QUIC Working Group | J. Iyengar, Editor |
Internet-Draft | I. Swett, Editor |
Intended status: Standards Track | |
Expires: December 15, 2017 | June 13, 2017 |
This 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.¶
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.¶
Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at http://datatracker.ietf.org/drafts/current/.¶
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This Internet-Draft will expire on December 15, 2017.¶
Copyright (c) 2017 IETF Trust and the persons identified as the document authors. All rights reserved.¶
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QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades of transport and security experience, and implements mechanisms that make it attractive as a modern general-purpose transport. The QUIC protocol is described in [QUIC-TRANSPORT].¶
QUIC implements the spirit of known TCP loss recovery mechanisms, described in RFCs, various Internet-drafts, and also those prevalent in the Linux TCP implementation. This document describes QUIC congestion control and loss recovery, and where applicable, attributes the TCP equivalent in RFCs, Internet-drafts, academic papers, and/or TCP implementations.¶
All transmissions in QUIC are sent with a packet-level header, which includes a packet sequence number (referred to below as a packet number). These packet numbers never repeat in the lifetime of a connection, and are monotonically increasing, which makes duplicate detection trivial. This fundamental design decision obviates the need for disambiguating between transmissions and retransmissions and eliminates significant complexity from QUIC’s interpretation of TCP loss detection mechanisms.¶
Every packet may contain several frames. We outline the frames that are important to the loss detection and congestion control machinery below.¶
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.¶
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 sequence number (referred to as the “packet number”) for transmissions, and any data that is to be delivered to the receiving application(s) is sent in one or more streams, with stream offsets encoded within STREAM frames inside of packets that determine delivery order.¶
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 equivalent 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 up to 256 ACK ranges, opposed to TCP’s 3 SACK ranges. In high loss environments, this speeds recovery.¶
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 uses a combination of ack information and alarms to detect lost packets. An unacknowledged QUIC packet is marked as lost in one of the following ways:¶
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.¶
Variables required to implement the congestion control mechanisms are described in this section.¶
At the beginning of the connection, initialize the loss detection variables as follows:¶
loss_detection_alarm.reset() handshake_count = 0 tlp_count = 0 rto_count = 0 if (UsingTimeLossDetection()) reordering_threshold = infinite time_reordering_fraction = kTimeReorderingFraction else: reordering_threshold = kReorderingThreshold time_reordering_fraction = infinite loss_time = 0 smoothed_rtt = 0 rttvar = 0 largest_sent_before_rto = 0 time_of_last_sent_packet = 0
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:¶
Pseudocode for OnPacketSent follows:¶
OnPacketSent(packet_number, is_retransmittable, sent_bytes): time_of_last_sent_packet = now; sent_packets[packet_number].packet_number = packet_number sent_packets[packet_number].time = now if is_retransmittable: sent_packets[packet_number].bytes = sent_bytes SetLossDetectionAlarm()
When an ack is received, it may acknowledge 0 or more packets.¶
Pseudocode for OnAckReceived and UpdateRtt follow:¶
OnAckReceived(ack): // If the largest acked is newly acked, update the RTT. if (sent_packets[ack.largest_acked]): latest_rtt = now - sent_packets[ack.largest_acked].time if (latest_rtt > ack.ack_delay): latest_rtt -= ack.delay UpdateRtt(latest_rtt) // Find all newly acked packets. for acked_packet in DetermineNewlyAckedPackets(): OnPacketAcked(acked_packet.packet_number) DetectLostPackets(ack.largest_acked_packet) SetLossDetectionAlarm() UpdateRtt(latest_rtt): // Based on {{RFC6298}}. if (smoothed_rtt == 0): smoothed_rtt = latest_rtt rttvar = latest_rtt / 2 else: rttvar = 3/4 * rttvar + 1/4 * (smoothed_rtt - latest_rtt) smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt
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 packet number of the newly acked packet, and returns a list of packet numbers that are detected as lost.¶
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 [RFC5682]¶
Pseudocode for OnPacketAcked follows:¶
OnPacketAcked(acked_packet_number): // If a packet sent prior to RTO was acked, then the RTO // was spurious. Otherwise, inform congestion control. if (rto_count > 0 && acked_packet_number > largest_sent_before_rto) OnRetransmissionTimeoutVerified() handshake_count = 0 tlp_count = 0 rto_count = 0 sent_packets.remove(acked_packet_number)
QUIC loss detection uses a single alarm for all timer-based loss detection. The duration of the alarm is based on the alarm’s mode, which is set in the packet and timer events further below. The function SetLossDetectionAlarm defined below shows how the single timer is set based on the alarm mode.¶
The initial flight has no prior RTT sample. A client SHOULD remember the previous RTT it observed when resumption is attempted and use that for an initial RTT value. If no previous RTT is available, the initial RTT defaults to 200ms.¶
Endpoints MUST retransmit handshake frames if not acknowledged within a time limit. This time limit will start as the largest of twice the rtt value and MinTLPTimeout. Each consecutive handshake retransmission doubles the time limit, until an acknowledgement is received.¶
Handshake frames may be cancelled by handshake state transitions. In particular, all non-protected frames SHOULD be no longer be transmitted once packet protection is available.¶
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 per handshake packet that uses an unsupported QUIC version, and MAY be sent in response to 0RTT packets.¶
Tail loss probes [LOSS-PROBE] and retransmission timeouts [RFC6298] are an alarm based mechanism to recover from cases when there are outstanding retransmittable packets, but an acknowledgement has not been received in a timely manner.¶
Pseudocode for SetLossDetectionAlarm follows:¶
SetLossDetectionAlarm(): if (retransmittable packets are not outstanding): loss_detection_alarm.cancel() return if (handshake packets are outstanding): // Handshake retransmission alarm. if (smoothed_rtt == 0): alarm_duration = 2 * kDefaultInitialRtt else: alarm_duration = 2 * smoothed_rtt alarm_duration = max(alarm_duration, kMinTLPTimeout) alarm_duration = alarm_duration * (2 ^ handshake_count) else if (loss_time != 0): // Early retransmit timer or time loss detection. alarm_duration = loss_time - now else if (tlp_count < kMaxTLPs): // Tail Loss Probe if (retransmittable_packets_outstanding = 1): alarm_duration = 1.5 * smoothed_rtt + kDelayedAckTimeout else: alarm_duration = kMinTLPTimeout alarm_duration = max(alarm_duration, 2 * smoothed_rtt) else: // RTO alarm alarm_duration = smoothed_rtt + 4 * rttvar alarm_duration = max(alarm_duration, kMinRTOTimeout) alarm_duration = alarm_duration * (2 ^ rto_count) loss_detection_alarm.set(now + alarm_duration)
QUIC uses one loss recovery alarm, which when set, can be in one of several modes. When the alarm fires, the mode determines the action to be performed.¶
Pseudocode for OnLossDetectionAlarm follows:¶
OnLossDetectionAlarm(): if (handshake packets are outstanding): // Handshake retransmission alarm. RetransmitAllHandshakePackets() handshake_count++ else if (loss_time != 0): // Early retransmit or Time Loss Detection DetectLostPackets(largest_acked_packet) else if (tlp_count < kMaxTLPs): // Tail Loss Probe. SendOnePacket() tlp_count++ else: // RTO. if (rto_count == 0) largest_sent_before_rto = largest_sent_packet SendTwoPackets() rto_count++ SetLossDetectionAlarm()
Packets in QUIC are only considered lost once a larger packet number is acknowledged. DetectLostPackets is called every time an ack is received. If the loss detection alarm fires and the loss_time is set, the previous largest acked packet is supplied.¶
The receiver MUST ignore unprotected packets that ack protected packets. The receiver MUST trust protected acks for unprotected packets, however. Aside from this, loss detection for handshake packets when an ack is processed is identical to other packets.¶
DetectLostPackets takes one parameter, acked, which is the largest acked packet.¶
Pseudocode for DetectLostPackets follows:¶
DetectLostPackets(largest_acked): loss_time = 0 lost_packets = {} delay_until_lost = infinite if (time_reordering_fraction != infinite): delay_until_lost = (1 + time_reordering_fraction) * max(latest_rtt, smoothed_rtt) else if (largest_acked.packet_number == largest_sent_packet): // Early retransmit alarm. delay_until_lost = 9/8 * max(latest_rtt, smoothed_rtt) foreach (unacked < largest_acked.packet_number): time_since_sent = now() - unacked.time_sent packet_delta = largest_acked.packet_number - unacked.packet_number if (time_since_sent > delay_until_lost): lost_packets.insert(unacked) else if (packet_delta > reordering_threshold) lost_packets.insert(unacked) else if (loss_time == 0 && delay_until_lost != infinite): loss_time = now() + delay_until_lost - time_since_sent // Inform the congestion controller of lost packets and // lets it decide whether to retransmit immediately. if (!lost_packets.empty()) OnPacketsLost(lost_packets) foreach (packet in lost_packets) sent_packets.remove(packet.packet_number)
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[RFC6582] congestion control to determine the congestion window and pacing rate.¶
QUIC begins every connection in slow start and exits slow start upon loss. While in slow start, QUIC increases the congestion window by the number of acknowledged bytes when each ack is processed.¶
Recovery is a period of time beginning with detection of a lost packet. It ends when all packets outstanding at the time recovery began have been acknowledged or lost. During recovery, the congestion window is not increased or decreased.¶
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.¶
Variables required to implement the congestion control mechanisms are described in this section.¶
At the beginning of the connection, initialize the loss detection variables as follows:¶
congestion_window = kInitialWindow bytes_in_flight = 0 end_of_recovery = 0 ssthresh = infinite
Invoked at the same time loss detection’s OnPacketAcked is called and supplied with the acked_packet from sent_packets.¶
Pseudocode for OnPacketAcked follows:¶
OnPacketAcked(acked_packet): if (acked_packet.packet_number < end_of_recovery): return if (congestion_window < ssthresh): congestion_window += acket_packets.bytes else: congestion_window += acked_packets.bytes / congestion_window
Invoked by loss detection from DetectLostPackets when new packets are detected lost.¶
OnPacketsLost(lost_packets): largest_lost_packet = lost_packets.last() // Start a new recovery epoch if the lost packet is larger // than the end of the previous recovery epoch. if (end_of_recovery < largest_lost_packet.packet_number): end_of_recovery = largest_sent_packet congestion_window *= kLossReductionFactor congestion_window = max(congestion_window, kMinimumWindow) ssthresh = congestion_window
QUIC decreases the congestion window to the minimum value once the retransmission timeout has been confirmed to not be spurious when the first post-RTO acknowledgement is processed.¶
OnRetransmissionTimeoutVerified() congestion_window = kMinimumWindow
QUIC sends a packet if there is available congestion window and sending the packet does not exceed the pacing rate.¶
TimeToSend returns infinite if the congestion controller is congestion window limited, a time in the past if the packet can be sent immediately, and a time in the future if sending is pacing limited.¶
TimeToSend(packet_size): if (bytes_in_flight + packet_size > congestion_window) return infinite return time_of_last_sent_packet + (packet_size * smoothed_rtt) / congestion_window
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