Internet Engineering Task Force (IETF)                   J. Iyengar, Ed.
Request for Comments: 9002                                        Fastly
Category: Standards Track                                  I. Swett, Ed.
ISSN: 2070-1721                                                   Google
                                                                May 2021

QUIC Loss Detection and Congestion Control




This document describes loss detection and congestion control mechanisms for QUIC.


Status of This Memo


This is an Internet Standards Track document.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841.

この文書は、インターネットエンジニアリングタスクフォース(IETF)の製品です。IETFコミュニティのコンセンサスを表します。それは公開レビューを受け、インターネットエンジニアリングステアリンググループ(IESG)による出版の承認を受けました。インターネット規格に関する詳細情報は、RFC 7841のセクション2で利用できます。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved.

著作権(C)2021 IETF信頼と文書著者として識別された人。全著作権所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

このドキュメントは、このドキュメントの発行日に有効なBCP 78およびIETFドキュメントに関連するIETFトラストの法的規定(の対象となります。 これらのドキュメントは、このドキュメントに関するお客様の権利と制限について説明しているため、注意深く確認してください。 このドキュメントから抽出されたコードコンポーネントには、Trust LegalProvisionsのセクション4.eで説明されているSimplifiedBSD Licenseテキストが含まれている必要があり、Simplified BSDLicenseで説明されているように保証なしで提供されます。

Table of Contents


   1.  Introduction
   2.  Conventions and Definitions
   3.  Design of the QUIC Transmission Machinery
   4.  Relevant Differences between QUIC and TCP
     4.1.  Separate Packet Number Spaces
     4.2.  Monotonically Increasing Packet Numbers
     4.3.  Clearer Loss Epoch
     4.4.  No Reneging
     4.5.  More ACK Ranges
     4.6.  Explicit Correction for Delayed Acknowledgments
     4.7.  Probe Timeout Replaces RTO and TLP
     4.8.  The Minimum Congestion Window Is Two Packets
     4.9.  Handshake Packets Are Not Special
   5.  Estimating the Round-Trip Time
     5.1.  Generating RTT Samples
     5.2.  Estimating min_rtt
     5.3.  Estimating smoothed_rtt and rttvar
   6.  Loss Detection
     6.1.  Acknowledgment-Based Detection
       6.1.1.  Packet Threshold
       6.1.2.  Time Threshold
     6.2.  Probe Timeout
       6.2.1.  Computing PTO
       6.2.2.  Handshakes and New Paths
       6.2.3.  Speeding up Handshake Completion
       6.2.4.  Sending Probe Packets
     6.3.  Handling Retry Packets
     6.4.  Discarding Keys and Packet State
   7.  Congestion Control
     7.1.  Explicit Congestion Notification
     7.2.  Initial and Minimum Congestion Window
     7.3.  Congestion Control States
       7.3.1.  Slow Start
       7.3.2.  Recovery
       7.3.3.  Congestion Avoidance
     7.4.  Ignoring Loss of Undecryptable Packets
     7.5.  Probe Timeout
     7.6.  Persistent Congestion
       7.6.1.  Duration
       7.6.2.  Establishing Persistent Congestion
       7.6.3.  Example
     7.7.  Pacing
     7.8.  Underutilizing the Congestion Window
   8.  Security Considerations
     8.1.  Loss and Congestion Signals
     8.2.  Traffic Analysis
     8.3.  Misreporting ECN Markings
   9.  References
     9.1.  Normative References
     9.2.  Informative References
   Appendix A.  Loss Recovery Pseudocode
     A.1.  Tracking Sent Packets
       A.1.1.  Sent Packet Fields
     A.2.  Constants of Interest
     A.3.  Variables of Interest
     A.4.  Initialization
     A.5.  On Sending a Packet
     A.6.  On Receiving a Datagram
     A.7.  On Receiving an Acknowledgment
     A.8.  Setting the Loss Detection Timer
     A.9.  On Timeout
     A.10. Detecting Lost Packets
     A.11. Upon Dropping Initial or Handshake Keys
   Appendix B.  Congestion Control Pseudocode
     B.1.  Constants of Interest
     B.2.  Variables of Interest
     B.3.  Initialization
     B.4.  On Packet Sent
     B.5.  On Packet Acknowledgment
     B.6.  On New Congestion Event
     B.7.  Process ECN Information
     B.8.  On Packets Lost
     B.9.  Removing Discarded Packets from Bytes in Flight
   Authors' Addresses
1. Introduction
1. はじめに

QUIC is a secure, general-purpose transport protocol, described in [QUIC-TRANSPORT]. This document describes loss detection and congestion control mechanisms for QUIC.


2. Conventions and Definitions
2. 表記法と定義

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.

この文書のキーワード "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", および "OPTIONAL" はBCP 14 [RFC2119] [RFC8174]で説明されているように、すべて大文字の場合にのみ解釈されます。

Definitions of terms that are used in this document:


Ack-eliciting frames: All frames other than ACK, PADDING, and CONNECTION_CLOSE are considered ack-eliciting.

ACK誘発フレーム: ACK、PADDING、およびCONNECTION_CLOSE以外のすべてのフレームはACK誘発と見なされます。

Ack-eliciting packets: Packets that contain ack-eliciting frames elicit an ACK from the receiver within the maximum acknowledgment delay and are called ack-eliciting packets.


In-flight packets: Packets are considered in flight when they are ack-eliciting or contain a PADDING frame, and they have been sent but are not acknowledged, declared lost, or discarded along with old keys.


3. Design of the QUIC Transmission Machinery
3. QUICトランスミッション機械の設計

All transmissions in QUIC are sent with a packet-level header, which indicates the encryption level and includes a packet sequence number (referred to below as a packet number). The encryption level indicates the packet number space, as described in Section 12.3 of [QUIC-TRANSPORT]. Packet numbers never repeat within a packet number space for the lifetime of a connection. Packet numbers are sent in monotonically increasing order within a space, preventing ambiguity. It is permitted for some packet numbers to never be used, leaving intentional gaps.


This design obviates the need for disambiguating between transmissions and retransmissions; this eliminates significant complexity from QUIC's interpretation of TCP loss detection mechanisms.


QUIC packets can contain multiple frames of different types. The recovery mechanisms ensure that data and frames that need reliable delivery are acknowledged or declared lost and sent in new packets as necessary. The types of frames contained in a packet affect recovery and congestion control logic:


* All packets are acknowledged, though packets that contain no ack-eliciting frames are only acknowledged along with ack-eliciting packets.

* ACK誘発フレームを含まないパケットは、ACK誘発パケットと一緒に承認されているだけですが、すべてのパケットが確認されます。

* Long header packets that contain CRYPTO frames are critical to the performance of the QUIC handshake and use shorter timers for acknowledgment.

* CRYPTOフレームを含む長いヘッダパケットは、QUICハンドシェイクの性能にとって重要であり、承認のために短いタイマーを使用します。

* Packets containing frames besides ACK or CONNECTION_CLOSE frames count toward congestion control limits and are considered to be in flight.

* ACKまたはCONNECTION_CLOSEフレーム以外のフレームを含むパケットは、輻輳制御限界に向かって数え、飛行中のと考えられています。

* PADDING frames cause packets to contribute toward bytes in flight without directly causing an acknowledgment to be sent.

* パディングフレームは、承認を送信することなく、直接アクセスされずにフライト内のバイトに貢献するようにパケットを貢献します。

4. Relevant Differences between QUIC and TCP
4. QUICとTCPの違い

Readers familiar with TCP's loss detection and congestion control will find algorithms here that parallel well-known TCP ones. However, protocol differences between QUIC and TCP contribute to algorithmic differences. These protocol differences are briefly described below.


4.1. Separate Packet Number Spaces
4.1. 別々のパケット番号スペース

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.


4.2. Monotonically Increasing Packet Numbers
4.2. 単調増加するパケット番号

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.


4.3. Clearer Loss Epoch
4.3. 明確な損失のエポック

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.


4.4. No Reneging
4.4. ノーレンギング

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 ACKフレームには、TCP選択承認(SACKS)[RFC2018]の情報と同様の情報が含まれています。ただし、QUICでは、パケット確認応答を再調整し、両側の実装を大幅に簡素化し、送信者のメモリ圧力を低減できます。

4.5. More ACK Ranges
4.5. より広いACK範囲

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.


4.6. Explicit Correction for Delayed Acknowledgments
4.6. 遅延確認応答の明示的な補正

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].


4.7. Probe Timeout Replaces RTO and TLP
4.7. プローブタイムアウトによるRTOとTLPの置換

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.

QUICはプローブタイムアウト(PTO; 6.2を参照)を使用し、TCPの再送信タイムアウト(RTO)計算に基づくタイマーを使用します。[RFC6298]を参照してください。QUICのPTOは、最小タイムアウトを使用する代わりに、ピアの最大予想確認応答遅延を含みます。

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.

TCP [RFC8985]のRACK-TLP損失検出アルゴリズムと同様に、PTOが期限切れになったときにQUICは輻輳ウィンドウを折りたたんでいません。代わりに、永続的な輻輳が宣言されている場合、QUICが輻輳ウィンドウを折りたたみます。7.6項を参照してください。これにより、不要な輻輳ウィンドウの削減は不要な輻輳ウィンドウの削減を避け、前方RTO回復(F-RTO)[RFC5682]などのメカニズムを修正する必要性を避けます。QUICはPTOの有効期限で輻輳ウィンドウを折りたたんではないので、QUICの送信者は、それでも利用可能な輻輳ウィンドウがある場合、PTOの有効期限後により多くの飛行パケットの送信を制限されません。これは、送信者がアプリケーション制限されており、PTOタイマーが期限切れになると発生します。これは、アプリケーション制限時のTCPのRTOメカニズムよりも積極的ですが、アプリケーション制限が制限されていない場合は同一です。

QUIC allows probe packets to temporarily exceed the congestion window whenever the timer expires.


4.8. The Minimum Congestion Window Is Two Packets
4.8. 最小輻輳ウィンドウは2つのパケット

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).


4.9. Handshake Packets Are Not Special
4.9. ハンドシェイクパケットは特別ではない

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.

TCP SYNまたはSYN-ACKパケットの損失を永続的な輻輳として扱い、輻輳ウィンドウを1パケットに短縮します。[RFC5681]を参照してください。QUICは、ハンドシェイクデータを含むパケットの損失を他の損失と同じで扱います。

5. Estimating the Round-Trip Time
5. 往復時間の推定

At a high level, an endpoint measures the time from when a packet was sent to when it is acknowledged as an RTT sample. The endpoint uses RTT samples and peer-reported host delays (see Section 13.2 of [QUIC-TRANSPORT]) to generate a statistical description of the network path's RTT. An endpoint computes the following three values for each path: the minimum value over a period of time (min_rtt), an exponentially weighted moving average (smoothed_rtt), and the mean deviation (referred to as "variation" in the rest of this document) in the observed RTT samples (rttvar).


5.1. Generating RTT Samples
5.1. RTTサンプルの生成

An endpoint generates an RTT sample on receiving an ACK frame that meets the following two conditions:


* the largest acknowledged packet number is newly acknowledged, and

* 最大の確認されたパケット番号は新しく認められ、

* at least one of the newly acknowledged packets was ack-eliciting.

* 新しく確認されたパケットのうちの少なくとも1つはACK誘発パケットでした。

The RTT sample, latest_rtt, is generated as the time elapsed since the largest acknowledged packet was sent:

RTTサンプル latest_rtt は、最大の承認されたパケットが送信されてから経過した時間として生成されます。

   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.


5.2. Estimating min_rtt
5.2. min_rttの推定

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.


5.3. Estimating smoothed_rtt and rttvar
5.3. smoothed_rttとrttvarの推定

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.


Therefore, when adjusting an RTT sample using peer-reported acknowledgment delays, an endpoint:


* MAY ignore the acknowledgment delay for Initial packets, since these acknowledgments are not delayed by the peer (Section 13.2.1 of [QUIC-TRANSPORT]);

* これらの確認応答はピアによって遅れていないため、初期パケットの確認遅延を無視することができます([QUIC-TRANSPORT]のセクション13.2.1)。

* SHOULD ignore the peer's max_ack_delay until the handshake is confirmed;

* ハンドシェイクが確認されるまで、ピアのmax_ack_delayを無視する必要があります。

* MUST use the lesser of the acknowledgment delay and the peer's max_ack_delay after the handshake is confirmed; and

* handshakeが確認された後に、肯定応答遅延のより少ないを使用する必要があります。そして

* 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.

* 結果の値がmin_rttより小さければ、RTTサンプルからの確認遅延を減算しないでください。これは、誤動作しているピアによるsmoothed_rttの過小評価を制限します。

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 and rttvar are initialized as follows, where kInitialRtt contains the initial RTT value:


   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].


On the first RTT sample after initialization, smoothed_rtt and rttvar are set as follows:


   smoothed_rtt = latest_rtt
   rttvar = latest_rtt / 2

On subsequent RTT samples, smoothed_rtt and rttvar evolve as follows:


   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
6. Loss Detection
6. 損失検出

QUIC senders use acknowledgments to detect lost packets and a PTO to ensure acknowledgments are received; see Section 6.2. This section provides a description of these algorithms.


If a packet is lost, the QUIC transport needs to recover from that loss, such as by retransmitting the data, sending an updated frame, or discarding the frame. For more information, see Section 13.3 of [QUIC-TRANSPORT].


Loss detection is separate per packet number space, unlike RTT measurement and congestion control, because RTT and congestion control are properties of the path, whereas loss detection also relies upon key availability.


6.1. Acknowledgment-Based Detection
6.1. 確認応答ベースの検出

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.

確認応答ベースの損失検出[RFC5681]、早期再送信[RFC5827]、前方承認[FACK]、SACK Loss Recovery [RFC6675]、およびRACK-TLP [RFC8985]の精神を実装しています。このセクションでは、これらのアルゴリズムがQUICでどのように実装されているかの概要を説明します。

A packet is declared lost if it meets all of the following conditions:


* The packet is unacknowledged, in flight, and was sent prior to an acknowledged packet.

* パケットは確認応答されておらず、飛行中であり、確認応答されたパケットの前に送信されました。

* 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).

* パケットは承認されたパケット(6.1.1項)の前にkpacketthresholdパケットに送信されたか、過去に十分な長さに送信されました(6.1.2項)。

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.


6.1.1. Packet Threshold
6.1.1. パケットしきい値

The RECOMMENDED initial value for the packet reordering threshold (kPacketThreshold) is 3, based on best practices for TCP loss detection [RFC5681] [RFC6675]. In order to remain similar to TCP, implementations SHOULD NOT use a packet threshold less than 3; see [RFC5681].

TCP損失検出のベストプラクティス[RFC5681] [RFC6675]に基づく、パケット並べ替え閾値(kPacketThreshold)の推奨初期値は3です。TCPと同様のままで、実装は3未満のパケットしきい値を使用しないでください。[RFC5681]を参照してください。

Some networks may exhibit higher degrees of packet reordering, causing a sender to detect spurious losses. Additionally, packet reordering could be more common with QUIC than TCP because network elements that could observe and reorder TCP packets cannot do that for QUIC and also because QUIC packet numbers are encrypted. Algorithms that increase the reordering threshold after spuriously detecting losses, such as RACK [RFC8985], have proven to be useful in TCP and are expected to be at least as useful in QUIC.


6.1.2. Time Threshold
6.1.2. 時間しきい値

Once a later packet within the same packet number space has been acknowledged, an endpoint SHOULD declare an earlier packet lost if it was sent a threshold amount of time in the past. To avoid declaring packets as lost too early, this time threshold MUST be set to at least the local timer granularity, as indicated by the kGranularity constant. The time threshold is:


   max(kTimeThreshold * max(smoothed_rtt, latest_rtt), kGranularity)

If packets sent prior to the largest acknowledged packet cannot yet be declared lost, then a timer SHOULD be set for the remaining time.


Using max(smoothed_rtt, latest_rtt) protects from the two following cases:

max(smoothed_rtt、latest_rtt) を使用すると、次の2つのケースから保護します。

* the latest RTT sample is lower than the smoothed RTT, perhaps due to reordering where the acknowledgment encountered a shorter path;

* 最新のRTTサンプルは、おそらく承認が短いパスに遭遇した並べ替えのために、平滑化されたRTTよりも低い。

* the latest RTT sample is higher than the smoothed RTT, perhaps due to a sustained increase in the actual RTT, but the smoothed RTT has not yet caught up.

* 最新のRTTサンプルは、おそらく実際のRTTの持続的な増加のために平滑化されたRTTよりも高いが、滑らかなRTTはまだ追いついていない。

The RECOMMENDED time threshold (kTimeThreshold), expressed as an RTT multiplier, is 9/8. The RECOMMENDED value of the timer granularity (kGranularity) is 1 millisecond.


Note: TCP's RACK [RFC8985] specifies a slightly larger threshold, equivalent to 5/4, for a similar purpose. Experience with QUIC shows that 9/8 works well.

注:TCPのRACK [RFC8985]は、同様の目的で、5/4に相当するわずかに大きいしきい値を指定しています。QUICの経験は、9/8がうまく機能することを示しています。

Implementations MAY experiment with absolute thresholds, thresholds from previous connections, adaptive thresholds, or the including of RTT variation. Smaller thresholds reduce reordering resilience and increase spurious retransmissions, and larger thresholds increase loss detection delay.


6.2. Probe Timeout
6.2. プローブタイムアウト

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].

QUICで使用されるPTOアルゴリズムは、TAIL損失プローブ[RFC8985]、RTO [RFC5681]、およびTCP [RFC5682]の信頼性機能を実装しています。タイムアウト計算はTCPのRTO期間[RFC6298]に基づいています。

6.2.1. Computing PTO
6.2.1. PTOの計算

When an ack-eliciting packet is transmitted, the sender schedules a timer for the PTO period as follows:


   PTO = smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay

The PTO period is the amount of time that a sender ought to wait for an acknowledgment of a sent packet. This time period includes the estimated network RTT (smoothed_rtt), the variation in the estimate (4*rttvar), and max_ack_delay, to account for the maximum time by which a receiver might delay sending an acknowledgment.

PTO期間は、送信者が送信されたパケットの確認応答を待つべきな時間です。この期間には、推定ネットワークRTT(smoothed_rtt)、推定値(4 * rttvar)、およびmax_ack_delayが含まれ、受信側が確認応答の送信を遅らせる可能性がある最大時間を考慮しています。

When the PTO is armed for Initial or Handshake packet number spaces, the max_ack_delay in the PTO period computation is set to 0, since the peer is expected to not delay these packets intentionally; see Section 13.2.1 of [QUIC-TRANSPORT].


The PTO period MUST be at least kGranularity to avoid the timer expiring immediately.


When ack-eliciting packets in multiple packet number spaces are in flight, the timer MUST be set to the earlier value of the Initial and Handshake packet number spaces.


An endpoint MUST NOT set its PTO timer for the Application Data packet number space until the handshake is confirmed. Doing so prevents the endpoint from retransmitting information in packets when either the peer does not yet have the keys to process them or the endpoint does not yet have the keys to process their acknowledgments. For example, this can happen when a client sends 0-RTT packets to the server; it does so without knowing whether the server will be able to decrypt them. Similarly, this can happen when a server sends 1-RTT packets before confirming that the client has verified the server's certificate and can therefore read these 1-RTT packets.

ハンドシェイクが確認されるまで、エンドポイントはアプリケーションデータパケット番号スペースのPTOタイマーを設定してはなりません。そうすることで、ピアのいずれかがそれらを処理するためのキーがまだある場合、またはエンドポイントがまだ承認を処理するためのキーを持っていないときに、エンドポイントがパケット内の情報を再送信するのを防ぎます。たとえば、クライアントが0-RTTパケットをサーバーに送信したときに発生する可能性があります。サーバーがそれらを復号化できるかどうかを知らずにそうします。同様に、これは、クライアントがサーバーの証明書を検証していることを確認する前にサーバーが1-RTTパケットを送信し、したがってこれらの1 RTTパケットを読み取ることができる場合に発生する可能性があります。

A sender SHOULD restart its PTO timer every time an ack-eliciting packet is sent or acknowledged, or when Initial or Handshake keys are discarded (Section 4.9 of [QUIC-TLS]). This ensures the PTO is always set based on the latest estimate of the RTT and for the correct packet across packet number spaces.


When a PTO timer expires, the PTO backoff MUST be increased, resulting in the PTO period being set to twice its current value. The PTO backoff factor is reset when an acknowledgment is received, except in the following case. A server might take longer to respond to packets during the handshake than otherwise. To protect such a server from repeated client probes, the PTO backoff is not reset at a client that is not yet certain that the server has finished validating the client's address. That is, a client does not reset the PTO backoff factor on receiving acknowledgments in Initial packets.


This exponential reduction in the sender's rate is important because consecutive PTOs might be caused by loss of packets or acknowledgments due to severe congestion. Even when there are ack-eliciting packets in flight in multiple packet number spaces, the exponential increase in PTO occurs across all spaces to prevent excess load on the network. For example, a timeout in the Initial packet number space doubles the length of the timeout in the Handshake packet number space.


The total length of time over which consecutive PTOs expire is limited by the idle timeout.


The PTO timer MUST NOT be set if a timer is set for time threshold loss detection; see Section 6.1.2. A timer that is set for time threshold loss detection will expire earlier than the PTO timer in most cases and is less likely to spuriously retransmit data.


6.2.2. Handshakes and New Paths
6.2.2. ハンドシェイクと新しい道

Resumed connections over the same network MAY use the previous connection's final smoothed RTT value as the resumed connection's initial RTT. When no previous RTT is available, the initial RTT SHOULD be set to 333 milliseconds. This results in handshakes starting with a PTO of 1 second, as recommended for TCP's initial RTO; see Section 2 of [RFC6298].


A connection MAY use the delay between sending a PATH_CHALLENGE and receiving a PATH_RESPONSE to set the initial RTT (see kInitialRtt in Appendix A.2) for a new path, but the delay SHOULD NOT be considered an RTT sample.


When the Initial keys and Handshake keys are discarded (see Section 6.4), any Initial packets and Handshake packets can no longer be acknowledged, so they are removed from bytes in flight. When Initial or Handshake keys are discarded, the PTO and loss detection timers MUST be reset, because discarding keys indicates forward progress and the loss detection timer might have been set for a now-discarded packet number space.

最初のキーとハンドシェイクキーが破棄されると(セクション6.4を参照)、最初のパケットとハンドシェイクのパケットは承認されなくなりますので、フライトのバイトから削除されます。初期キーまたはハンドシェイクキーを破棄すると、キーが廃棄されることを示し、損失検出タイマが現在廃棄されたパケット番号スペースに設定されている可能性があるため、PTOおよび損失検出タイマをリセットする必要があります。 Before Address Validation アドレス検証の前に

Until the server has validated the client's address on the path, the amount of data it can send is limited to three times the amount of data received, as specified in Section 8.1 of [QUIC-TRANSPORT]. If no additional data can be sent, the server's PTO timer MUST NOT be armed until datagrams have been received from the client because packets sent on PTO count against the anti-amplification limit.


When the server receives a datagram from the client, the amplification limit is increased and the server resets the PTO timer. If the PTO timer is then set to a time in the past, it is executed immediately. Doing so avoids sending new 1-RTT packets prior to packets critical to the completion of the handshake. In particular, this can happen when 0-RTT is accepted but the server fails to validate the client's address.


Since the server could be blocked until more datagrams are received from the client, it is the client's responsibility to send packets to unblock the server until it is certain that the server has finished its address validation (see Section 8 of [QUIC-TRANSPORT]). That is, the client MUST set the PTO timer if the client has not received an acknowledgment for any of its Handshake packets and the handshake is not confirmed (see Section 4.1.2 of [QUIC-TLS]), even if there are no packets in flight. When the PTO fires, the client MUST send a Handshake packet if it has Handshake keys, otherwise it MUST send an Initial packet in a UDP datagram with a payload of at least 1200 bytes.


6.2.3. Speeding up Handshake Completion
6.2.3. ハンドシェイク終了の高速化

When a server receives an Initial packet containing duplicate CRYPTO data, it can assume the client did not receive all of the server's CRYPTO data sent in Initial packets, or the client's estimated RTT is too small. When a client receives Handshake or 1-RTT packets prior to obtaining Handshake keys, it may assume some or all of the server's Initial packets were lost.


To speed up handshake completion under these conditions, an endpoint MAY, for a limited number of times per connection, send a packet containing unacknowledged CRYPTO data earlier than the PTO expiry, subject to the address validation limits in Section 8.1 of [QUIC-TRANSPORT]. Doing so at most once for each connection is adequate to quickly recover from a single packet loss. An endpoint that always retransmits packets in response to receiving packets that it cannot process risks creating an infinite exchange of packets.


Endpoints can also use coalesced packets (see Section 12.2 of [QUIC-TRANSPORT]) to ensure that each datagram elicits at least one acknowledgment. For example, a client can coalesce an Initial packet containing PING and PADDING frames with a 0-RTT data packet, and a server can coalesce an Initial packet containing a PING frame with one or more packets in its first flight.

エンドポイントは、各データグラムが少なくとも1回の確認応答を誘発するように、合体パケットを使用することもできます([QUIC-TRANSPORT]のセクション12.2参照)。例えば、クライアントは、PINGフレームを含む初期パケットと0 - RTTデータパケットを含む初期パケットを合体することができ、サーバはその最初のフライト内の1つまたは複数のパケットを含むPingフレームを含む初期パケットを合体することができる。

6.2.4. Sending Probe Packets
6.2.4. プローブパケットの送信

When a PTO timer expires, a sender MUST send at least one ack-eliciting packet in the packet number space as a probe. An endpoint MAY send up to two full-sized datagrams containing ack-eliciting packets to avoid an expensive consecutive PTO expiration due to a single lost datagram or to transmit data from multiple packet number spaces. All probe packets sent on a PTO MUST be ack-eliciting.


In addition to sending data in the packet number space for which the timer expired, the sender SHOULD send ack-eliciting packets from other packet number spaces with in-flight data, coalescing packets if possible. This is particularly valuable when the server has both Initial and Handshake data in flight or when the client has both Handshake and Application Data in flight because the peer might only have receive keys for one of the two packet number spaces.


If the sender wants to elicit a faster acknowledgment on PTO, it can skip a packet number to eliminate the acknowledgment delay.


An endpoint SHOULD include new data in packets that are sent on PTO expiration. Previously sent data MAY be sent if no new data can be sent. Implementations MAY use alternative strategies for determining the content of probe packets, including sending new or retransmitted data based on the application's priorities.


It is possible the sender has no new or previously sent data to send. As an example, consider the following sequence of events: new application data is sent in a STREAM frame, deemed lost, then retransmitted in a new packet, and then the original transmission is acknowledged. When there is no data to send, the sender SHOULD send a PING or other ack-eliciting frame in a single packet, rearming the PTO timer.


Alternatively, instead of sending an ack-eliciting packet, the sender MAY mark any packets still in flight as lost. Doing so avoids sending an additional packet but increases the risk that loss is declared too aggressively, resulting in an unnecessary rate reduction by the congestion controller.


Consecutive PTO periods increase exponentially, and as a result, connection recovery latency increases exponentially as packets continue to be dropped in the network. Sending two packets on PTO expiration increases resilience to packet drops, thus reducing the probability of consecutive PTO events.


When the PTO timer expires multiple times and new data cannot be sent, implementations must choose between sending the same payload every time or sending different payloads. Sending the same payload may be simpler and ensures the highest priority frames arrive first. Sending different payloads each time reduces the chances of spurious retransmission.


6.3. Handling Retry Packets
6.3. リトライパケットの処理

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.


6.4. Discarding Keys and Packet State
6.4. キーとパケット状態の破棄

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 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].


7. Congestion Control
7. 輻輳制御

This document specifies a sender-side congestion controller for QUIC similar to TCP NewReno [RFC6582].

このドキュメントでは、TCP Newreno [RFC6582]と同様のQUIC用の送信側輻輳コントローラを指定します。

The signals QUIC provides for congestion control are generic and are designed to support different sender-side algorithms. A sender can unilaterally choose a different algorithm to use, such as CUBIC [RFC8312].


If a sender uses a different controller than that specified in this document, the chosen controller MUST conform to the congestion control guidelines specified in Section 3.1 of [RFC8085].


Similar to TCP, packets containing only ACK frames do not count toward bytes in flight and are not congestion controlled. Unlike TCP, QUIC can detect the loss of these packets and MAY use that information to adjust the congestion controller or the rate of ACK-only packets being sent, but this document does not describe a mechanism for doing so.


The congestion controller is per path, so packets sent on other paths do not alter the current path's congestion controller, as described in Section 9.4 of [QUIC-TRANSPORT].


The algorithm in this document specifies and uses the controller's congestion window in bytes.


An endpoint MUST NOT send a packet if it would cause bytes_in_flight (see Appendix B.2) to be larger than the congestion window, unless the packet is sent on a PTO timer expiration (see Section 6.2) or when entering recovery (see Section 7.3.2).


7.1. Explicit Congestion Notification
7.1. 明示的な輻輳通知

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].

明示的な輻輳通知(ECN)[RFC3168] [RFC8311]をサポートするためにパスが検証されている場合、QUICはIPヘッダー内の輻輳経験豊富な(CE)コードポイントを輻輳のシグナルとして扱います。このドキュメントは、ピアポートされたECN-CEカウントが増加したときのエンドポイントの応答を指定します。[QUIC-TRANSPORT]の13.4.2項を参照してください。

7.2. Initial and Minimum Congestion Window
7.2. 初期および最小輻輳ウィンドウ

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.

最小輻輳ウィンドウは、損失に応じて輻輳ウィンドウが達成できる最小値です。ピアポートされたECN-CEカウントの増加、または永続的な輻輳があります。推奨値は2 * max_datagram_sizeです。

7.3. Congestion Control States
7.3. 輻輳制御状態

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


                    New path or      +------------+
               persistent congestion |   Slow     |
           (O)---------------------->|   Start    |
                                   Loss or |
                           ECN-CE increase |
    +------------+     Loss or       +------------+
    | Congestion |  ECN-CE increase  |  Recovery  |
    | Avoidance  |------------------>|   Period   |
    +------------+                   +------------+
              ^                            |
              |                            |
                 Acknowledgment of packet
                   sent during recovery

Figure 1: Congestion Control States and Transitions


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


7.3.1. Slow Start
7.3.1. 遅いスタート

A NewReno sender is in slow start any time the congestion window is below the slow start threshold. A sender begins in slow start because the slow start threshold is initialized to an infinite value.

Newreno Senderは、輻輳ウィンドウがスロースタートしきい値を下回っている時期を遅くします。スロースタートしきい値が無限値に初期化されるため、送信者はスロースタートで始まります。

While a sender is in slow start, the congestion window increases by the number of bytes acknowledged when each acknowledgment is processed. This results in exponential growth of the congestion window.


The sender MUST exit slow start and enter a recovery period when a packet is lost or when the ECN-CE count reported by its peer increases.


A sender reenters slow start any time the congestion window is less than the slow start threshold, which only occurs after persistent congestion is declared.


7.3.2. Recovery
7.3.2. 回復

A NewReno sender enters a recovery period when it detects the loss of a packet or when the ECN-CE count reported by its peer increases. A sender that is already in a recovery period stays in it and does not reenter it.

Newreno Senderは、パケットの損失を検出したとき、またはそのピアによって報告されたECN-CEカウントが増加するときに回復期間を入力します。復旧期間中にすでに復旧している送信者はそれにまとまり、それを再入力しません。

On entering a recovery period, a sender MUST set the slow start threshold to half the value of the congestion window when loss is detected. The congestion window MUST be set to the reduced value of the slow start threshold before exiting the recovery period.


Implementations MAY reduce the congestion window immediately upon entering a recovery period or use other mechanisms, such as Proportional Rate Reduction [PRR], to reduce the congestion window more gradually. If the congestion window is reduced immediately, a single packet can be sent prior to reduction. This speeds up loss recovery if the data in the lost packet is retransmitted and is similar to TCP as described in Section 5 of [RFC6675].


The recovery period aims to limit congestion window reduction to once per round trip. Therefore, during a recovery period, the congestion window does not change in response to new losses or increases in the ECN-CE count.


A recovery period ends and the sender enters congestion avoidance when a packet sent during the recovery period is acknowledged. This is slightly different from TCP's definition of recovery, which ends when the lost segment that started recovery is acknowledged [RFC5681].


7.3.3. Congestion Avoidance
7.3.3. 輻輳回避

A NewReno sender is in congestion avoidance any time the congestion window is at or above the slow start threshold and not in a recovery period.

Newreno Senderは、輻輳ウィンドウが遅いスタートしきい値以上であり、回復期間内ではなく、輻輳回避です。

A sender in congestion avoidance uses an Additive Increase Multiplicative Decrease (AIMD) approach that MUST limit the increase to the congestion window to at most one maximum datagram size for each congestion window that is acknowledged.


The sender exits congestion avoidance and enters a recovery period when a packet is lost or when the ECN-CE count reported by its peer increases.


7.4. Ignoring Loss of Undecryptable Packets
7.4. 復号化できないパケット損失の無視

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.


7.5. Probe Timeout
7.5. プローブタイムアウト

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.


7.6. Persistent Congestion
7.6. 永続的な輻輳

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


7.6.1. Duration
7.6.1. 期間

The persistent congestion duration is computed as follows:


   (smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay) *

Unlike the PTO computation in Section 6.2, this duration includes the max_ack_delay irrespective of the packet number spaces in which losses are established.


This duration allows a sender to send as many packets before establishing persistent congestion, including some in response to PTO expiration, as TCP does with Tail Loss Probes [RFC8985] and an RTO [RFC5681].

この期間により、TCPがTail Loss Probes [RFC8985]とRTO [RFC5681]を使用すると、PTOの有効期限に応じていくらかのものを含めて、永続的な輻輳を確立する前に送信者が多くのパケットを送信できます。

Larger values of kPersistentCongestionThreshold cause the sender to become less responsive to persistent congestion in the network, which can result in aggressive sending into a congested network. Too small a value can result in a sender declaring persistent congestion unnecessarily, resulting in reduced throughput for the sender.


The RECOMMENDED value for kPersistentCongestionThreshold is 3, which results in behavior that is approximately equivalent to a TCP sender declaring an RTO after two TLPs.


This design does not use consecutive PTO events to establish persistent congestion, since application patterns impact PTO expiration. For example, a sender that sends small amounts of data with silence periods between them restarts the PTO timer every time it sends, potentially preventing the PTO timer from expiring for a long period of time, even when no acknowledgments are being received. The use of a duration enables a sender to establish persistent congestion without depending on PTO expiration.


7.6.2. Establishing Persistent Congestion
7.6.2. 持続的輻輳の確立

A sender establishes persistent congestion after the receipt of an acknowledgment if two packets that are ack-eliciting are declared lost, and:


* across all packet number spaces, none of the packets sent between the send times of these two packets are acknowledged;

* すべてのパケット番号スペースでは、これら2つのパケットの送信時間の間に送信されたパケットはどれも認証されません。

* the duration between the send times of these two packets exceeds the persistent congestion duration (Section 7.6.1); and

* これら2つのパケットの送信時間間の期間は永続的な輻輳期間を超えています(7.6.1項)。そして

* a prior RTT sample existed when these two packets were sent.

* これら2つのパケットが送信されたときに以前のRTTサンプルが存在しました。

These two packets MUST be ack-eliciting, since a receiver is required to acknowledge only ack-eliciting packets within its maximum acknowledgment delay; see Section 13.2 of [QUIC-TRANSPORT].


The persistent congestion period SHOULD NOT start until there is at least one RTT sample. Before the first RTT sample, a sender arms its PTO timer based on the initial RTT (Section 6.2.2), which could be substantially larger than the actual RTT. Requiring a prior RTT sample prevents a sender from establishing persistent congestion with potentially too few probes.


Since network congestion is not affected by packet number spaces, persistent congestion SHOULD consider packets sent across packet number spaces. A sender that does not have state for all packet number spaces or an implementation that cannot compare send times across packet number spaces MAY use state for just the packet number space that was acknowledged. This might result in erroneously declaring persistent congestion, but it will not lead to a failure to detect persistent congestion.


When persistent congestion is declared, the sender's congestion window MUST be reduced to the minimum congestion window (kMinimumWindow), similar to a TCP sender's response on an RTO [RFC5681].

永続的な輻輳が宣言されると、RTO [RFC5681]でのTCP送信者の応答と同様に、送信者の輻輳ウィンドウを最小輻輳ウィンドウ(KminimumWindow)に縮小する必要があります。

7.6.3. Example
7.6.3. 例

The following example illustrates how a sender might establish persistent congestion. Assume:


   smoothed_rtt + max(4*rttvar, kGranularity) + max_ack_delay = 2
   kPersistentCongestionThreshold = 3

Consider the following sequence of events:


              | Time   | Action                            |
              | t=0    | Send packet #1 (application data) |
              | t=1    | Send packet #2 (application data) |
              | t=1.2  | Receive acknowledgment of #1      |
              | t=2    | Send packet #3 (application data) |
              | t=3    | Send packet #4 (application data) |
              | t=4    | Send packet #5 (application data) |
              | t=5    | Send packet #6 (application data) |
              | t=6    | Send packet #7 (application data) |
              | t=8    | Send packet #8 (PTO 1)            |
              | t=12   | Send packet #9 (PTO 2)            |
              | t=12.2 | Receive acknowledgment of #9      |

Table 1


Packets 2 through 8 are declared lost when the acknowledgment for packet 9 is received at "t = 12.2".

パケット2~8は、パケット9の確認応答が「T = 12.2」で受信されたときに失われた宣言されている。

The congestion period is calculated as the time between the oldest and newest lost packets: "8 - 1 = 7". The persistent congestion duration is "2 * 3 = 6". Because the threshold was reached and because none of the packets between the oldest and the newest lost packets were acknowledged, the network is considered to have experienced persistent congestion.

輻輳期間は、最も古くて最新の紛失したパケットの間の時間として計算されます。 "8 - 1 = 7"。永続的な輻輳期間は「2 * 3 = 6」です。しきい値に達し、最も古い失われたパケットと最新のパケットの間のパケットのどれもが確認されていないため、ネットワークは永続的な輻輳を経験したと考えられています。

While this example shows PTO expiration, they are not required for persistent congestion to be established.


7.7. Pacing
7.7. ペーシング

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.


Endpoints can implement pacing as they choose. A perfectly paced sender spreads packets exactly evenly over time. For a window-based congestion controller, such as the one in this document, that rate can be computed by averaging the congestion window over the RTT. Expressed as a rate in units of bytes per time, where congestion_window is in bytes:


   rate = N * congestion_window / smoothed_rtt

Or expressed as an inter-packet interval in units of time:


   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.


7.8. Underutilizing the Congestion Window
7.8. 輻輳ウィンドウを十分に活用していない

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].


8. Security Considerations
8. セキュリティに関する考慮事項
8.1. Loss and Congestion Signals
8.1. 損失と輻輳の信号

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.


8.2. Traffic Analysis
8.2. トラフィック分析

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.


8.3. Misreporting ECN Markings
8.3. ECNマーキングを誤動産する

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].


9. References
9. 参考文献
9.1. Normative References
9.1. 引用文献

[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure QUIC", RFC 9001, DOI 10.17487/RFC9001, May 2021, <>.

[QUIC-TLS] Thomson、M.、ED。S.ターナー、ed。、「TLSをセキュリティに使用する」、RFC 9001、DOI 10.17487 / RFC9001、2021年5月、<>。

[QUIC-TRANSPORT] Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based Multiplexed and Secure Transport", RFC 9000, DOI 10.17487/RFC9000, May 2021, <>.

[QUIC-TRANSPORT] Iyengar、J.、ED。そして、「Q. Thomson」、「QUIC:UDPベースの多重化および安全な輸送」、RFC 9000、DOI 10.17487 / RFC9000、<>。

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <>.

[RFC2119] BRADNER、S、「RFCSで使用するためのキーワード」、BCP 14、RFC 2119、DOI 10.17487 / RFC2119、1997年3月、<>。

[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, DOI 10.17487/RFC3168, September 2001, <>.

[RFC3168] Ramakrishnan、K.、Floyd、S.、およびD. Black、「IPへの明示的輻輳通知(ECN)の追加」、RFC 3168、DOI 10.17487 / RFC3168、2001年9月、<https:// www。>。

[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, March 2017, <>.

[RFC8085] eggert、L.、Fairhurst、G.、およびG.Shepherd、 "UDP使用ガイドライン"、BCP 145、RFC 8085、DOI 10.17487 / RFC8085、2017年3月、< info / rfc8085>。

[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017, <>.

[RFC8174] Leiba、B、「RFC 2119キーワードの大文字の曖昧さ」、BCP 14、RFC 8174、DOI 10.17487 / RFC8174、2017年5月、<>。

9.2. Informative References
9.2. 参考引用

[FACK] Mathis, M. and J. Mahdavi, "Forward acknowledgement: Refining TCP Congestion Control", ACM SIGCOMM Computer Communication Review, DOI 10.1145/248157.248181, August 1996, <>.

[FACK] Mathis、M.およびJ.Mahdavi、「前向き承認:TCP輻輳制御」、ACM SIGMOCMM Computer Compulment Review、DOI 10.1145 / 248157.248181、<>。

[PRR] Mathis, M., Dukkipati, N., and Y. Cheng, "Proportional Rate Reduction for TCP", RFC 6937, DOI 10.17487/RFC6937, May 2013, <>.

Mathis、M.、Dukkipati、N.、Y. Cheng、RFC 6937、DOI 10.17487 / RFC6937、2013年5月、<情報/ RFC6937>。

[RETRANSMISSION] Karn, P. and C. Partridge, "Improving Round-Trip Time Estimates in Reliable Transport Protocols", ACM Transactions on Computer Systems, DOI 10.1145/118544.118549, November 1991, <>.

[再送]カーン、P.およびC.パーリッジ、「信頼できる輸送プロトコルにおける往復時間推定の改善」、コンピュータシステム、DOI 10.1145 / 118544.118549、1991年11月、<>。

[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, DOI 10.17487/RFC2018, October 1996, <>.

[RFC2018] Mathis、M.、Mahdavi、J.、Floyd、S.、およびA. Romanow、「TCP選択認証オプション」、RFC 2018、DOI 10.17487 / RFC2018、<https:///>

[RFC3465] Allman, M., "TCP Congestion Control with Appropriate Byte Counting (ABC)", RFC 3465, DOI 10.17487/RFC3465, February 2003, <>.

[RFC3465] Allman、M.、「適切なバイトカウント(ABC)」、RFC 3465、DOI 10.17487 / RFC3465、<>。

[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, <>.

[RFC5681] Allman、M.、Paxson、V.およびE.Blanton、「TCP輻輳制御」、RFC 5681、DOI 10.17487 / RFC5681、2009年9月、<>。

[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata, "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious Retransmission Timeouts with TCP", RFC 5682, DOI 10.17487/RFC5682, September 2009, <>.

[RFC5682] Sarolahti、P.、Kojo、M.、Yamamoto、K.、およびM. HATA、「前方RTO回復(F-RTO):TCPでのスプリアス再送タイムアウトを検出するためのアルゴリズム、RFC 5682、DOI 10.17487/ RFC5682、2009年9月、<>。

[RFC5827] Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and P. Hurtig, "Early Retransmit for TCP and Stream Control Transmission Protocol (SCTP)", RFC 5827, DOI 10.17487/RFC5827, May 2010, <>.

[RFC5827] Allman、M.、Avrachenkov、K.、Ayesta、U.、Blanton、J.、およびP. Hurtig、「TCPおよびストリーム制御伝送プロトコル(SCTP)」、RFC 5827、DOI 10.17487 / RFC58272010年5月、<>。

[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, DOI 10.17487/RFC6298, June 2011, <>.

[RFC6298] Paxson、V.、Allman、M.、Chu、J.、およびM.Sargent、「コンピューティングTCPの再送信タイマー」、RFC 6298、DOI 10.17487 / RFC6298、2011年6月、<https:///>。

[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The NewReno Modification to TCP's Fast Recovery Algorithm", RFC 6582, DOI 10.17487/RFC6582, April 2012, <>.

[RFC6582] Henderson、T.、Floyd、S.、Gurtov、A.、およびY。西田、「TCPの高速回復アルゴリズムへのニューレノ修正」、RFC 6582、DOI 10.17487 / RFC6582、2012年4月、<>。

[RFC6675] Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M., and Y. Nishida, "A Conservative Loss Recovery Algorithm Based on Selective Acknowledgment (SACK) for TCP", RFC 6675, DOI 10.17487/RFC6675, August 2012, <>.

[RFC6675] Blanton、E.、Allman、M.、Wang、L.、Jarvinen、I.、Kojo、M.、Y. Nishida、「TCPのための選択認識(SACK)に基づく保守的な損失回復アルゴリズム」RFC 6675、DOI 10.17487 / RFC6675、2012年8月、<>。

[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis, "Increasing TCP's Initial Window", RFC 6928, DOI 10.17487/RFC6928, April 2013, <>.

[RFC6928] Chu、J.、Dukkipati、N.、Cheng、Y.、およびM. Mathis、「TCPの初期ウィンドウの増加」、RFC 6928、DOI 10.17487 / RFC6928、2013年4月、<>。

[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating TCP to Support Rate-Limited Traffic", RFC 7661, DOI 10.17487/RFC7661, October 2015, <>.

[RFC7661] FairHurst、G.、SathiaseLan、A.、R.Secchi、RFC 7661、DOI 10.17487 / RFC7661、2015年10月、<https:///www.rfc-編集者.ORG / INFO / RFC7661>。

[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018, <>.

[RFC8311]ブラック、D.、「明示的輻輳通知(ECN)実験(ECN)実験に関するリラックス制限」、RFC 8311、DOI 10.17487 / RFC8311、2018年1月、<>。

[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and R. Scheffenegger, "CUBIC for Fast Long-Distance Networks", RFC 8312, DOI 10.17487/RFC8312, February 2018, <>.

[RFC8312] Rhee、I.、Xu、L.、HA、S・、Zimmermann、A.、Heghert、L.、およびR.ScheffeNgger、RFC 8312、DOI 10.17487 / RFC83122018年2月、<>。

[RFC8511] Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst, "TCP Alternative Backoff with ECN (ABE)", RFC 8511, DOI 10.17487/RFC8511, December 2018, <>.

[RFC8511] Khademi、N.、Welzl、M.、Armitage、G.、およびG. FairHurst、「ECN(ABE)のTCP代替バックオフ、RFC 8511、DOI 10.17487 / RFC8511、2017年12月、<>。

[RFC8985] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The RACK-TLP Loss Detection Algorithm for TCP", RFC 8985, DOI 10.17487/RFC8985, February 2021, <>.

[RFC8985] Cheng、Y.、Cardwell、N.、Dukkipati、N.、およびP.JHA、「TCP用ラック-TLP損失検出アルゴリズム」、RFC 8985、DOI 10.17487 / RFC8985、2021年2月、<>。

Appendix A. Loss Recovery Pseudocode
付録A. 損失回復疑似コード

We now describe an example implementation of the loss detection mechanisms described in Section 6.


The pseudocode segments in this section are licensed as Code Components; see the copyright notice.


A.1. Tracking Sent Packets
A.1. トラッキング送信パケット

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.


A.1.1. Sent Packet Fields
A.1.1. パケットフィールドを送信しました

packet_number: The packet number of the sent packet.


ack_eliciting: A Boolean that indicates whether a packet is ack-eliciting. If true, it is expected that an acknowledgment will be received, though the peer could delay sending the ACK frame containing it by up to the max_ack_delay.


in_flight: A Boolean that indicates whether the packet counts toward bytes in flight.


sent_bytes: The number of bytes sent in the packet, not including UDP or IP overhead, but including QUIC framing overhead.


time_sent: The time the packet was sent.


A.2. Constants of Interest
A.2. 関心の定数

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.

kInitialRtt:RTTサンプルの前に使用されたRTTが撮影されます。6.2.2項で推奨される値は333 msです。

kPacketNumberSpace: An enum to enumerate the three packet number spaces:


   enum kPacketNumberSpace {
A.3. Variables of Interest
A.3. 関心の変数

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.


A.4. Initialization
A.4. 初期化

At the beginning of the connection, initialize the loss detection variables as follows:


   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
A.5. On Sending a Packet
A.5. パケットを送信する

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.


Pseudocode for OnPacketSent follows:


   OnPacketSent(packet_number, pn_space, ack_eliciting,
                in_flight, sent_bytes):
     sent_packets[pn_space][packet_number].packet_number =
     sent_packets[pn_space][packet_number].time_sent = now()
     sent_packets[pn_space][packet_number].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()
A.6. On Receiving a Datagram
A.6. データグラムを受信する

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.


Pseudocode for OnDatagramReceived follows:


     // If this datagram unblocks the server, arm the
     // PTO timer to avoid deadlock.
     if (server was at anti-amplification limit):
       if loss_detection_timer.timeout < now():
         // Execute PTO if it would have expired
         // while the amplification limit applied.
A.7. On Receiving an Acknowledgment
A.7. ACKの受信

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


Pseudocode for OnAckReceived and UpdateRtt follow:


     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
       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()):
     // 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 &&
       latest_rtt =
         now() - newly_acked_packets.largest().time_sent
     // Process ECN information if present.
     if (ACK frame contains ECN information):
         ProcessECN(ack, pn_space)
     lost_packets = DetectAndRemoveLostPackets(pn_space)
     if (!lost_packets.empty()):
     // Reset pto_count unless the client is unsure if
     // the server has validated the client's address.
     if (PeerCompletedAddressValidation()):
       pto_count = 0
     if (first_rtt_sample == 0):
       min_rtt = latest_rtt
       smoothed_rtt = latest_rtt
       rttvar = latest_rtt / 2
       first_rtt_sample = now()
     // 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
A.8. Setting the Loss Detection Timer
A.8. 損失検出タイマーの設定

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.


Pseudocode for SetLossDetectionTimer follows (where the "^" operator represents exponentiation):


     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
     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):
       if (has handshake keys):
         return (now() + duration), Handshake
         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):
       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
     // 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
     earliest_loss_time, _ = GetLossTimeAndSpace()
     if (earliest_loss_time != 0):
       // Time threshold loss detection.
     if (server is at anti-amplification limit):
       // The server's timer is not set if nothing can be sent. 
     if (no ack-eliciting packets in flight &&
       // 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.
     timeout, _ = GetPtoTimeAndSpace()
A.9. On Timeout
A.9. タイムアウトについて

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


Pseudocode for OnLossDetectionTimeout follows:


     earliest_loss_time, pn_space = GetLossTimeAndSpace()
     if (earliest_loss_time != 0):
       // Time threshold loss Detection
       lost_packets = DetectAndRemoveLostPackets(pn_space)
     if (no ack-eliciting packets in flight):
       // 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):
       // PTO. Send new data if available, else retransmit old data.
       // If neither is available, send a single PING frame.
       _, pn_space = GetPtoTimeAndSpace()
A.10. Detecting Lost Packets
A.10. 失われたパケットの検出

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.


Pseudocode for DetectAndRemoveLostPackets follows:


     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]):
       // 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):
         if (loss_time[pn_space] == 0):
           loss_time[pn_space] = unacked.time_sent + loss_delay
           loss_time[pn_space] = min(loss_time[pn_space],
                                     unacked.time_sent + loss_delay)
     return lost_packets
A.11. Upon Dropping Initial or Handshake Keys
A.11. 初期またはハンドシェイクキーを落とすと

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


Pseudocode for OnPacketNumberSpaceDiscarded follows:


     assert(pn_space != ApplicationData)
     // Reset the loss detection and PTO timer
     time_of_last_ack_eliciting_packet[pn_space] = 0
     loss_time[pn_space] = 0
     pto_count = 0
Appendix B. Congestion Control Pseudocode
付録B. 輻輳制御疑似コード

We now describe an example implementation of the congestion controller described in Section 7.


The pseudocode segments in this section are licensed as Code Components; see the copyright notice.


B.1. Constants of Interest
B.1. 関心の定数

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.


B.2. Variables of Interest
B.2. 関心の変数

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.

max_datagram_size:送信者の現在の最大ペイロードサイズ。これにはUDPまたはIPオーバーヘッドは含まれません。MAXデータグラムのサイズは、輻輳ウィンドウの計算に使用されます。エンドポイントは、そのパス最大伝送ユニット(PMTU; [QUICトランスポート]のセクション14.2参照)に基づいてこの変数の値を設定します。最小値1200バイトです。

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.


B.3. Initialization
B.3. 初期化

At the beginning of the connection, initialize the congestion control variables as follows:


   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
B.4. On Packet Sent
B.4. 送信されたパケットについて

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


     bytes_in_flight += sent_bytes
B.5. On Packet Acknowledgment
B.5. パケット承認について

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].


     return sent_time <= congestion_recovery_start_time
     for acked_packet in acked_packets:
     if (!acked_packet.in_flight):
     // 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())
     // Do not increase congestion window in recovery period.
     if (InCongestionRecovery(acked_packet.time_sent)):
     if (congestion_window < ssthresh):
       // Slow start.
       congestion_window += acked_packet.sent_bytes
       // Congestion avoidance.
       congestion_window +=
         max_datagram_size * acked_packet.sent_bytes
         / congestion_window
B.6. On New Congestion Event
B.6. 新しい輻輳イベントについて

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.


     // No reaction if already in a recovery period.
     if (InCongestionRecovery(sent_time)):
     // 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.
B.7. Process ECN Information
B.7. プロセスECN情報

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
B.8. On Packets Lost
B.8. 失われたパケットについて

This is invoked when DetectAndRemoveLostPackets deems packets lost.


     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):
     // 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):
     pc_lost = []
     for lost in lost_packets:
       if lost.time_sent > first_rtt_sample:
     if (InPersistentCongestion(pc_lost)):
       congestion_window = kMinimumWindow
       congestion_recovery_start_time = 0
B.9. Removing Discarded Packets from Bytes in Flight
B.9. 飛行中のバイトから破棄されたパケットを削除する

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


Pseudocode for RemoveFromBytesInFlight follows:


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



The IETF QUIC Working Group received an enormous amount of support from many people. The following people provided substantive contributions to this document:


* Alessandro Ghedini * Benjamin Saunders * Gorry Fairhurst * 山本和彦 (Kazu Yamamoto) * 奥 一穂 (Kazuho Oku) * Lars Eggert * Magnus Westerlund * Marten Seemann * Martin Duke * Martin Thomson * Mirja Kühlewind * Nick Banks * Praveen Balasubramanian

* Alessandro Ghedini * Benjamin Saunders * Gorry Fairhurst * 山本和彦 (Kazu Yamamoto) * 奥 一穂 (Kazuho Oku) * Lars Eggert * Magnus Westerlund * Marten Seemann * Martin Duke * Martin Thomson * Mirja Kühlewind * Nick Banks * Praveen Balasubramanian

Authors' Addresses


Jana Iyengar (editor) Fastly

Jana Iyengar(編集者)Fastly


Ian Swett (editor) Google

IAN SWETT(編集)Google