QUIC Loss Detection and Congestion Control

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.¶

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.¶

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.¶

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.¶

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.¶

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].¶

QUIC uses a probe timeout (PTO; see Section 6.2), 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 Section 7.6. 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.¶

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 (Section 6.2), 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 (Section 6.2).¶

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.¶

• the largest acknowledged packet number is newly acknowledged, and
• at least one of the newly acknowledged packets was ack-eliciting.

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 (Section 5.3).¶

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.¶

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 (Section 5.1) 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 Section 5.3) 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 Section 5.3.¶

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.¶

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.¶

• 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 Section 6.2.2.¶

smoothed_rtt = kInitialRtt
rttvar = kInitialRtt / 2

RTT samples for the network path are recorded in latest_rtt; see Section 5.1. 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].¶

smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2

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

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.¶

• The packet is unacknowledged, in flight, and was sent prior to an acknowledged packet.
• The packet was sent kPacketThreshold packets before an acknowledged packet (Section 6.1.1), or it was sent long enough in the past (Section 6.1.2).

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.¶

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 Section 6.1.¶

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].¶

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.¶

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].¶

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].¶

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.¶

The NewReno congestion controller described in this document has three distinct states, as shown in Figure 1.¶

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

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 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.¶

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

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 Section 7.2. 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.¶

rate = N * congestion_window / smoothed_rtt

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.¶

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 Section 7.7) 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].¶

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.¶

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.¶

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].¶

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 (Appendix A.1.1) 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.¶

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 Section 6.1.1 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 Section 6.1.2 is 9/8.

kGranularity:

Timer granularity. This is a system-dependent value, and Section 6.1.2 recommends a value of 1 ms.

kInitialRtt:

The RTT used before an RTT sample is taken. The value recommended in Section 6.2.2 is 333 ms.

kPacketNumberSpace:

An enum to enumerate the three packet number spaces:

enum kPacketNumberSpace {
  Initial,
  Handshake,
  ApplicationData,
}

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 Section 5.3.

rttvar:

The RTT variation, computed as described in Section 5.3.

min_rtt:

The minimum RTT seen over a period of time, ignoring acknowledgment delay, as described in Section 5.2.

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 Appendix A.1.

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

After a packet is sent, information about the packet is stored. The parameters to OnPacketSent are described in detail above in Appendix A.1.1.¶

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()

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.¶

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()

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

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

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.¶

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)

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

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()

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.¶

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

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

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()

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 Section 7.2.

kMinimumWindow:

Minimum congestion window in bytes as described in Section 7.2.

kLossReductionFactor:

Scaling factor applied to reduce the congestion window when a new loss event is detected. Section 7 recommends a value of 0.5.

kPersistentCongestionThreshold:

Period of time for persistent congestion to be established, specified as a PTO multiplier. Section 7.6 recommends a value of 3.

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.¶

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

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

OnPacketSentCC(sent_bytes):
  bytes_in_flight += sent_bytes

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].¶

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

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.¶

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()

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

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)

This is invoked when DetectAndRemoveLostPackets deems packets lost.¶

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

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

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