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 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 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 non-trivial 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 ACKs contain information that is similar to TCP SACK, but QUIC does not allow any acknowledged 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 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 round-trip time 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 RTO computation. QUIC’s PTO includes the peer’s maximum expected acknowledgement delay instead of using a fixed minimum timeout. QUIC does not collapse the congestion window until persistent congestion (Section 7.6) is declared, unlike TCP, which collapses the congestion window upon expiry of an RTO. Instead of collapsing the congestion window and declaring everything in-flight lost, QUIC allows probe packets to temporarily exceed the congestion window whenever the timer expires.¶

In doing this, QUIC avoids unnecessary congestion window reductions, obviating the need for correcting mechanisms such as 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.¶

A single packet loss at the tail does not indicate persistent congestion, so QUIC specifies a time-based definition to ensure one or more packets are sent prior to a dramatic decrease in congestion window; see Section 7.6.¶

TCP uses a minimum congestion window of one packet. However, loss of that single packet means that the sender needs to waiting for a PTO (Section 6.2) to recover, which can be much longer than a round-trip time. Sending a single ack-eliciting packet also increases the chances of incurring additional latency when a receiver delays its acknowledgement.¶

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).¶

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

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 acknowledgements 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. 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 is to allow a connection to reset its estimate of min_rtt and smoothed_rtt (Section 5.3) after a disruptive network event, and because it is possible that an increase in path delay resulted in persistent congestion being incorrectly declared.¶

Endpoints MAY re-establish the min_rtt at other times in the connection, such as when traffic volume is low and an acknowledgement is received with a low acknowledgement 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 is the variation in the RTT samples, estimated using a mean variation.¶

The calculation of smoothed_rtt uses RTT samples after adjusting them for acknowledgement 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 acknowledgement 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 (Section 4.1.2 of [QUIC-TLS]). When they occur, these large acknowledgement 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 however that a large acknowledgement delay can result in a substantially inflated smoothed_rtt, if there is either an error in the peer’s reporting of the acknowledgement 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 acknowledgement delay causes the sample to be less than the min_rtt.¶

After the handshake is confirmed, any acknowledgement 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 acknowledgements. Excess delays could also be due to a non-compliant receiver. Therefore, these extra delays are considered effectively part of path delay and incorporated into the RTT estimate.¶

• MAY ignore the acknowledgement delay for Initial packets, since these acknowledgements 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 acknowledgement delay and the peer’s max_ack_delay after the handshake is confirmed; and
• MUST NOT subtract the acknowledgement 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 acknowledgements when the corresponding decryption keys are not immediately available. For example, a client might receive an acknowledgement 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.¶

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.¶

Acknowledgement-based loss detection implements the spirit of TCP’s Fast Retransmit ([RFC5681]), Early Retransmit ([RFC5827]), FACK ([FACK]), SACK loss recovery ([RFC6675]), and RACK ([RACK]). 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 acknowledgement 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 reordering threshold in packets or time 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 sending 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 acknowledgements.¶

As with loss detection, the probe timeout 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 acknowledgement is received that newly acknowledges packets, loss detection proceeds as dictated by packet and time threshold mechanisms; see Section 6.1.¶

The PTO algorithm used in QUIC implements the reliability functions of Tail Loss Probe [RACK], RTO [RFC5681], and F-RTO algorithms for TCP [RFC5682]. The timeout computation is based on TCP’s retransmission timeout 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 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 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 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 acknowledgements cannot be processed anymore. 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 after packets encrypted with them would be acknowledged or declared lost. However, Initial secrets are discarded as soon as handshake 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 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 10 times the maximum datagram size (max_datagram_size), limited to the larger of 14720 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 decrease to as a response to loss, 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 acknowledgement 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. Packets containing only ACK frames SHOULD therefore not be paced, to avoid delaying their delivery to the peer.¶

Using a value for N that is small, but at least 1 (for example, 1.25) ensures that variations in round-trip time do not result in under-utilization 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 a round-trip time.¶

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 under-utilized. When this occurs, the congestion window SHOULD NOT be increased in either slow start or congestion avoidance. This can happen due to insufficient application data or flow control limits.¶

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 under-utilization, such as those proposed for TCP in [RFC7661].¶

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. 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 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 1ms.

kInitialRtt:

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

kPacketNumberSpace:

An enum to enumerate the three packet number spaces.

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

latest_rtt:

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

smoothed_rtt:

The smoothed RTT of the connection, computed as described in Section 5.3.

rttvar:

The RTT variation, computed as described in Section 5.3.

min_rtt:

The minimum RTT seen in the connection, 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 ack.

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 will 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.

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.¶

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 re-armed.¶

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

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.¶

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

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.¶

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

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:

Reduction in congestion window when a new loss event is detected. Section 7 recommends a value is 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. 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 AEAD overhead. Packets only containing ACK frames do not count towards bytes_in_flight to ensure congestion control does not impede congestion feedback.

congestion_window:

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

congestion_recovery_start_time:

The time when QUIC first detects congestion due to loss or ECN, causing it to enter congestion recovery. 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.¶

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

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].¶

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.¶

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

Invoked when DetectAndRemoveLostPackets deems packets lost.¶

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

• Clarifications to definition of persistent congestion (#4413, #4414, #4421, #4429, #4437)

• Limit the number of Initial packets sent in response to unauthenticated packets (#4183, #4188)

Editorial changes only.¶

• Allow caching of packets that can’t be decrypted, by allowing the reported acknowledgment delay to exceed max_ack_delay prior to confirming the handshake (#3821, #3980, #4035, #3874)
• Persistent congestion cannot include packets sent before the first RTT sample for the path (#3875, #3889)
• Recommend reset of min_rtt in persistent congestion (#3927, #3975)
• Persistent congestion is independent of packet number space (#3939, #3961)
• Only limit bursts to the initial window without information about the path (#3892, #3936)
• Add normative requirements for increasing and reducing the congestion window (#3944, #3978, #3997, #3998)

• Refactored pseudocode to correct PTO calculation (#3564, #3674, #3681)

• Added recommendations for speeding up handshake under some loss conditions (#3078, #3080)
• PTO count is reset when handshake progress is made (#3272, #3415)
• PTO count is not reset by a client when the server might be awaiting address validation (#3546, #3551)
• Recommend repairing losses immediately after entering the recovery period (#3335, #3443)
• Clarified what loss conditions can be ignored during the handshake (#3456, #3450)
• Allow, but don’t recommend, using RTT from previous connection to seed RTT (#3464, #3496)
• Recommend use of adaptive loss detection thresholds (#3571, #3572)

No changes.¶

No significant changes.¶

• Require congestion control of some sort (#3247, #3244, #3248)
• Set a minimum reordering threshold (#3256, #3240)
• PTO is specific to a packet number space (#3067, #3074, #3066)

• Define under-utilizing the congestion window (#2630, #2686, #2675)
• PTO MUST send data if possible (#3056, #3057)
• Connection Close is not ack-eliciting (#3097, #3098)
• MUST limit bursts to the initial congestion window (#3160)
• Define the current max_datagram_size for congestion control (#3041, #3167)

• PTO should always send an ack-eliciting packet (#2895)
• Unify the Handshake Timer with the PTO timer (#2648, #2658, #2886)
• Move ACK generation text to transport draft (#1860, #2916)

• No changes

• Path validation can be used as initial RTT value (#2644, #2687)
• max_ack_delay transport parameter defaults to 0 (#2638, #2646)
• ACK delay only measures intentional delays induced by the implementation (#2596, #2786)

• Change kPersistentThreshold from an exponent to a multiplier (#2557)
• Send a PING if the PTO timer fires and there’s nothing to send (#2624)
• Set loss delay to at least kGranularity (#2617)
• Merge application limited and sending after idle sections. Always limit burst size instead of requiring resetting CWND to initial CWND after idle (#2605)
• Rewrite RTT estimation, allow RTT samples where a newly acked packet is ack-eliciting but the largest_acked is not (#2592)
• Don’t arm the handshake timer if there is no handshake data (#2590)
• Clarify that the time threshold loss alarm takes precedence over the crypto handshake timer (#2590, #2620)
• Change initial RTT to 500ms to align with RFC6298 (#2184)

• Change IW byte limit to 14720 from 14600 (#2494)
• Update PTO calculation to match RFC6298 (#2480, #2489, #2490)
• Improve loss detection’s description of multiple packet number spaces and pseudocode (#2485, #2451, #2417)
• Declare persistent congestion even if non-probe packets are sent and don’t make persistent congestion more aggressive than RTO verified was (#2365, #2244)
• Move pseudocode to the appendices (#2408)
• What to send on multiple PTOs (#2380)

• After Probe Timeout discard in-flight packets or send another (#2212, #1965)
• Endpoints discard initial keys as soon as handshake keys are available (#1951, #2045)
• 0-RTT state is discarded when 0-RTT is rejected (#2300)
• Loss detection timer is cancelled when ack-eliciting frames are in flight (#2117, #2093)
• Packets are declared lost if they are in flight (#2104)
• After becoming idle, either pace packets or reset the congestion controller (#2138, 2187)
• Process ECN counts before marking packets lost (#2142)
• Mark packets lost before resetting crypto_count and pto_count (#2208, #2209)
• Congestion and loss recovery state are discarded when keys are discarded (#2327)

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

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

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

• Improved description of constants and ACK behavior

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