QUIC Loss Detection and Congestion ControlGooglejri@google.comGoogleianswett@google.com
Transport
QUICQUIC 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. QUIC implements the spirit of
known TCP loss detection mechanisms, described in RFCs, various Internet-drafts,
and also those prevalent in the Linux TCP implementation. This document
describes QUIC loss detection and congestion control, and attributes the TCP
equivalent in RFCs, Internet-drafts, academic papers, and TCP implementations.Discussion of this draft takes place on the QUIC working group mailing list
(quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic.Working Group information can be found at https://github.com/quicwg; source
code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/recovery.QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades
of transport and security experience, and implements mechanisms that make it
attractive as a modern general-purpose transport. The QUIC protocol is
described in .QUIC implements the spirit of known TCP loss recovery mechanisms, described in
RFCs, various Internet-drafts, and also those prevalent in the Linux TCP
implementation. This document describes QUIC congestion control and loss
recovery, and where applicable, attributes the TCP equivalent in RFCs,
Internet-drafts, academic papers, and/or TCP implementations.This document first describes pre-requisite parts of the QUIC transmission
machinery, then discusses QUIC’s default congestion control and loss detection
mechanisms, and finally lists the various TCP mechanisms that QUIC loss
detection implements (in spirit.)The words “MUST”, “MUST NOT”, “SHOULD”, and “MAY” are used in this document.
It’s not shouting; when they are capitalized, they have the special meaning
defined in .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.Retransmittable frames are frames requiring reliable delivery. The most
common are STREAM frames, which typically contain application data.Crypto handshake data is also sent as STREAM data, and uses the reliability
machinery of QUIC underneath.ACK frames contain acknowledgment information. QUIC uses a SACK-based
scheme, where acks express up to 256 ranges. The ACK frame also includes a
receive timestamp for each packet newly acked.There are some notable differences between QUIC and TCP which are important for
reasoning about the differences between the loss recovery mechanisms employed by
the two protocols. We briefly describe these 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.We now describe QUIC’s loss detection as functions that should be called on
packet transmission, when a packet is acked, and timer expiration events.Constants used in loss recovery and 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.
Maximum number of tail loss probes before an RTO fires.
Maximum reordering in packet number space before FACK style loss detection
considers a packet lost.
Maximum reordering in time sapce before time based loss detection considers
a packet lost. In fraction of an RTT.
Minimum time in the future a tail loss probe alarm may be set for.
Minimum time in the future an RTO alarm may be set for.
The length of the peer’s delayed ack timer.
The default RTT used before an RTT sample is taken.We first describe the variables required to implement the loss detection
mechanisms described in this section.
Multi-modal alarm used for loss detection.
The number of times the handshake packets have been
retransmitted without receiving an ack.
The number of times a tail loss probe has been sent without
receiving an ack.
The number of times an rto has been sent without receiving an ack.
The smoothed RTT of the connection, computed as described in
The RTT variance, computed as described in
The initial RTT used before any RTT measurements have been made.
The largest delta between the largest acked
retransmittable packet and a packet containing retransmittable frames before
it’s declared lost.
The reordering window as a fraction of max(smoothed_rtt, latest_rtt).
The time at which the next packet will be considered lost based on early
transmit or exceeding the reordering window in time.
An association of packet numbers to information about them, including a number
field indicating the packet number, a time field indicating the time a packet
was sent, and a bytes field indicating the packet’s size. sent_packets is
ordered by packet number, and packets remain in sent_packets until
acknowledged or lost.At the beginning of the connection, initialize the loss detection variables as
follows:After any packet is sent, be it a new transmission or a rebundled transmission,
the following OnPacketSent function is called. The parameters to OnPacketSent
are as follows:packet_number: The packet number of the sent packet.is_retransmittble: A boolean that indicates whether the packet contains at
least one frame requiring reliable deliver. The retransmittability of various
QUIC frames is described in . If false, it is still
acceptable for an ack to be received for this packet. However, a caller MUST
NOT set is_retransmittable to true if an ack is not expected.sent_bytes: The number of bytes sent in the packet.Pseudocode for OnPacketSent follows:When an ack is received, it may acknowledge 0 or more packets.Pseudocode for OnAckReceived and UpdateRtt follow:When a packet is acked for the first time, the following OnPacketAcked function
is called. Note that a single ACK frame may newly acknowledge several packets.
OnPacketAcked must be called once for each of these newly acked packets.OnPacketAcked takes one parameter, acked_packet, which is the packet number of
the newly acked packet, and returns a list of packet numbers that are detected
as lost.Pseudocode for OnPacketAcked follows: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. Once an RTT measurement is taken, it MUST replace initial_rtt.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 and retransmission
timeouts 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.Early retransmit is implemented with a 1/4 RTT timer. It is
part of QUIC’s time based loss detection, but is always enabled, even when
only packet reordering loss detection is enabled.Pseudocode for SetLossDetectionAlarm follows: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: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:(describe NewReno-style congestion control for QUIC.)
(describe appropriate byte counting.)
(define recovery based on packet numbers.)
(describe min_rtt based hystart.)
(describe how QUIC’s F-RTO delays reducing CWND.)
(describe PRR )This document has no IANA actions. Yet.QUIC: A UDP-Based Multiplexed and Secure TransportGoogleMozillaKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Computing TCP's Retransmission TimerThis document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST. This document obsoletes RFC 2988. [STANDARDS-TRACK]Tail Loss Probe (TLP): An Algorithm for Fast Recovery of Tail LossesRetransmission timeouts are detrimental to application latency, especially for short transfers such as Web transactions where timeouts can often take longer than all of the rest of a transaction. The primary cause of retransmission timeouts are lost segments at the tail of transactions. This document describes an experimental algorithm for TCP to quickly recover lost segments at the end of transactions or when an entire window of data or acknowledgments are lost. Tail Loss Probe (TLP) is a sender-only algorithm that allows the transport to recover tail losses through fast recovery as opposed to lengthy retransmission timeouts. If a connection is not receiving any acknowledgments for a certain period of time, TLP transmits the last unacknowledged segment (loss probe). In the event of a tail loss in the original transmissions, the acknowledgment from the loss probe triggers SACK/FACK based fast recovery. TLP effectively avoids long timeouts and thereby improves TCP performance.Early Retransmit for TCP and Stream Control Transmission Protocol (SCTP)This document proposes a new mechanism for TCP and Stream Control Transmission Protocol (SCTP) that can be used to recover lost segments when a connection's congestion window is small. The "Early Retransmit" mechanism allows the transport to reduce, in certain special circumstances, the number of duplicate acknowledgments required to trigger a fast retransmission. This allows the transport to use fast retransmit to recover segment losses that would otherwise require a lengthy retransmission timeout. [STANDARDS-TRACK]The NewReno Modification to TCP's Fast Recovery AlgorithmRFC 5681 documents the following four intertwined TCP congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. RFC 5681 explicitly allows certain modifications of these algorithms, including modifications that use the TCP Selective Acknowledgment (SACK) option (RFC 2883), and modifications that respond to "partial acknowledgments" (ACKs that cover new data, but not all the data outstanding when loss was detected) in the absence of SACK. This document describes a specific algorithm for responding to partial acknowledgments, referred to as "NewReno". This response to partial acknowledgments was first proposed by Janey Hoe. This document obsoletes RFC 3782. [STANDARDS-TRACK]Forward RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious Retransmission Timeouts with TCPThe purpose of this document is to move the F-RTO (Forward RTO-Recovery) functionality for TCP in RFC 4138 from Experimental to Standards Track status. The F-RTO support for Stream Control Transmission Protocol (SCTP) in RFC 4138 remains with Experimental status. See Appendix B for the differences between this document and RFC 4138.Spurious retransmission timeouts cause suboptimal TCP performance because they often result in unnecessary retransmission of the last window of data. This document describes the F-RTO detection algorithm for detecting spurious TCP retransmission timeouts. F-RTO is a TCP sender-only algorithm that does not require any TCP options to operate. After retransmitting the first unacknowledged segment triggered by a timeout, the F-RTO algorithm of the TCP sender monitors the incoming acknowledgments to determine whether the timeout was spurious. It then decides whether to send new segments or retransmit unacknowledged segments. The algorithm effectively helps to avoid additional unnecessary retransmissions and thereby improves TCP performance in the case of a spurious timeout. [STANDARDS-TRACK]Proportional Rate Reduction for TCPThis document describes an experimental Proportional Rate Reduction (PRR) algorithm as an alternative to the widely deployed Fast Recovery and Rate-Halving algorithms. These algorithms determine the amount of data sent by TCP during loss recovery. PRR minimizes excess window adjustments, and the actual window size at the end of recovery will be as close as possible to the ssthresh, as determined by the congestion control algorithm.RFC Editor’s Note: Please remove this section prior to publication of a
final version of this document.Changes initial default RTT to 100msAdded time-based loss detection and fixes early retransmitClarified loss recovery for handshake packetsFixed references and made TCP references informativeImproved description of constants and ACK behaviorAdopted as base for draft-ietf-quic-recovery.Updated authors/editors list.Added table of contents.