowledgement is received for a handshake packet, the new RTT is
computed and the timer SHOULD be set for twice the newly computed smoothed RTT.A Retry or Version Negotiation packet causes a client to send another Initial
packet, effectively restarting the connection process.Either packet indicates that the Initial was received but not processed.
Neither packet can be treated as an acknowledgment for the Initial, but they MAY
be used to improve the RTT estimate.The algorithm described in this section is an adaptation of the Tail Loss Probe
algorithm proposed for TCP .A packet sent at the tail is particularly vulnerable to slow loss detection,
since acks of subsequent packets are needed to trigger ack-based detection. To
ameliorate this weakness of tail packets, the sender schedules a timer when the
last retransmittable packet before quiescence is transmitted. Upon timeout,
a Tail Loss Probe (TLP) packet is sent to evoke an acknowledgement from the
receiver.The timer duration, or Probe Timeout (PTO), is set based on the following
conditions:PTO SHOULD be scheduled for max(1.5*SRTT+MaxAckDelay, kMinTLPTimeout)If RTO () is earlier, schedule a TLP in its place. That is,
PTO SHOULD be scheduled for min(RTO, PTO).QUIC includes MaxAckDelay in all probe timeouts, because it assumes the ack
delay may come into play, regardless of the number of packets outstanding.
TCP’s TLP assumes if at least 2 packets are outstanding, acks will not be
delayed.A PTO value of at least 1.5*SRTT ensures that the ACK is overdue. The 1.5 is
based on , but implementations MAY experiment with other constants.To reduce latency, it is RECOMMENDED that the sender set and allow the TLP timer
to fire twice before setting an RTO timer. In other words, when the TLP timer
expires the first time, a TLP packet is sent, and it is RECOMMENDED that the TLP
timer be scheduled for a second time. When the TLP timer expires the second
time, a second TLP packet is sent, and an RTO timer SHOULD be scheduled .A TLP packet SHOULD carry new data when possible. If new data is unavailable or
new data cannot be sent due to flow control, a TLP packet MAY retransmit
unacknowledged data to potentially reduce recovery time. Since a TLP timer is
used to send a probe into the network prior to establishing any packet loss,
prior unacknowledged packets SHOULD NOT be marked as lost when a TLP timer
expires.A sender may not know that a packet being sent is a tail packet. Consequently,
a sender may have to arm or adjust the TLP timer on every sent retransmittable
packet.A Retransmission Timeout (RTO) timer is the final backstop for loss
detection. The algorithm used in QUIC is based on the RTO algorithm for TCP
and is additionally resilient to spurious RTO events .When the last TLP packet is sent, a timer is set for the RTO period. When
this timer expires, the sender sends two packets, to evoke acknowledgements from
the receiver, and restarts the RTO timer.Similar to TCP , the RTO period is set based on the following
conditions:When the final TLP packet is sent, the RTO period is set to max(SRTT +
4*RTTVAR + MaxAckDelay, kMinRTOTimeout)When an RTO timer expires, the RTO period is doubled.The sender typically has incurred a high latency penalty by the time an RTO
timer expires, and this penalty increases exponentially in subsequent
consecutive RTO events. Sending a single packet on an RTO event therefore makes
the connection very sensitive to single packet loss. Sending two packets instead
of one significantly increases resilience to packet drop in both directions,
thus reducing the probability of consecutive RTO events.QUIC’s RTO algorithm differs from TCP in that the firing of an RTO timer is not
considered a strong enough signal of packet loss, so does not result in an
immediate change to congestion window or recovery state. An RTO timer expires
only when there’s a prolonged period of network silence, which could be caused
by a change in the underlying network RTT.QUIC also diverges from TCP by including MaxAckDelay in the RTO period. Since
QUIC corrects for this delay in its SRTT and RTTVAR computations, it is
necessary to add this delay explicitly in the TLP and RTO computation.When an acknowledgment is received for a packet sent on an RTO event, any
unacknowledged packets with lower packet numbers than those acknowledged MUST be
marked as lost.A packet sent when an RTO timer expires MAY carry new data if available or
unacknowledged data to potentially reduce recovery time. Since this packet is
sent as a probe into the network prior to establishing any packet loss, prior
unacknowledged packets SHOULD NOT be marked as lost.A packet sent on an RTO timer MUST NOT be blocked by the sender’s congestion
controller. A sender MUST however count these bytes as additional bytes in
flight, since this packet adds network load without establishing packet loss.QUIC SHOULD delay sending acknowledgements in response to packets, but MUST NOT
excessively delay acknowledgements of packets containing frames other than ACK
or ACN_ECN. Specifically, implementaions MUST attempt to enforce a maximum ack
delay to avoid causing the peer spurious timeouts. The RECOMMENDED maximum ack
delay in QUIC is 25ms.An acknowledgement MAY be sent for every second full-sized packet, as TCP does
, or may be sent less frequently, as long as the delay does not
exceed the maximum ack delay. QUIC recovery algorithms do not assume the peer
generates an acknowledgement immediately when receiving a second full-sized
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.In order to quickly complete the handshake and avoid spurious retransmissions
due to handshake timeouts, handshake packets SHOULD use a very short ack
delay, such as 1ms. ACK frames MAY be sent immediately when the crypto stack
indicates all data for that encryption level has been received.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
spuriusly retransmitting the frames it contains.Below is one recommended approach for determining what packets to include in an
ACK frame.When a packet containing an ACK frame is sent, the largest acknowledged in that
frame may be saved. When a packet containing an ACK frame is acknowledged, the
receiver can stop acknowledging packets less than or equal to the largest
acknowledged in the sent ACK frame.In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT of
reordering. In cases with ACK frame loss, this approach does not guarantee that
every acknowledgement is seen by the sender before it is no longer included in
the ACK frame. Packets could be received out of order and all subsequent ACK
frames containing them could be lost. In this case, the loss recovery algorithm
may cause spurious retransmits, but the sender will continue making forward
progress.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.
Maximum number of tail loss probes before an RTO expires.
Maximum reordering in packet number space before FACK style loss detection
considers a packet lost.
Maximum reordering in time space before time based loss detection considers
a packet lost. In fraction of an RTT.
Whether time based loss detection is in use. If false, uses FACK style
loss detection.
Minimum time in the future a tail loss probe timer may be set for.
Minimum time in the future an RTO timer may be set for.
The length of the peer’s delayed ack timer.
The RTT used before an RTT sample is taken.Variables required to implement the congestion control mechanisms
are described in this section.
Multi-modal timer used for loss detection.
The number of times all unacknowledged handshake data has been
retransmitted without receiving an ack.
The number of times a tail loss probe has been sent without
receiving an ack.
The number of times an rto has been sent without receiving an ack.
The last packet number sent prior to the first retransmission
timeout.
The time the most recent retransmittable packet was sent.
The time the most recent packet containing a CRYPTO frame was sent.
The packet number of the most recently sent packet.
The largest packet number acknowledged in an ACK frame.
The most recent RTT measurement made when receiving an ack for
a previously unacked packet.
The smoothed RTT of the connection, computed as described in
The RTT variance, computed as described in
The minimum RTT seen in the connection, ignoring ack delay.
The maximum ack delay in an incoming ACK frame for this connection.
Excludes ack delays for non-retransmittable packets and those
that create an RTT sample less than min_rtt.
The largest packet number gap between the largest acknowledged
retransmittable packet and an unacknowledged
retransmittable packet before it is declared lost.
The reordering window as a fraction of max(smoothed_rtt, latest_rtt).
The time at which the next packet will be considered lost based on early
transmit or exceeding the reordering window in time.
An association of packet numbers to information about them, including a number
field indicating the packet number, a time field indicating the time a packet
was sent, a boolean indicating whether the packet is ack-only, a boolean
indicating whether it counts towards bytes in flight, and a bytes
field indicating the packet’s size. sent_packets is ordered by packet number,
and packets remain in sent_packets until acknowledged or lost. A sent_packets
data structure is maintained per packet number space, and ACK processing only
applies to a single space.At the beginning of the connection, initialize the loss detection variables as
follows:After any packet is sent, be it a new transmission or a rebundled transmission,
the following OnPacketSent function is called. The parameters to OnPacketSent
are as follows:packet_number: The packet number of the sent packet.ack_only: A boolean that indicates whether a packet contains only
ACK or PADDING frame(s). If true, it is still expected an ack will
be received for this packet, but it is not retransmittable.in_flight: A boolean that indicates whether the packet counts towards
bytes in flight.is_handshake_packet: A boolean that indicates whether the packet contains
cryptographic handshake messages critical to the completion of the QUIC
handshake. In this version of QUIC, this includes any packet with the long
header that includes a CRYPTO frame.sent_bytes: The number of bytes sent in the packet, not including UDP or IP
overhead, but including QUIC framing overhead.Pseudocode for OnPacketSent follows:When an ACK frame is received, it may newly acknowledge any number of packets.Pseudocode for OnAckReceived and UpdateRtt follow:When a packet is acked for the first time, the following OnPacketAcked function
is called. Note that a single ACK frame may newly acknowledge several packets.
OnPacketAcked must be called once for each of these newly acked packets.OnPacketAcked takes one parameter, acked_packet, which is the struct of the
newly acked packet.If this is the first acknowledgement following RTO, check if the smallest newly
acknowledged packet is one sent by the RTO, and if so, inform congestion control
of a verified RTO, similar to F-RTO .Pseudocode for OnPacketAcked follows:QUIC loss detection uses a single timer for all timer-based loss detection. The
duration of the timer is based on the timer’s mode, which is set in the packet
and timer events further below. The function SetLossDetectionTimer defined
below shows how the single timer is set.When a connection has unacknowledged handshake data, the handshake timer is
set and when it expires, all unacknowledgedd handshake data is retransmitted.When stateless rejects are in use, the connection is considered immediately
closed once a reject is sent, so no timer is set to retransmit the reject.Version negotiation packets are always stateless, and MUST be sent once per
handshake packet that uses an unsupported QUIC version, and MAY be sent in
response to 0-RTT packets.Tail loss probes and retransmission timeouts
are a timer based mechanism to recover from cases when there are
outstanding retransmittable packets, but an acknowledgement has
not been received in a timely manner.The TLP and RTO timers are armed when there is not unacknowledged handshake
data. The TLP timer is set until the max number of TLP packets have been
sent, and then the RTO timer is set.Early retransmit is implemented with a 1/4 RTT timer. It is
part of QUIC’s time based loss detection, but is always enabled, even when
only packet reordering loss detection is enabled.Pseudocode for SetLossDetectionTimer follows:QUIC uses one loss recovery timer, which when set, can be in one of several
modes. When the timer expires, the mode determines the action to be performed.Pseudocode for OnLossDetectionTimeout follows:Packets in QUIC are only considered lost once a larger packet number in
the same packet number space is acknowledged. DetectLostPackets is called
every time an ack is received and operates on the sent_packets for that
packet number space. If the loss detection timer expires and the loss_time
is set, the previous largest acked packet is supplied.DetectLostPackets takes one parameter, acked, which is the largest acked packet.Pseudocode for DetectLostPackets follows: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.QUIC’s congestion control is based on TCP NewReno . 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 .QUIC hosts MUST NOT send packets if they would increase bytes_in_flight
(defined in ) beyond the available congestion window,
unless the packet is a probe packet sent after the TLP or RTO timer expires,
as described in and .Implementations MAY use other congestion control algorithms, and endpoints MAY
use different algorithms from one another. The signals QUIC provides for
congestion control are generic and are designed to support different algorithms.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 , endpoints are permitted to
experiment with other response functions.QUIC begins every connection in slow start and exits slow start upon loss or
upon increase in the ECN-CE counter. QUIC re-enters slow start anytime the
congestion window is less than ssthresh, which typically only occurs after an
RTO. While in slow start, QUIC increases the congestion window by the number of
bytes acknowledged when each ack is processed.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 is a period of time beginning with detection of a lost packet or an
increase in the ECN-CE counter. Because QUIC retransmits stream data and control
frames, not packets, it defines the end of recovery as a packet sent after the
start of recovery being acknowledged. This is slightly different from TCP’s
definition of recovery, which ends when the lost packet that started recovery is
acknowledged.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.A TLP packet MUST NOT be blocked by the sender’s congestion controller. The
sender MUST however count these bytes as additional bytes-in-flight, since a TLP
adds network load without establishing packet loss.Acknowledgement or loss of tail loss probes are treated like any other packet.When retransmissions are sent due to a retransmission timeout timer, no change
is made to the congestion window until the next acknowledgement arrives. The
retransmission timeout is considered spurious when this acknowledgement
acknowledges packets sent prior to the first retransmission timeout. The
retransmission timeout is considered valid when this acknowledgement
acknowledges no packets sent prior to the first retransmission timeout. In this
case, the congestion window MUST be reduced to the minimum congestion window and
slow start is re-entered.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).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.
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.
Default limit on the initial amount of outstanding data in bytes.
Taken from .
Minimum congestion window in bytes.
Reduction in congestion window when a new loss event is detected.Variables required to implement the congestion control mechanisms
are described in this section.
The highest value reported for the ECN-CE counter by the peer in an ACK_ECN
frame. This variable is used to detect increases in the reported ECN-CE
counter.
The sum of the size in bytes of all sent packets that contain at least
one retransmittable or PADDING frame, and have not been acked or declared
lost. The size does not include IP or UDP overhead, but does include the
QUIC header and AEAD overhead.
Packets only containing ACK frames do not count towards bytes_in_flight
to ensure congestion control does not impede congestion feedback.
Maximum number of bytes-in-flight that may be sent.
The largest packet number sent when QUIC detects a loss. When a larger
packet is acknowledged, QUIC exits recovery.
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.At the beginning of the connection, initialize the congestion control
variables as follows:Whenever a packet is sent, and it contains non-ACK frames,
the packet increases bytes_in_flight.Invoked from loss detection’s OnPacketAcked and is supplied with
acked_packet from sent_packets.Invoked from ProcessECN and OnPacketsLost when a new congestion event is
detected. Starts a new recovery period and reduces the congestion window.Invoked when an ACK_ECN frame is received from the peer.Invoked by loss detection from DetectLostPackets when new packets
are detected lost.QUIC decreases the congestion window to the minimum value once the
retransmission timeout has been verified and removes any packets
sent before the newly acknowledged RTO packet.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.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.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
, the exact response for each controller could be different. Failure
to correctly respond to information about ECN markings is therefore difficult to
detect.This document has no IANA actions. Yet.QUIC: A UDP-Based Multiplexed and Secure TransportFastlyMozillaKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.Relaxing Restrictions on Explicit Congestion Notification (ECN) ExperimentationThis memo updates RFC 3168, which specifies Explicit Congestion Notification (ECN) as an alternative to packet drops for indicating network congestion to endpoints. It relaxes restrictions in RFC 3168 that hinder experimentation towards benefits beyond just removal of loss. This memo summarizes the anticipated areas of experimentation and updates RFC 3168 to enable experimentation in these areas. An Experimental RFC in the IETF document stream is required to take advantage of any of these enabling updates. In addition, this memo makes related updates to the ECN specifications for RTP in RFC 6679 and for the Datagram Congestion Control Protocol (DCCP) in RFCs 4341, 4342, and 5622. This memo also records the conclusion of the ECN nonce experiment in RFC 3540 and provides the rationale for reclassification of RFC 3540 from Experimental to Historic; this reclassification enables new experimental use of the ECT(1) codepoint.Computing TCP's Retransmission TimerThis document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST. This document obsoletes RFC 2988. [STANDARDS-TRACK]TCP Congestion ControlThis document defines TCP's four intertwined congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. In addition, the document specifies how TCP should begin transmission after a relatively long idle period, as well as discussing various acknowledgment generation methods. This document obsoletes RFC 2581. [STANDARDS-TRACK]Early Retransmit for TCP and Stream Control Transmission Protocol (SCTP)This document proposes a new mechanism for TCP and Stream Control Transmission Protocol (SCTP) that can be used to recover lost segments when a connection's congestion window is small. The "Early Retransmit" mechanism allows the transport to reduce, in certain special circumstances, the number of duplicate acknowledgments required to trigger a fast retransmission. This allows the transport to use fast retransmit to recover segment losses that would otherwise require a lengthy retransmission timeout. [STANDARDS-TRACK]A Conservative Loss Recovery Algorithm Based on Selective Acknowledgment (SACK) for TCPThis document presents a conservative loss recovery algorithm for TCP that is based on the use of the selective acknowledgment (SACK) TCP option. The algorithm presented in this document conforms to the spirit of the current congestion control specification (RFC 5681), but allows TCP senders to recover more effectively when multiple segments are lost from a single flight of data. This document obsoletes RFC 3517 and describes changes from it. [STANDARDS-TRACK]Improving the Robustness of TCP to Non-Congestion EventsThis document specifies Non-Congestion Robustness (NCR) for TCP. In the absence of explicit congestion notification from the network, TCP uses loss as an indication of congestion. One of the ways TCP detects loss is using the arrival of three duplicate acknowledgments. However, this heuristic is not always correct, notably in the case when network paths reorder segments (for whatever reason), resulting in degraded performance. TCP-NCR is designed to mitigate this degraded performance by increasing the number of duplicate acknowledgments required to trigger loss recovery, based on the current state of the connection, in an effort to better disambiguate true segment loss from segment reordering. This document specifies the changes to TCP, as well as the costs and benefits of these modifications. This memo defines an Experimental Protocol for the Internet community.Tail Loss Probe (TLP): An Algorithm for Fast Recovery of Tail LossesRetransmission timeouts are detrimental to application latency, especially for short transfers such as Web transactions where timeouts can often take longer than all of the rest of a transaction. The primary cause of retransmission timeouts are lost segments at the tail of transactions. This document describes an experimental algorithm for TCP to quickly recover lost segments at the end of transactions or when an entire window of data or acknowledgments are lost. Tail Loss Probe (TLP) is a sender-only algorithm that allows the transport to recover tail losses through fast recovery as opposed to lengthy retransmission timeouts. If a connection is not receiving any acknowledgments for a certain period of time, TLP transmits the last unacknowledged segment (loss probe). In the event of a tail loss in the original transmissions, the acknowledgment from the loss probe triggers SACK/FACK based fast recovery. TLP effectively avoids long timeouts and thereby improves TCP performance.Forward RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious Retransmission Timeouts with TCPThe purpose of this document is to move the F-RTO (Forward RTO-Recovery) functionality for TCP in RFC 4138 from Experimental to Standards Track status. The F-RTO support for Stream Control Transmission Protocol (SCTP) in RFC 4138 remains with Experimental status. See Appendix B for the differences between this document and RFC 4138.Spurious retransmission timeouts cause suboptimal TCP performance because they often result in unnecessary retransmission of the last window of data. This document describes the F-RTO detection algorithm for detecting spurious TCP retransmission timeouts. F-RTO is a TCP sender-only algorithm that does not require any TCP options to operate. After retransmitting the first unacknowledged segment triggered by a timeout, the F-RTO algorithm of the TCP sender monitors the incoming acknowledgments to determine whether the timeout was spurious. It then decides whether to send new segments or retransmit unacknowledged segments. The algorithm effectively helps to avoid additional unnecessary retransmissions and thereby improves TCP performance in the case of a spurious timeout. [STANDARDS-TRACK]The NewReno Modification to TCP's Fast Recovery AlgorithmRFC 5681 documents the following four intertwined TCP congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. RFC 5681 explicitly allows certain modifications of these algorithms, including modifications that use the TCP Selective Acknowledgment (SACK) option (RFC 2883), and modifications that respond to "partial acknowledgments" (ACKs that cover new data, but not all the data outstanding when loss was detected) in the absence of SACK. This document describes a specific algorithm for responding to partial acknowledgments, referred to as "NewReno". This response to partial acknowledgments was first proposed by Janey Hoe. This document obsoletes RFC 3782. [STANDARDS-TRACK]TCP Congestion Control with Appropriate Byte Counting (ABC)This document proposes a small modification to the way TCP increases its congestion window. Rather than the traditional method of increasing the congestion window by a constant amount for each arriving acknowledgment, the document suggests basing the increase on the number of previously unacknowledged bytes each ACK covers. This change improves the performance of TCP, as well as closes a security hole TCP receivers can use to induce the sender into increasing the sending rate too rapidly. This memo defines an Experimental Protocol for the Internet community.Increasing TCP's Initial WindowThis document proposes an experiment to increase the permitted TCP initial window (IW) from between 2 and 4 segments, as specified in RFC 3390, to 10 segments with a fallback to the existing recommendation when performance issues are detected. It discusses the motivation behind the increase, the advantages and disadvantages of the higher initial window, and presents results from several large-scale experiments showing that the higher initial window improves the overall performance of many web services without resulting in a congestion collapse. The document closes with a discussion of usage and deployment for further experimental purposes recommended by the IETF TCP Maintenance and Minor Extensions (TCPM) working group.RFC Editor’s Note: Please remove this section prior to
publication of a final version of this document.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)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)No significant changes.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)No significant changes.Clarified pacing and RTO (#967, #977)Include Ack Delay in RTO(and TLP) computations (#981)Ack Delay in SRTT computation (#961)Default RTT and Slow Start (#590)Many editorial fixes.No significant changes.Add more congestion control text (#776)No significant changes.No significant changes.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)Overview added to loss detectionChanges initial default RTT to 100msAdded time-based loss detection and fixes early retransmitClarified loss recovery for handshake packetsFixed references and made TCP references informativeImproved description of constants and ACK behaviorAdopted as base for draft-ietf-quic-recoveryUpdated authors/editors listAdded table of contents