Internet Engineering Task Force (IETF)                         M. Allman
Request for Comments: 8961                                          ICSI
BCP: 233                                                   November 2020
Category: Best Current Practice
ISSN: 2070-1721

Requirements for Time-Based Loss Detection




Many protocols must detect packet loss for various reasons (e.g., to ensure reliability using retransmissions or to understand the level of congestion along a network path). While many mechanisms have been designed to detect loss, ultimately, protocols can only count on the passage of time without delivery confirmation to declare a packet "lost". Each implementation of a time-based loss detection mechanism represents a balance between correctness and timeliness; therefore, no implementation suits all situations. This document provides high-level requirements for time-based loss detectors appropriate for general use in unicast communication across the Internet. Within the requirements, implementations have latitude to define particulars that best address each situation.


Status of This Memo


This memo documents an Internet Best Current Practice.


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 BCPs 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。BCPの詳細情報はRFC 7841のセクション2で入手できます。

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) 2020 IETF Trust and the persons identified as the document authors. All rights reserved.

Copyright(C)2020 IETFの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
     1.1.  Terminology
   2.  Context
   3.  Scope
   4.  Requirements
   5.  Discussion
   6.  Security Considerations
   7.  IANA Considerations
   8.  References
     8.1.  Normative References
     8.2.  Informative References
   Author's Address
1. Introduction
1. はじめに

As a network of networks, the Internet consists of a large variety of links and systems that support a wide variety of tasks and workloads. The service provided by the network varies from best-effort delivery among loosely connected components to highly predictable delivery within controlled environments (e.g., between physically connected nodes, within a tightly controlled data center). Each path through the network has a set of path properties, e.g., available capacity, delay, and packet loss. Given the range of networks that make up the Internet, these properties range from largely static to highly dynamic.


This document provides guidelines for developing an understanding of one path property: packet loss. In particular, we offer guidelines for developing and implementing time-based loss detectors that have been gradually learned over the last several decades. We focus on the general case where the loss properties of a path are (a) unknown a priori and (b) dynamically varying over time. Further, while there are numerous root causes of packet loss, we leverage the conservative notion that loss is an implicit indication of congestion [RFC5681]. While this stance is not always correct, as a general assumption it has historically served us well [Jac88]. As we discuss further in Section 2, the guidelines in this document should be viewed as a general default for unicast communication across best-effort networks and not as optimal -- or even applicable -- for all situations.


Given that packet loss is routine in best-effort networks, loss detection is a crucial activity for many protocols and applications and is generally undertaken for two major reasons:


(1) Ensuring reliable data delivery

(1) 信頼できるデータ配信を確実にする

This requires a data sender to develop an understanding of which transmitted packets have not arrived at the receiver. This knowledge allows the sender to retransmit missing data.


(2) Congestion control

(2) 渋滞管理

As we mention above, packet loss is often taken as an implicit indication that the sender is transmitting too fast and is overwhelming some portion of the network path. Data senders can therefore use loss to trigger transmission rate reductions.


Various mechanisms are used to detect losses in a packet stream. Often, we use continuous or periodic acknowledgments from the recipient to inform the sender's notion of which pieces of data are missing. However, despite our best intentions and most robust mechanisms, we cannot place ultimate faith in receiving such acknowledgments but can only truly depend on the passage of time. Therefore, our ultimate backstop to ensuring that we detect all loss is a timeout. That is, the sender sets some expectation for how long to wait for confirmation of delivery for a given piece of data. When this time period passes without delivery confirmation, the sender concludes the data was lost in transit.


The specifics of time-based loss detection schemes represent a tradeoff between correctness and responsiveness. In other words, we wish to simultaneously:


* wait long enough to ensure the detection of loss is correct, and

* 損失の検出が正しいことを保証するのに十分な長さを待つ

* minimize the amount of delay we impose on applications (before repairing loss) and the network (before we reduce the congestion).

* 遅延量を最小限に抑える(損失を修復する前)、ネットワーク(渋滞を減らす前に)に課す遅延量を最小限に抑えます。

Serving both of these goals is difficult, as they pull in opposite directions [AP99]. By not waiting long enough to accurately determine a packet has been lost, we may provide a needed retransmission in a timely manner but risk both sending unnecessary ("spurious") retransmissions and needlessly lowering the transmission rate. By waiting long enough that we are unambiguously certain a packet has been lost, we cannot repair losses in a timely manner and we risk prolonging network congestion.


Many protocols and applications -- such as TCP [RFC6298], SCTP [RFC4960], and SIP [RFC3261] -- use their own time-based loss detection mechanisms. At this point, our experience leads to a recognition that often specific tweaks that deviate from standardized time-based loss detectors do not materially impact network safety with respect to congestion control [AP99]. Therefore, in this document we outline a set of high-level, protocol-agnostic requirements for time-based loss detection. The intent is to provide a safe foundation on which implementations have the flexibility to instantiate mechanisms that best realize their specific goals.

TCP [RFC6298]、SCTP [RFC4960]、およびSIP [RFC3261]のような多くのプロトコルとアプリケーション - 独自の時間ベースの損失検出メカニズムを使用してください。この時点で、私たちの経験は、標準化された時間ベースの損失検出器から逸脱する頻繁な調整が輻輳制御に関するネットワークの安全性に大きな影響を与えないという認識につながります[AP99]。したがって、この文書では、時間ベースの損失検出のための高レベルのプロトコルアンジネック要件のセットを概説します。意図は、その実装がそれらの特定の目標を最もよく実現するメカニズムをインスタンス化するための柔軟性を持つ安全な基礎を提供することです。

1.1. Terminology
1.1. 用語

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。

2. Context
2. 環境

This document is different from the way we ideally like to engineer systems. Usually, we strive to understand high-level requirements as a starting point. We then methodically engineer specific protocols, algorithms, and systems that meet these requirements. Within the IETF standards process, we have derived many time-based loss detection schemes without the benefit of some over-arching requirements document -- because we had no idea how to write such a document! Therefore, we made the best specific decisions we could in response to specific needs.

この文書は、エンジニアシステムを理想的にする方法とは異なります。通常、私たちは高水準の要件を出発点として理解するよう努めています。その後、これらの要件を満たす特定のプロトコル、アルゴリズム、およびシステムが専門的にエンジニアリングされます。IETF規格プロセス内では、いくつかのオーバーアーチリング要件文書の利点を持たずに多くの時間ベースの損失検出スキームを導き出しました - そのような文書を書く方法はわかりませんでした!したがって、私たちは特定のニーズに応えて、私たちが可能な限り最高の特定の決定を下しました。

At this point, however, the community's experience has matured to the point where we can define a set of general, high-level requirements for time-based loss detection schemes. We now understand how to separate the strategies these mechanisms use that are crucial for network safety from those small details that do not materially impact network safety. The requirements in this document may not be appropriate in all cases. In particular, the guidelines in Section 4 are concerned with the general case, but specific situations may allow for more flexibility in terms of loss detection because specific facets of the environment are known (e.g., when operating over a single physical link or within a tightly controlled data center). Therefore, variants, deviations, or wholly different time-based loss detectors may be necessary or useful in some cases. The correct way to view this document is as the default case and not as one-size-fits-all guidance that is optimal in all cases.


Adding a requirements umbrella to a body of existing specifications is inherently messy and we run the risk of creating inconsistencies with both past and future mechanisms. Therefore, we make the following statements about the relationship of this document to past and future specifications:


* This document does not update or obsolete any existing RFC. These previous specifications -- while generally consistent with the requirements in this document -- reflect community consensus, and this document does not change that consensus.

* この文書は既存のRFCを更新または時代遅れにしません。これらの以前の仕様は、一般的にこの文書の要件と一致していますが、コミュニティコンセンサスを反映しており、この文書はそのコンセンサスを変更しません。

* The requirements in this document are meant to provide for network safety and, as such, SHOULD be used by all future time-based loss detection mechanisms.

* この文書の要件は、ネットワークの安全性を提供することを意図しており、そのように、すべての将来の時間ベースの損失検出メカニズムによって使用されるべきである。

* The requirements in this document may not be appropriate in all cases; therefore, deviations and variants may be necessary in the future (hence the "SHOULD" in the last bullet). However, inconsistencies MUST be (a) explained and (b) gather consensus.

* この文書の要件はすべての場合において適切ではないかもしれません。したがって、将来的には逸脱や変形が必要になる可能性があります(したがって、「最後の箇条書き」の「「はるい」)。ただし、矛盾は説明されている(a)、(b)コンセンサスを集める必要があります。

3. Scope
3. 範囲

The principles we outline in this document are protocol-agnostic and widely applicable. We make the following scope statements about the application of the requirements discussed in Section 4:


(S.1) While there are a bevy of uses for timers in protocols -- from rate-based pacing to connection failure detection and beyond -- this document is focused only on loss detection.


(S.2) The requirements for time-based loss detection mechanisms in this document are for the primary or "last resort" loss detection mechanism, whether the mechanism is the sole loss repair strategy or works in concert with other mechanisms.


While a straightforward time-based loss detector is sufficient for simple protocols like DNS [RFC1034] [RFC1035], more complex protocols often use more advanced loss detectors to aid performance. For instance, TCP and SCTP have methods to detect (and repair) loss based on explicit endpoint state sharing [RFC2018] [RFC4960] [RFC6675]. Such mechanisms often provide more timely and precise loss detection than time-based loss detectors. However, these mechanisms do not obviate the need for a "retransmission timeout" or "RTO" because, as we discuss in Section 1, only the passage of time can ultimately be relied upon to detect loss. In other words, we ultimately cannot count on acknowledgments to arrive at the data sender to indicate which packets never arrived at the receiver. In cases such as these, we need a time-based loss detector to function as a "last resort".

DNS [RFC1034] [RFC1035]のような単純なプロトコルには、直接的な時間ベースの損失検出器が十分であるが、より複雑なプロトコルは、パフォーマンスを支援するためにより高度な損失検出器を使用することが多い。たとえば、TCPとSCTPは、明示的なエンドポイント状態の共有[RFC2018] [RFC4960] [RFC6675]に基づいて損失を検出(および修理)する方法を持っています。そのようなメカニズムは、時間ベースの損失検出器よりもタイムリーで正確な損失検出を提供することが多い。しかしながら、これらのメカニズムは「再送信タイムアウト」または「RTO」の必要性を排除するものではないので、セクション1で説明したので、時間の経過のみが最終的に損失を検出することに頼ることができるからである。言い換えれば、我々は最終的にはデータ送信者に到着するために承認を頼りにすることはできません。これらのような場合には、「最後のリゾート」として機能する時間ベースの損失検出器が必要です。

Also, note that some recent proposals have incorporated time as a component of advanced loss detection methods either as an aggressive first loss detector in certain situations or in conjunction with endpoint state sharing [DCCM13] [CCDJ20] [IS20]. While these mechanisms can aid timely loss recovery, the protocol ultimately leans on another more conservative timer to ensure reliability when these mechanisms break down. The requirements in this document are only directly applicable to last-resort loss detection. However, we expect that many of the requirements can serve as useful guidelines for more aggressive non-last-resort timers as well.

また、いくつかの最近の提案は、特定の状況での積極的な第1の損失検出器として、またはエンドポイント状態の共有[DCDJ20] [IS20]と共に、高度な損失検出方法の構成要素として時間を組み込んでいます。これらのメカニズムはタイムリーな損失回復を助けることができるが、プロトコルは最終的には別のより保守的なタイマーに陥りながら、これらのメカニズムが故障したときに信頼性を確保する。この文書の要件は、最後のリゾート損失検出にのみ直接適用されます。しかし、多くの要件は、より積極的な非最後のリゾートタイマーのための有用なガイドラインとして役立つことがあります。

(S.3) The requirements in this document apply only to endpoint-to-endpoint unicast communication. Reliable multicast (e.g., [RFC5740]) protocols are explicitly outside the scope of this document.


Protocols such as SCTP [RFC4960] and Multipath TCP (MP-TCP) [RFC6182] that communicate in a unicast fashion with multiple specific endpoints can leverage the requirements in this document provided they track state and follow the requirements for each endpoint independently. That is, if host A communicates with addresses B and C, A needs to use independent time-based loss detector instances for traffic sent to B and C.

複数の特定のエンドポイントとユニキャストファッションで通信するSCTP [RFC4960]とマルチパスTCP(MP-TCP)[RFC6182]などのプロトコルは、この文書内の要件を活用し、それらが状態を追跡し、各エンドポイントの要件に応じて独立して要件を守ることができます。すなわち、ホストAがアドレスBおよびCと通信する場合、AおよびCに送信されたトラフィックに対して独立した時間ベースの損失検出器インスタンスを使用する必要がある。

(S.4) There are cases where state is shared across connections or flows (e.g., [RFC2140] and [RFC3124]). State pertaining to time-based loss detection is often discussed as sharable. These situations raise issues that the simple flow-oriented time-based loss detection mechanism discussed in this document does not consider (e.g., how long to preserve state between connections). Therefore, while the general principles given in Section 4 are likely applicable, sharing time-based loss detection information across flows is outside the scope of this document.


4. Requirements
4. 要件

We now list the requirements that apply when designing primary or last-resort time-based loss detection mechanisms. For historical reasons and ease of exposition, we refer to the time between sending a packet and determining the packet has been lost due to lack of delivery confirmation as the "retransmission timeout" or "RTO". After the RTO passes without delivery confirmation, the sender may safely assume the packet is lost. However, as discussed above, the detected loss need not be repaired (i.e., the loss could be detected only for congestion control and not reliability purposes).


(1) As we note above, loss detection happens when a sender does not receive delivery confirmation within some expected period of time. In the absence of any knowledge about the latency of a path, the initial RTO MUST be conservatively set to no less than 1 second.

(1) 上記に注意しても、損失検知は、送信者が期待された期間内に配達確認を受けない場合に発生します。パスの待ち時間に関する知識がない場合、初期RTOは10秒以上に保守的に設定されている必要があります。

Correctness is of the utmost importance when transmitting into a network with unknown properties because:


* Premature loss detection can trigger spurious retransmits that could cause issues when a network is already congested.

* 早期損失検出は、ネットワークがすでに混雑しているときに問題を引き起こす可能性があるスプリアス再送信を引き起こす可能性があります。

* Premature loss detection can needlessly cause congestion control to dramatically lower the sender's allowed transmission rate, especially since the rate is already likely low at this stage of the communication. Recovering from such a rate change can take a relatively long time.

* 早期損失検出は、特にこの通信段階でレートがすでに低いほど低い伝送速度を劇的に低下させることが不必要に輻輳制御を劇的に低下させる可能性があります。そのようなレート変化からの回復は比較的長い時間をかける可能性があります。

* Finally, as discussed below, sometimes using time-based loss detection and retransmissions can cause ambiguities in assessing the latency of a network path. Therefore, it is especially important for the first latency sample to be free of ambiguities such that there is a baseline for the remainder of the communication.

* 最後に、以下で論じるように、時には時間ベースの損失検出を使用し、再送信を使用すると、ネットワーク経路の待ち時間を評価する際にあいまいさが発生する可能性があります。したがって、第1の待ち時間サンプルが、残りの通信のためのベースラインがあるように、曖昧さをなくすことが特に重要である。

The specific constant (1 second) comes from the analysis of Internet round-trip times (RTTs) found in Appendix A of [RFC6298].


(2) We now specify four requirements that pertain to setting an expected time interval for delivery confirmation.

(2) 配信確認の期待時間間隔を設定するための4つの要件を指定しました。

Often, measuring the time required for delivery confirmation is framed as assessing the RTT of the network path. The RTT is the minimum amount of time required to receive delivery confirmation and also often follows protocol behavior whereby acknowledgments are generated quickly after data arrives. For instance, this is the case for the RTO used by TCP [RFC6298] and SCTP [RFC4960]. However, this is somewhat misleading, and the expected latency is better framed as the "feedback time" (FT). In other words, the expectation is not always simply a network property; it can include additional time before a sender should reasonably expect a response.

多くの場合、配達確認に必要な時間を測定することは、ネットワーク経路のRTTを評価するものとして囲まれています。RTTは、配送確認を受信するのに必要な最小時間、またデータが到着した後に承認が迅速に生成されるプロトコル動作に従うことが多い。たとえば、これはTCP [RFC6298]とSCTP [RFC4960]で使用されるRTOの場合です。ただし、これはやや誤解を招くようなもので、期待された待ち時間は「フィードバック時間」(FT)と同じように囲まれています。言い換えれば、期待は必ずしも単にネットワークプロパティではありません。送信者が応答を合理的に期待する必要がある前の追加の時間を含めることができます。

For instance, consider a UDP-based DNS request from a client to a recursive resolver [RFC1035]. When the request can be served from the resolver's cache, the feedback time (FT) likely well approximates the network RTT between the client and resolver. However, on a cache miss, the resolver will request the needed information from one or more authoritative DNS servers, which will non-trivially increase the FT compared to the network RTT between the client and resolver.


Therefore, we express the requirements in terms of FT. Again, for ease of exposition, we use "RTO" to indicate the interval between a packet transmission and the decision that the packet has been lost, regardless of whether the packet will be retransmitted.

したがって、FTの観点から要件を表します。また、博覧会を使用して、パケット送信とパケットが再送信されるかどうかにかかわらず、パケット送信とパケットが失われたという決定を示すために "RTO"を使用します。

(a) The RTO SHOULD be set based on multiple observations of the FT when available.

(a) RTOは、利用可能なときのFTの複数の観測値に基づいて設定する必要があります。

In other words, the RTO should represent an empirically derived reasonable amount of time that the sender should wait for delivery confirmation before deciding the given data is lost. Network paths are inherently dynamic; therefore, it is crucial to incorporate multiple recent FT samples in the RTO to take into account the delay variation across time.


For example, TCP's RTO [RFC6298] would satisfy this requirement due to its use of an exponentially weighted moving average (EWMA) to combine multiple FT samples into a "smoothed RTT". In the name of conservativeness, TCP goes further to also include an explicit variance term when computing the RTO.

たとえば、TCPのRTO [RFC6298]は、指数関数的に重み付けされた移動平均(EWMA)を使用して、複数のFTサンプルを「平滑化されたRTT」に組み合わせることで、この要件を満たします。保守性の名数では、TCPはさらにRTOを計算するときに明示的な分散用語を含むようになります。

While multiple FT samples are crucial for capturing the delay dynamics of a path, we explicitly do not tightly specify the process -- including the number of FT samples to use and how/when to age samples out of the RTO calculation -- as the particulars could depend on the situation and/or goals of each specific loss detector.


Finally, FT samples come from packet exchanges between peers. We encourage protocol designers -- especially for new protocols -- to strive to ensure the feedback is not easily spoofable by on- or off-path attackers such that they can perturb a host's notion of the FT. Ideally, all messages would be cryptographically secure, but given that this is not always possible -- especially in legacy protocols -- using a healthy amount of randomness in the packets is encouraged.

最後に、FTサンプルはピア間のパケット交換から来ます。プロトコル設計者 - 特に新しいプロトコルのために、フィードバックが、FTのホストの概念を乱すことができるようにフィードバックが容易になりやすいことを保証することを努める。理想的には、すべてのメッセージは暗号的に安全であると考えられますが、これは必ずしも可能ではないことを考えると、特にレガシープロトコルでは - パケット内の健全な量のランダム性を使用することが奨励されます。

(b) FT observations SHOULD be taken and incorporated into the RTO at least once per RTT or as frequently as data is exchanged in cases where that happens less frequently than once per RTT.

(b) FT観測は、少なくともRTTに1回、またはデータが頻繁に発生した場合にはRTOに1回、または頻繁に頻繁に行われる必要があります。

Internet measurements show that taking only a single FT sample per TCP connection results in a relatively poorly performing RTO mechanism [AP99], hence this requirement that the FT be sampled continuously throughout the lifetime of communication.


As an example, TCP takes an FT sample roughly once per RTT, or, if using the timestamp option [RFC7323], on each acknowledgment arrival. [AP99] shows that both these approaches result in roughly equivalent performance for the RTO estimator.


(c) FT observations MAY be taken from non-data exchanges.

(c) FT観察は、非データ交換から取られることがあります。

Some protocols use non-data exchanges for various reasons, e.g., keepalives, heartbeats, and control messages. To the extent that the latency of these exchanges mirrors data exchange, they can be leveraged to take FT samples within the RTO mechanism. Such samples can help protocols keep their RTO accurate during lulls in data transmission. However, given that these messages may not be subject to the same delays as data transmission, we do not take a general view on whether this is useful or not.


(d) An RTO mechanism MUST NOT use ambiguous FT samples.

(d) RTOメカニズムは、あいまいなFTサンプルを使用してはいけません。

Assume two copies of some packet X are transmitted at times t0 and t1. Then, at time t2, the sender receives confirmation that X in fact arrived. In some cases, it is not clear which copy of X triggered the confirmation; hence, the actual FT is either t2-t1 or t2-t0, but which is a mystery. Therefore, in this situation, an implementation MUST NOT use either version of the FT sample and hence not update the RTO (as discussed in [KP87] and [RFC6298]).


There are cases where two copies of some data are transmitted in a way whereby the sender can tell which is being acknowledged by an incoming ACK. For example, TCP's timestamp option [RFC7323] allows for packets to be uniquely identified and hence avoid the ambiguity. In such cases, there is no ambiguity and the resulting samples can update the RTO.


(3) Loss detected by the RTO mechanism MUST be taken as an indication of network congestion and the sending rate adapted using a standard mechanism (e.g., TCP collapses the congestion window to one packet [RFC5681]).

(3) RTOメカニズムによって検出された損失は、ネットワークの輻輳の表示と標準的なメカニズム(例えば、TCPが輻輳ウィンドウを1つのパケット(RFC5681]に崩壊させる)として適応された送信速度ととして取らなければなりません。

This ensures network safety.


An exception to this rule is if an IETF standardized mechanism determines that a particular loss is due to a non-congestion event (e.g., packet corruption). In such a case, a congestion control action is not required. Additionally, congestion control actions taken based on time-based loss detection could be reversed when a standard mechanism post facto determines that the cause of the loss was not congestion (e.g., [RFC5682]).


(4) Each time the RTO is used to detect a loss, the value of the RTO MUST be exponentially backed off such that the next firing requires a longer interval. The backoff SHOULD be removed after either (a) the subsequent successful transmission of non-retransmitted data, or (b) an RTO passes without detecting additional losses. The former will generally be quicker. The latter covers cases where loss is detected but not repaired.

(4) RTOが損失を検出するために使用されるたびに、RTOの値は次の発射が長い間隔を必要とするように指数関数的に後退させる必要があります。バックオフは(a)後に再送信されていないデータの送信を成功させた後、または(b)追加の損失を検出せずにパスを通過する(a)の後に除去する必要があります。前者は一般的に速くなります。後者は損失が検出されたが修復されない場合をカバーする。

A maximum value MAY be placed on the RTO. The maximum RTO MUST NOT be less than 60 seconds (as specified in [RFC6298]).


This ensures network safety.


As with guideline (3), an exception to this rule exists if an IETF standardized mechanism determines that a particular loss is not due to congestion.


5. Discussion
5. 考察

We note that research has shown the tension between the responsiveness and correctness of time-based loss detection seems to be a fundamental tradeoff in the context of TCP [AP99]. That is, making the RTO more aggressive (e.g., via changing TCP's exponentially weighted moving average (EWMA) gains, lowering the minimum RTO, etc.) can reduce the time required to detect actual loss. However, at the same time, such aggressiveness leads to more cases of mistakenly declaring packets lost that ultimately arrived at the receiver. Therefore, being as aggressive as the requirements given in the previous section allow in any particular situation may not be the best course of action because detecting loss, even if falsely, carries a requirement to invoke a congestion response that will ultimately reduce the transmission rate.

研究は、TCP [AP99]の文脈では、時間ベースの損失検出の応答性と正確さの間の緊張を示していることに注意しています。すなわち、RTOをより積極的にする(例えば、TCPの指数関数的に重み付け移動平均(EWMA)利得を介して、最小RTOなどを介して)、実際の損失を検出するのに必要な時間を短縮することができる。しかしながら、同時に、そのような攻撃性は、最終的に受信者に到着したパケットが誤って宣言されている誤って宣言された場合のより多くのケースをもたらす。したがって、前のセクションで与えられた要件が、誤って特定の状況で与えられた要件を考慮して、最終的に伝送速度を短縮するための輻輳応答を呼び出す必要があるため、特定の状況では最高の行動方針ではない可能性があります。

While the tradeoff between responsiveness and correctness seems fundamental, the tradeoff can be made less relevant if the sender can detect and recover from mistaken loss detection. Several mechanisms have been proposed for this purpose, such as Eifel [RFC3522], Forward RTO-Recovery (F-RTO) [RFC5682], and Duplicate Selective Acknowledgement (DSACK) [RFC2883] [RFC3708]. Using such mechanisms may allow a data originator to tip towards being more responsive without incurring (as much of) the attendant costs of mistakenly declaring packets to be lost.

応答性と正当性の間のトレードオフは基本的に思われますが、送信者が誤った損失の検出から検出して回復できる場合、トレードオフは関連性が低くなります。EIFEL [RFC3522]、前方RTO回復(F-RTO)[RFC5682]、および重複選択確認(DSACK)[RFC2883] [RFC2883]のようないくつかのメカニズムが提案されています。そのようなメカニズムを使用することは、誤って宣言されるべきパケットを誤って宣言することの偶然のコストが存在することなく、データ発信者がより敏感であることをより答えることを可能にし得る。

Also, note that, in addition to the experiments discussed in [AP99], the Linux TCP implementation has been using various non-standard RTO mechanisms for many years seemingly without large-scale problems (e.g., using different EWMA gains than specified in [RFC6298]). Further, a number of TCP implementations use a steady-state minimum RTO that is less than the 1 second specified in [RFC6298]. While the implication of these deviations from the standard may be more spurious retransmits (per [AP99]), we are aware of no large-scale network safety issues caused by this change to the minimum RTO. This informs the guidelines in the last section (e.g., there is no minimum RTO specified).

また、[AP99]で説明した実験に加えて、Linux TCP実装は、大規模な問題なしに、さまざまな数年間、さまざまな非標準RTOメカニズムを使用しています(たとえば、[RFC6298)。])。さらに、いくつかのTCP実装は、[RFC6298]で指定された1秒未満の定常状態最小RTOを使用する。これらの偏差を標準からの影響を受けている可能性があるが、よりスプリアスの再送信(AP99)は、この変更によって最小RTOへの変更によって引き起こされる大規模なネットワークの安全性の問題を認識しています。これにより、最後のセクション(例えば、最小RTOが指定されていない)のガイドラインに通知します。

Finally, we note that while allowing implementations to be more aggressive could in fact increase the number of needless retransmissions, the above requirements fail safely in that they insist on exponential backoff and a transmission rate reduction. Therefore, providing implementers more latitude than they have traditionally been given in IETF specifications of RTO mechanisms does not somehow open the flood gates to aggressive behavior. Since there is a downside to being aggressive, the incentives for proper behavior are retained in the mechanism.


6. Security Considerations
6. セキュリティに関する考慮事項

This document does not alter the security properties of time-based loss detection mechanisms. See [RFC6298] for a discussion of these within the context of TCP.


7. IANA Considerations
7. IANAの考慮事項

This document has no IANA actions.


8. References
8. 参考文献
8.1. Normative References
8.1. 引用文献

[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、<>。

[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、<>。

8.2. Informative References
8.2. 参考引用

[AP99] Allman, M. and V. Paxson, "On Estimating End-to-End Network Path Properties", Proceedings of the ACM SIGCOMM Technical Symposium, September 1999.

[AP99] Allman、M.およびV. Paxson、「エンドツーエンドネットワークパスのプロパティの見積もり」、1999年9月のACM SIGMOMM技術シンポジウムの手続き。

[CCDJ20] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The RACK-TLP loss detection algorithm for TCP", Work in Progress, Internet-Draft, draft-ietf-tcpm-rack-13, 2 November 2020, <>.

[CCDJ20] 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」、進行中の作業、インターネットドラフト、ドラフト-TCPMラック2020年11月2日、<>。

[DCCM13] Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis, "Tail Loss Probe (TLP): An Algorithm for Fast Recovery of Tail Losses", Work in Progress, Internet-Draft, draft-dukkipati-tcpm-tcp-loss-probe-01, 25 February 2013, <>.

[DCCM13] Dukkipati、N.、Cardwell、N.、Cheng、Y.、およびM Mathis、「テールロスプローブ(TLP):テールロスの迅速な回復のためのアルゴリズム」、進行中、インターネットドラフト、ドラフト-dukkipati-tcpm-tcp-loss-probe-01,2013、<>。

[IS20] Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection and Congestion Control", Work in Progress, Internet-Draft, draft-ietf-quic-recovery-32, 20 October 2020, <>.

[IS20] Iyengar、J.、ED。そしてI.Swett、Ed。、「QUICの喪失検知と輻輳制御」、進行中の作業、インターネットドラフト、ドラフト - IETF-QUIC-Recovery-32,20、< / froms-ietf-quic-recovery-32>。

[Jac88] Jacobson, V., "Congestion avoidance and control", ACM SIGCOMM, DOI 10.1145/52325.52356, August 1988, <>.

[JAC88] Jacobson、V.、「輻輳回避・制御」、ACM SIGCOMM、DOI 10.1145 / 52325.52356、<>。

[KP87] Karn, P. and C. Partridge, "Improving Round-Trip Time Estimates in Reliable Transport Protocols", SIGCOMM 87.


[RFC1034] Mockapetris, P., "Domain names - concepts and facilities", STD 13, RFC 1034, DOI 10.17487/RFC1034, November 1987, <>.

[RFC1034] Mockapetris、P.、「ドメイン名 - コンセプトと施設」、STD 13、RFC 1034、DOI 10.17487 / RFC1034、1987年11月、<>。

[RFC1035] Mockapetris, P., "Domain names - implementation and specification", STD 13, RFC 1035, DOI 10.17487/RFC1035, November 1987, <>.

[RFC1035] Mockapetris、P.、「ドメイン名 - 実装と仕様」、STD 13、RFC 1035、DOI 10.17487 / RFC1035、1987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:///>

[RFC2140] Touch, J., "TCP Control Block Interdependence", RFC 2140, DOI 10.17487/RFC2140, April 1997, <>.

[RFC2140] Touch、J.、 "TCP Control Block InterDependence"、RFC 2140、DOI 10.17487 / RFC2140、1997年4月、<>。

[RFC2883] Floyd, S., Mahdavi, J., Mathis, M., and M. Podolsky, "An Extension to the Selective Acknowledgement (SACK) Option for TCP", RFC 2883, DOI 10.17487/RFC2883, July 2000, <>.

[RFC2883] Floyd、S.、Mahdavi、J.、Mathis、M.、およびM. Podolsky、「TCPのための延長」、RFC 2883、DOI 10.17487 / RFC2883、2000年7月、<>。

[RFC3124] Balakrishnan, H. and S. Seshan, "The Congestion Manager", RFC 3124, DOI 10.17487/RFC3124, June 2001, <>.

[RFC3124] Balakrishnan、H.およびS.Seshan、「渋滞マネージャー」、RFC 3124、DOI 10.17487 / RFC3124、2001年6月、<>。

[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, June 2002, <>.

[RFC3261] Rosenberg、J.、Schulzrinne、H.、Camarillo、G.、Johnston、A.、Peterson、J.、Sparks、R.、Handley、M.、E. Schooler、「SIP:セッション開始プロトコル」、RFC 3261、DOI 10.17487 / RFC3261、2002年6月、<>。

[RFC3522] Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm for TCP", RFC 3522, DOI 10.17487/RFC3522, April 2003, <>.

[RFC3522] Ludwig、R.およびM.Meyer、「TCPのためのeipel検出アルゴリズム」、RFC 3522、DOI 10.17487 / RFC3522、2003年4月、<>。

[RFC3708] Blanton, E. and M. Allman, "Using TCP Duplicate Selective Acknowledgement (DSACKs) and Stream Control Transmission Protocol (SCTP) Duplicate Transmission Sequence Numbers (TSNs) to Detect Spurious Retransmissions", RFC 3708, DOI 10.17487/RFC3708, February 2004, <>.

[RFC3708] Blanton、E.およびM. Allman、「TCP複製選択確認応答(DSACKS)およびストリーム制御伝送プロトコル(SCTP)の重複伝送シーケンス番号(TSNS)の再送信シーケンス番号(TSNS)、RFC 3708、DOI 10.17487 / RFC3708、2004年2月、<>。

[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007, <>.

[RFC4960] Stewart、R.、Ed。、「ストリーム制御伝送プロトコル」、RFC 4960、DOI 10.17487 / RFC4960、2007年9月、<>。

[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、<>。

[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, "NACK-Oriented Reliable Multicast (NORM) Transport Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009, <>.

[RFC5740] Adamson、B.、Bormann、C.、Handley、M.、およびJ. Macker、「NACK-志向の信頼できるマルチキャスト(NORT)トランスポートプロトコル」、RFC 5740、DOI 10.17487 / RFC5740、2009年11月、<>。

[RFC6182] Ford, A., Raiciu, C., Handley, M., Barre, S., and J. Iyengar, "Architectural Guidelines for Multipath TCP Development", RFC 6182, DOI 10.17487/RFC6182, March 2011, <>.

[RFC6182]フォード、A.、RaiCy、C.、Handley、M.、Barre、S.、およびJ.Iyengar、「マルチパスTCP開発のための建築ガイドライン」、RFC 6182、DOI 10.17487 / RFC6182、2011年3月、<HTTPS//>。

[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:///>。

[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、<>。

[RFC7323] Borman, D., Braden, B., Jacobson, V., and R. Scheffenegger, Ed., "TCP Extensions for High Performance", RFC 7323, DOI 10.17487/RFC7323, September 2014, <>.

[RFC7323] Borman、D.、Braden、B.、Jacobson、V.、およびR.Scheffenegger、ED。、「高性能のためのTCP拡張」、RFC 7323、DOI 10.17487 / RFC7323、2014年9月、<>。



This document benefits from years of discussions with Ethan Blanton, Sally Floyd, Jana Iyengar, Shawn Ostermann, Vern Paxson, and the members of the TCPM and TCPIMPL Working Groups. Ran Atkinson, Yuchung Cheng, David Black, Stewart Bryant, Martin Duke, Wesley Eddy, Gorry Fairhurst, Rahul Arvind Jadhav, Benjamin Kaduk, Mirja Kühlewind, Nicolas Kuhn, Jonathan Looney, and Michael Scharf provided useful comments on previous draft versions of this document.

この文書は、Ethan Blanton、Sally Floyd、Jana Iyengar、Shawn Ostermann、Vern Paxson、およびTCPMおよびTCPIMPLワーキンググループのメンバーとの議論の年々の恩恵を受けています。Ran Atkinson、Yuchung Cheng、David Black、Stewart Bryant、Martin Duke、Wesley Eddy、Gorry FairHurst、Rahul Arvind Jadhav、Benjamin Kuhlewind、Nicolas Kuhn、Jonathan Looney、Michael Scharfはこの文書の以前のドラフト。

Author's Address


Mark Allman International Computer Science Institute 2150 Shattuck Ave., Suite 1100 Berkeley, CA 94704 United States of America

Mark Allman International Computer Science Institute 2150 Shattuck Ave.、Suite 1100 Berkeley、CA 94704アメリカ合衆国