Internet Engineering Task Force (IETF)                        C. Perkins
Request for Comments: 8083                         University of Glasgow
Updates: 3550                                                   V. Singh
Category: Standards Track                         
ISSN: 2070-1721                                               March 2017
Internet Engineering Task Force (IETF)                        C. Perkins
Request for Comments: 8083                         University of Glasgow
Updates: 3550                                                   V. Singh
Category: Standards Track                         
ISSN: 2070-1721                                               March 2017

Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions




The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.


This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.


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.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。有关互联网标准的更多信息,请参见RFC 7841第2节。

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) 2017 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2017 IETF信托基金和确定为文件作者的人员。版权所有。

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.

本文件受BCP 78和IETF信托有关IETF文件的法律规定的约束(自本文件出版之日起生效。请仔细阅读这些文件,因为它们描述了您对本文件的权利和限制。从本文件中提取的代码组件必须包括信托法律条款第4.e节中所述的简化BSD许可证文本,并提供简化BSD许可证中所述的无担保。

Table of Contents


   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   4.  RTP Circuit Breakers for Systems Using the RTP/AVP Profile  .   8
     4.1.  RTP/AVP Circuit Breaker #1: RTCP Timeout  . . . . . . . .  10
     4.2.  RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . .  11
     4.3.  RTP/AVP Circuit Breaker #3: Congestion  . . . . . . . . .  12
     4.4.  RTP/AVP Circuit Breaker #4: Media Usability . . . . . . .  16
     4.5.  Ceasing Transmission  . . . . . . . . . . . . . . . . . .  17
   5.  RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles   18
   6.  Impact of RTCP Extended Reports (XR)  . . . . . . . . . . . .  19
   7.  Impact of Explicit Congestion Notification (ECN)  . . . . . .  19
   8.  Impact of Bundled Media and Layered Coding  . . . . . . . . .  20
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  20
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  21
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  21
     10.2.  Informative References . . . . . . . . . . . . . . . . .  22
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  25
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  25
   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   4.  RTP Circuit Breakers for Systems Using the RTP/AVP Profile  .   8
     4.1.  RTP/AVP Circuit Breaker #1: RTCP Timeout  . . . . . . . .  10
     4.2.  RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . .  11
     4.3.  RTP/AVP Circuit Breaker #3: Congestion  . . . . . . . . .  12
     4.4.  RTP/AVP Circuit Breaker #4: Media Usability . . . . . . .  16
     4.5.  Ceasing Transmission  . . . . . . . . . . . . . . . . . .  17
   5.  RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles   18
   6.  Impact of RTCP Extended Reports (XR)  . . . . . . . . . . . .  19
   7.  Impact of Explicit Congestion Notification (ECN)  . . . . . .  19
   8.  Impact of Bundled Media and Layered Coding  . . . . . . . . .  20
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  20
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  21
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  21
     10.2.  Informative References . . . . . . . . . . . . . . . . .  22
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  25
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  25
1. Introduction
1. 介绍

The Real-time Transport Protocol (RTP) [RFC3550] is widely used in voice-over-IP, video teleconferencing, and telepresence systems. Many of these systems run over best-effort UDP/IP networks and can suffer from packet loss and increased latency if network congestion occurs. Designing effective RTP congestion control algorithms to adapt the transmission of RTP-based media to match the available network capacity while also maintaining the user experience is a difficult but important problem. Many such congestion control and media adaptation algorithms have been proposed, but to date there is no consensus on the correct approach or even that a single standard algorithm is desirable.


This memo does not attempt to propose a new RTP congestion control algorithm. Instead, we propose a small set of RTP circuit breakers: mechanisms that terminate RTP flows in conditions under which there is general agreement that serious network congestion is occurring. The RTP circuit breakers proposed in this memo are a specific instance of the general class of network transport circuit breakers [RFC8084] designed to act as a protection mechanism of last resort to avoid persistent excessive congestion. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by the RTP circuit breaker algorithms defined by this memo.


2. Background
2. 出身背景

We consider congestion control for unicast RTP traffic flows. This is the problem of adapting the transmission of an audio/visual data flow, encapsulated within an RTP transport session, from one sender to one receiver so that it does not use more capacity than is available along the network path. Such adaptation needs to be done in a way that limits the disruption to the user experience caused by both packet loss and excessive rate changes. Congestion control for multicast flows is outside the scope of this memo. Multicast traffic needs different solutions since the available capacity estimator for a group of receivers will differ from that for a single receiver, and because multicast congestion control has to consider issues of fairness across groups of receivers that do not apply to unicast flows.


Congestion control for unicast RTP traffic can be implemented in one of two places in the protocol stack. One approach is to run the RTP traffic over a congestion-controlled transport protocol (for example, over TCP), and to adapt the media encoding to match the dictates of the transport-layer congestion control algorithm. This is safe for the network but can be suboptimal for the media quality unless the


transport protocol is designed to support real-time media flows. We do not consider this class of applications further in this memo, as their network safety is guaranteed by the underlying transport.


Alternatively, RTP flows can be run over a non-congestion-controlled transport protocol (for example, UDP) performing rate adaptation at the application layer based on RTP Control Protocol (RTCP) feedback. With a well-designed, network-aware application, this allows highly effective media quality adaptation, but there is potential to cause persistent congestion in the network if the application does not adapt its sending rate in a timely and effective manner. We consider this class of applications in this memo.


Congestion control relies on monitoring the delivery of a media flow and responding to adapt the transmission of that flow when there are signs that the network path is congested. Network congestion can be detected in one of three ways:


1) a receiver can infer the onset of congestion by observing an increase in one-way delay caused by queue build-up within the network;

1) 接收器可以通过观察由网络内的队列建立引起的单向延迟的增加来推断拥塞的开始;

2) if Explicit Congestion Notification (ECN) [RFC3168] is supported, the network can signal the presence of congestion by marking packets using ECN Congestion Experienced (CE) marks (this could potentially be augmented by mechanisms such as Congestion Exposure (ConEx) [RFC7713] or other future protocol extensions for network signaling of congestion); or

2) 如果支持显式拥塞通知(ECN)[RFC3168],则网络可以通过使用ECN拥塞经历(CE)标记标记分组来发出拥塞存在的信号(这可能通过诸如拥塞暴露(ConEx)[RFC7713]或用于网络拥塞信令的其他未来协议扩展等机制来增强); 或

3) in the extreme case, congestion will cause packet loss that can be detected by observing a gap in the received RTP sequence numbers.

3) 在极端情况下,拥塞将导致数据包丢失,可以通过观察接收到的RTP序列号中的间隙来检测数据包丢失。

Once the onset of congestion is observed, the receiver has to send feedback to the sender to indicate that the transmission rate needs to be reduced. How the sender reduces the transmission rate is highly dependent on the media codec being used and is outside the scope of this memo.


There are several ways in which a receiver can send feedback to a media sender within the RTP framework:


o The base RTP specification [RFC3550] defines RTCP Receiver Report (RR) packets to convey reception quality feedback information and Sender Report (SR) packets to convey information about the media transmission. RTCP SR packets contain data that can be used to reconstruct media timing at a receiver along with a count of the total number of octets and packets sent. RTCP RR packets report

o 基本RTP规范[RFC3550]定义了RTCP接收器报告(RR)数据包以传递接收质量反馈信息,以及发送器报告(SR)数据包以传递有关媒体传输的信息。RTCP SR数据包包含可用于在接收器处重建媒体定时的数据,以及发送的八位组和数据包总数的计数。RTCP RR数据包报告

on the fraction of packets lost in the last reporting interval, the cumulative number of packets lost, the highest sequence number received, and the inter-arrival jitter. The RTCP RR packets also contain timing information that allows the sender to estimate the network Round-Trip Time (RTT) to the receivers. RTCP reports are sent periodically, with the reporting interval being determined by the number of Synchronization Sources (SSRCs) used in the session and a configured session bandwidth estimate (the number of SSRCs) used is usually two in a unicast session, one for each participant, but can be greater if the participants send multiple media streams). The interval between reports sent from each receiver is on the order of a few seconds on average; although it varies with the session bandwidth, it is randomized to avoid synchronization of reports from multiple receivers. The interval can be less than a second in a high-bandwidth session. RTCP RR packets allow a receiver to report ongoing network congestion to the sender. However, if a receiver detects the onset of congestion part way through a reporting interval, the base RTP specification contains no provision for sending the RTCP RR packet early, and the receiver has to wait until the next scheduled reporting interval.

在上一个报告间隔内丢失的数据包分数、累计丢失的数据包数、接收到的最高序列号和到达间抖动。RTCP RR数据包还包含允许发送方估计到接收方的网络往返时间(RTT)的定时信息。RTCP报告定期发送,报告间隔由会话中使用的同步源(SSRC)数量确定,配置的会话带宽估计(SSRC数量)在单播会话中通常为两个,每个参与者一个,但如果参与者发送多个媒体流,则可能更大)。每个接收器发送报告的间隔平均为几秒钟;虽然它随会话带宽而变化,但它是随机的,以避免来自多个接收器的报告同步。在高带宽会话中,间隔可以小于1秒。RTCP RR数据包允许接收方向发送方报告正在发生的网络拥塞。但是,如果接收器在报告间隔的一段时间内检测到拥塞的开始,则基本RTP规范不包含提前发送RTCP RR数据包的规定,并且接收器必须等到下一个计划的报告间隔。

o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more complex and sophisticated reception quality metrics but do not change the RTCP timing rules. RTCP extended reports of potential interest for congestion control purposes are the extended packet loss, discard, and burst metrics [RFC3611] [RFC7002] [RFC7097] [RFC7003] [RFC6958] as well as the extended delay metrics [RFC6843] [RFC6798]. Other RTCP Extended Reports that could be helpful for congestion control purposes might be developed in future.

o RTCP扩展报告(XR)[RFC3611]允许报告更复杂和复杂的接收质量指标,但不改变RTCP定时规则。用于拥塞控制目的的RTCP扩展报告包括扩展数据包丢失、丢弃和突发度量[RFC3611][RFC7002][RFC7097][RFC7003][RFC6958]以及扩展延迟度量[RFC6843][RFC6798]。将来可能会开发其他RTCP扩展报告,这些报告可能有助于拥塞控制。

o Rapid feedback about the occurrence of congestion events can be achieved using the Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124]) in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP timing rules to allow RTCP reports to be sent early, in some cases immediately, provided the RTCP transmission rate keeps within its bandwidth allocation. It also defines transport-layer feedback messages, including Negative Acknowledgements (NACKs), that can be used to report on specific congestion events. RTP Codec Control Messages [RFC5104] extend the RTP/AVPF profile with additional feedback messages that can be used to influence the way in which rate adaptation occurs but do not further change the dynamics of how rapidly feedback can be sent. Use of the RTP/AVPF profile is dependent on signaling.

o 使用基于RTCP的反馈(RTP/AVPF)[RFC4585](或其安全变体RTP/SAVPF[RFC5124])的扩展RTP配置文件代替RTP/AVP配置文件[RFC3551],可以实现拥塞事件发生的快速反馈。这修改了RTCP定时规则,以允许提前发送RTCP报告,在某些情况下,如果RTCP传输速率保持在其带宽分配范围内,则可以立即发送。它还定义了传输层反馈消息,包括可用于报告特定拥塞事件的否定确认(NACK)。RTP编解码器控制消息[RFC5104]使用附加反馈消息扩展RTP/AVPF配置文件,这些消息可用于影响速率自适应发生的方式,但不会进一步改变反馈发送速度的动态。RTP/AVPF配置文件的使用取决于信令。

o Finally, ECN for RTP over UDP [RFC6679] can be used to provide feedback on the number of packets that received an ECN-CE mark. This RTCP extension builds on the RTP/AVPF profile to allow rapid congestion feedback when ECN is supported.

o 最后,UDP上RTP的ECN[RFC6679]可用于提供有关接收ECN-CE标记的数据包数量的反馈。此RTCP扩展建立在RTP/AVPF配置文件的基础上,以允许在支持ECN时进行快速拥塞反馈。

In addition to these mechanisms for providing feedback, the sender can include an RTP header extension in each packet to record packet transmission times [RFC5450]. Accurate transmission timestamps can be helpful for estimating queuing delays to get an early indication of the onset of congestion.


Taken together, these various mechanisms allow receivers to provide feedback on the senders when congestion events occur, with varying degrees of timeliness and accuracy. The key distinction is between systems that use only the basic RTCP mechanisms, without RTP/AVPF rapid feedback, and those that use the RTP/AVPF extensions to respond to congestion more rapidly.


3. Terminology
3. 术语

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119]. This interpretation of these key words applies only when written in ALL CAPS. Mixed- or lower-case uses of these key words are not to be interpreted as carrying special significance in this memo.

本文件中的关键词“必须”、“不得”、“要求”、“应”、“不应”、“应”、“不应”、“建议”、“可”和“可选”应按照RFC 2119[RFC2119]中所述进行解释。对这些关键词的解释仅适用于所有大写字母。在本备忘录中,这些关键词的混合或小写用法不得解释为具有特殊意义。

The definition of the RTP circuit breaker is specified in terms of the following variables:


o Td is the deterministic RTCP reporting interval, as defined in Section 6.3.1 of [RFC3550].

o Td是确定性RTCP报告间隔,如[RFC3550]第6.3.1节所定义。

o Tdr is the sender's estimate of the deterministic RTCP reporting interval, Td, calculated by a receiver of the data it is sending. Tdr is not known at the sender but can be estimated by executing the algorithm in Section 6.2 of [RFC3550] using the average RTCP packet size seen at the sender, the number of members reported in the receiver's SR/RR report blocks, and whether the receiver is sending SR or RR packets. Tdr is recalculated when each new RTCP SR/RR report is received, but the media timeout circuit breaker (see Section 4.2) is only reconsidered when Tdr increases.

o Tdr是发送方对确定性RTCP报告间隔Td的估计值,由其发送数据的接收方计算得出。发送方不知道Tdr,但可以通过执行[RFC3550]第6.2节中的算法,使用发送方看到的平均RTCP数据包大小、接收方SR/RR报告块中报告的成员数量以及接收方是否发送SR或RR数据包来估计Tdr。当收到每个新的RTCP SR/RR报告时,重新计算Tdr,但只有当Tdr增加时,才重新考虑介质超时断路器(见第4.2节)。

o Tr is the network round-trip time, which is calculated by the sender using the algorithm in Section 6.4.1 of [RFC3550] and is smoothed using an exponentially weighted moving average as Tr = (0.8 * Tr) + (0.2 * Tr_new) where Tr_new is the latest RTT estimate obtained from an RTCP report. The weight is chosen so old estimates decay over k intervals.

o Tr是网络往返时间,由发送方使用[RFC3550]第6.4.1节中的算法计算,并使用指数加权移动平均值进行平滑,Tr=(0.8*Tr)+(0.2*Tr_new),其中Tr_new是从RTCP报告中获得的最新RTT估计值。选择权重是为了使旧的估计值在k个区间内衰减。

o k is the non-reporting threshold (see Section 4.2).

o k是非报告阈值(见第4.2节)。

o Tf is the media framing interval at the sender. For applications sending at a constant frame rate, Tf is the inter-frame interval. For applications that switch between a small set of possible frame rates (for example, when sending speech with comfort noise, such that comfort noise frames are sent less often than speech frames), Tf is set to the longest of the inter-frame intervals of the different frame rates. For applications that send periodic frames but dynamically vary their frame rate, Tf is set to the largest inter-frame interval used in the last 10 seconds. For applications that send less than one frame every 10 seconds, or that have no concept of periodic frames (e.g., text conversation [RFC4103], or pointer events [RFC2862]), when each frame is sent, Tf is set to the time interval since the previous frame.

o Tf是发送方的媒体帧间隔。对于以恒定帧速率发送的应用程序,Tf是帧间间隔。对于在一小组可能的帧速率之间切换的应用(例如,当发送带有舒适噪声的语音时,使得舒适噪声帧的发送频率小于语音帧),Tf被设置为不同帧速率的帧间间隔的最长值。对于发送周期性帧但动态改变其帧速率的应用程序,Tf设置为过去10秒内使用的最大帧间间隔。对于每10秒发送少于一帧的应用程序,或没有周期帧概念的应用程序(例如,文本对话[RFC4103]或指针事件[RFC2862]),当发送每个帧时,Tf设置为自上一帧起的时间间隔。

o G is the frame group size. That is, the number of frames that are coded together based on a particular sending rate setting. If the codec used by the sender can change its rate on each frame, then G = 1; otherwise, G is set to the number of frames before the codec can adjust to the new rate. For codecs that have the concept of a Group of Pictures (GOP), G is likely the GOP length.

o G是帧组大小。即,基于特定发送速率设置编码在一起的帧数。如果发送方使用的编解码器可以更改其在每个帧上的速率,则G=1;否则,G设置为编解码器可以调整到新速率之前的帧数。对于具有一组图片(GOP)概念的编解码器,G很可能是GOP长度。

o T_rr_interval is the minimal interval between RTCP reports, as defined in Section 3.4 of [RFC4585]; it is only meaningful for implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124].

o T_rr_interval是RTCP报告之间的最小间隔,如[RFC4585]第3.4节所定义;它仅对RTP/AVPF配置文件[RFC4585]或RTP/SAVPF配置文件[RFC5124]的实现有意义。

o X is the estimated throughput a TCP connection would achieve over a path, in bytes per second.

o X是TCP连接在路径上实现的估计吞吐量,以字节/秒为单位。

o s is the size of RTP packets being sent, in bytes. If the RTP packets being sent vary in size, then the average size over the packet comprising the last 4 * G frames MUST be used (this is intended to be comparable to the four loss intervals used in [RFC5348]).

o s是正在发送的RTP数据包的大小,以字节为单位。如果发送的RTP数据包大小不同,则必须使用包含最后4*G帧的数据包的平均大小(这与[RFC5348]中使用的四个丢失间隔相当)。

o p is the loss event rate, between 0.0 and 1.0, that would be seen by a TCP connection over a particular path. When used in the RTP congestion circuit breaker, this is approximated as described in Section 4.3.

o p是丢失事件率,介于0.0和1.0之间,通过特定路径上的TCP连接可以看到。当用于RTP拥塞断路器时,其近似值如第4.3节所述。

o t_RTO is the retransmission timeout value that would be used by a TCP connection over a particular path, in seconds. This MUST be approximated using t_RTO = 4 * Tr when used as part of the RTP congestion circuit breaker.

o t_RTO是TCP连接在特定路径上使用的重传超时值,以秒为单位。当用作RTP拥塞断路器的一部分时,必须使用t_RTO=4*Tr近似。

o b is the number of packets that are acknowledged by a single TCP acknowledgement. Following [RFC5348], it is RECOMMENDED that the value b = 1 is used as part of the RTP congestion circuit breaker.

o b是单个TCP确认确认的数据包数。根据[RFC5348],建议将值b=1用作RTP拥塞断路器的一部分。

4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
4. 使用RTP/AVP配置文件的系统的RTP断路器

The feedback mechanisms defined in [RFC3550] and available under the RTP/AVP profile [RFC3551] are the minimum that can be assumed for a baseline circuit breaker mechanism that is suitable for all unicast applications of RTP. Accordingly, for an RTP circuit breaker to be useful, it needs to be able to detect that an RTP flow is causing excessive congestion using only basic RTCP features without needing RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.

[RFC3550]中定义并在RTP/AVP配置文件[RFC3551]下可用的反馈机制是适用于RTP的所有单播应用的基线断路器机制可以假设的最小值。因此,为了使RTP断路器有用,它需要能够仅使用基本RTCP功能而不需要RTCP XR反馈或快速RTCP报告的RTP/AVPF配置文件来检测RTP流是否导致过度拥塞。

RTCP is a fundamental part of the RTP protocol, and the mechanisms described here rely on the implementation of RTCP. Implementations that claim to support RTP, but that do not implement RTCP, will be unable to use the circuit breaker mechanisms described in this memo. Such implementations SHOULD NOT be used on networks that might be subject to congestion unless equivalent mechanisms are defined using some non-RTCP feedback channel to report congestion and signal circuit breaker conditions.


The RTCP timeout circuit breaker (Section 4.1) will trigger if an implementation of this memo attempts to interwork with an endpoint that does not support RTCP. Implementations that sometimes need to interwork with endpoints that do not support RTCP need to disable the RTP circuit breakers if they don't receive some confirmation via signaling that the remote endpoint implements RTCP (the presence of a Session Description Protocol (SDP) "a=rtcp:" attribute in an answer might be such an indication). The RTP circuit breaker SHOULD NOT be disabled on networks that might be subject to congestion unless equivalent mechanisms are defined using some non-RTCP feedback channel to report congestion and signal circuit breaker conditions [RFC8084].


Three potential congestion signals are available from the basic RTCP SR/RR packets and are reported for each SSRC in the RTP session:

基本RTCP SR/RR数据包提供三个潜在拥塞信号,并在RTP会话中为每个SSRC报告:

1. The sender can estimate the network round-trip time once per RTCP reporting interval based on the contents and timing of RTCP SR and RR packets.

1. 发送方可以根据RTCP SR和RR数据包的内容和时间,在每个RTCP报告间隔内估计一次网络往返时间。

2. Receivers report a jitter estimate (the statistical variance of the RTP data packet inter-arrival time) calculated over the RTCP reporting interval. Due to the nature of the jitter calculation (Section 6.4.4. of [RFC3550]), the jitter is only meaningful for RTP flows that send a single data packet for each RTP timestamp value (i.e., audio flows, or video flows where each packet comprises one video frame).

2. 接收机报告在RTCP报告间隔内计算的抖动估计值(RTP数据包到达时间的统计方差)。由于抖动计算的性质(RFC3550的第6.4.4节),抖动仅对为每个RTP时间戳值发送单个数据包的RTP流有意义(即,音频流或视频流,其中每个包包括一个视频帧)。

3. Receivers report the fraction of RTP data packets lost during the RTCP reporting interval and the cumulative number of RTP packets lost over the entire RTP session.

3. 接收方报告在RTCP报告间隔期间丢失的RTP数据包的分数,以及整个RTP会话期间丢失的RTP数据包的累积数量。

These congestion signals limit the possible circuit breakers since they give only limited visibility into the behavior of the network.


RTT estimates are widely used in congestion control algorithms as a proxy for queuing delay measures in delay-based congestion control or to determine connection timeouts. RTT estimates derived from RTCP SR and RR packets sent according to the RTP/AVP timing rules are too infrequent to be useful for congestion control and don't give enough information to distinguish a delay change due to routing updates from queuing delay caused by congestion. Accordingly, we cannot use the RTT estimate alone as an RTP circuit breaker.

RTT估计广泛应用于拥塞控制算法中,作为基于延迟的拥塞控制中排队延迟度量的代理,或用于确定连接超时。根据RTP/AVP定时规则发送的RTCP SR和RR数据包得出的RTT估计太少,无法用于拥塞控制,并且无法提供足够的信息来区分路由更新引起的延迟变化和拥塞引起的排队延迟。因此,我们不能将RTT估算单独用作RTP断路器。

Increased jitter can be a signal of transient network congestion, but in the highly aggregated form reported in RTCP RR packets, it offers insufficient information to estimate the extent or persistence of congestion. Jitter reports are a useful early warning of potential network congestion but provide an insufficiently strong signal to be used as a circuit breaker.

抖动增加可能是暂时性网络拥塞的信号,但在RTCP RR数据包中报告的高度聚合形式中,它提供的信息不足以估计拥塞的程度或持续性。抖动报告是潜在网络拥塞的有用预警,但提供的信号不够强,无法用作断路器。

The remaining congestion signals are the packet loss fraction and the cumulative number of packets lost. If considered carefully, and over an appropriate time frame to distinguish transient problems from long term issues [RFC8084], these can be effective indicators that persistent excessive congestion is occurring in networks where packet loss is primarily due to queue overflows, although loss caused by non-congestive packet corruption can distort the result in some networks. TCP congestion control [RFC5681] intentionally tries to fill the router queues and uses the resulting packet loss as congestion feedback. An RTP flow competing with TCP traffic will therefore expect to see a non-zero packet loss fraction, and some variation in queuing latency, in normal operation when sharing a path with other flows, which needs to be accounted for when determining the circuit breaker threshold [RFC8084]. This behavior of TCP is reflected in the congestion circuit breaker below and will affect the design of any RTP congestion control protocol.


Two packet loss regimes can be observed: 1) RTCP RR packets show a non-zero packet loss fraction while the extended highest sequence number received continues to increment; and 2) RR packets show a loss fraction of zero, but the extended highest sequence number received does not increment even though the sender has been transmitting RTP data packets. The former corresponds to the TCP congestion avoidance state and indicates a congested path that is still delivering data; the latter corresponds to a TCP timeout and is most likely due to a path failure. A third condition is that data is being sent but no RTCP feedback is received at all, corresponding to a failure of the reverse path. We derive circuit breaker conditions for these loss regimes in the following.

可以观察到两种丢包情况:1)RTCP RR数据包显示非零丢包分数,而接收到的扩展最高序列号继续增加;和2)RR分组显示零的丢失分数,但是即使发送方已经发送RTP数据分组,所接收的扩展的最高序列号也不会增加。前者对应于TCP拥塞避免状态并指示仍在传送数据的拥塞路径;后者对应于TCP超时,很可能是由于路径故障。第三个条件是发送数据,但根本没有接收到RTCP反馈,这与反向路径故障相对应。我们在下面推导了这些损耗状态的断路器条件。

4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout
4.1. RTP/AVP断路器#1:RTCP超时

An RTCP timeout can occur when RTP data packets are being sent, but there are no RTCP reports returned from the receiver. This is either due to a failure of the receiver to send RTCP reports or a failure of the return path that is preventing those RTCP reporting from being delivered. In either case, it is not safe to continue transmission since the sender has no way of knowing if it is causing congestion.


An RTP sender that has not received any RTCP SR or RTCP RR packets reporting on the SSRC it is using, for a time period of at least three times its deterministic RTCP reporting interval, Td (where Td is calculated without the randomization factor and using the fixed minimum interval of Tmin=5 seconds), SHOULD cease transmission (see Section 4.5). The rationale for this choice of timeout is as described in Section 6.2 of [RFC3550] ("so that implementations which do not use the reduced value for transmitting RTCP packets are not timed out by other participants prematurely") and has been updated by Section 6.1.4 of [RFC8108] to account for the use of the RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124].

如果RTP发送方在至少三倍于其确定RTCP报告间隔Td的时间段内未收到任何报告SSRC的RTCP SR或RTCP RR数据包(其中Td的计算不含随机化因子,且使用固定的最小间隔Tmin=5秒),则应停止传输(见第4.5节)。此超时选择的基本原理如[RFC3550]第6.2节所述(“以便不使用减少值传输RTCP数据包的实现不会被其他参与者过早超时”),并已由[RFC8108]第6.1.4节更新,以说明RTP/AVPF配置文件[RFC4585]的使用或RTP/SAVPF配置文件[RFC5124]。

To reduce the risk of premature timeout, implementations SHOULD NOT configure the RTCP bandwidth such that Td is larger than 5 seconds. Similarly, implementations that use the RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to values larger than 4 seconds (the reduced limit for T_rr_interval follows Section 6.1.3 of [RFC8108]).


The choice of three RTCP reporting intervals as the timeout is made following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that participants in an RTP session will timeout and remove an RTP sender from the list of active RTP senders if no RTP data packets have been received from that RTP sender within the last two RTCP reporting intervals. Using a timeout of three RTCP reporting intervals is therefore large enough that the other participants will have timed

按照RFC 3550[RFC3550]第6.3.5节的规定,选择三个RTCP报告间隔作为超时时间。这指定如果在最近两个RTP报告间隔内没有从RTP发送方收到RTP数据包,RTP会话的参与者将超时并从活动RTP发送方列表中删除该RTP发送方。因此,使用三个RTCP报告间隔的超时时间足够大,其他参与者将有时间

out the sender if a network problem stops the data packets it is sending from reaching the receivers, even allowing for loss of some RTCP packets.


If a sender is transmitting a large number of RTP media streams, such that the corresponding RTCP SR or RR packets are too large to fit into the network MTU, the receiver will generate RTCP SR or RR packets in a round-robin manner. In this case, the sender SHOULD treat receipt of an RTCP SR or RR packet corresponding to any SSRC it sent on the same 5-tuple of source and destination IP address, port, and protocol as an indication that the receiver and return path are working and thus preventing the RTCP timeout circuit breaker from triggering.

如果发送方正在发送大量RTP媒体流,使得相应的RTCP SR或RR数据包太大而无法装入网络MTU,则接收方将以循环方式生成RTCP SR或RR数据包。在这种情况下,发送方应将接收到的RTCP SR或RR数据包(对应于其在源和目标IP地址、端口和协议的相同5元组上发送的任何SSRC)视为接收方和返回路径正在工作的指示,从而防止触发RTCP超时断路器。

4.2. RTP/AVP Circuit Breaker #2: Media Timeout
4.2. RTP/AVP断路器#2:媒体超时

If RTP data packets are being sent but the RTCP SR or RR packets reporting on that SSRC indicate a non-increasing extended highest sequence number received, this is an indication that those RTP data packets are not reaching the receiver. This could be a short-term issue affecting only a few RTP packets, perhaps caused by a slow-to-open firewall or a transient connectivity problem, but if the issue persists, it is a sign of a more ongoing and significant problem (a "media timeout").

如果正在发送RTP数据包,但在该SSRC上报告的RTCP SR或RR数据包指示接收到的扩展最高序列号未增加,则表示这些RTP数据包未到达接收器。这可能是一个仅影响少数RTP数据包的短期问题,可能是由于防火墙打开缓慢或暂时性连接问题造成的,但如果问题持续存在,则表明存在一个更为严重的问题(“媒体超时”)。

The time needed to declare a media timeout depends on the parameters Tdr, Tr, Tf, and on the non-reporting threshold k. The value of k is chosen so that when Tdr is large compared to Tr and Tf, receipt of at least k RTCP reports with non-increasing extended highest sequence number received gives reasonable assurance that the forward path has failed and that the RTP data packets have not been lost by chance. The RECOMMENDED value for k is 5 reports.


When Tdr < Tf, then RTP data packets are being sent at a rate less than one per RTCP reporting interval of the receiver, so the extended highest sequence number received can be expected to be non-increasing for some receiver RTCP reporting intervals. Similarly, when Tdr < Tr, some receiver RTCP reporting intervals might pass before the RTP data packets arrive at the receiver, also leading to reports where the extended highest sequence number received is non-increasing. Both issues require the media timeout interval to be scaled relative to the threshold, k.


The media timeout RTP circuit breaker is therefore as follows. When starting sending, calculate MEDIA_TIMEOUT using:


      MEDIA_TIMEOUT = ceil(k * max(Tf, Tr, Tdr) / Tdr)
      MEDIA_TIMEOUT = ceil(k * max(Tf, Tr, Tdr) / Tdr)

When a sender receives an RTCP packet that indicates reception of the media it has been sending, then it cancels the media timeout circuit breaker. If it is still sending, then it MUST calculate a new value for MEDIA_TIMEOUT and set a new media timeout circuit breaker.


If a sender receives an RTCP packet indicating that its media was not received, it MUST calculate a new value for MEDIA_TIMEOUT. If the new value is larger than the previous, it replaces MEDIA_TIMEOUT with the new value, extending the media timeout circuit breaker; otherwise, it keeps the original value of MEDIA_TIMEOUT. This process is known as reconsidering the media timeout circuit breaker.


If MEDIA_TIMEOUT consecutive RTCP packets are received indicating that the media being sent was not received, and the media timeout circuit breaker has not been canceled, then the media timeout circuit breaker triggers. When the media timeout circuit breaker triggers, the sender SHOULD cease transmission (see Section 4.5).


When stopping sending an RTP stream, a sender MUST cancel the corresponding media timeout circuit breaker.


4.3. RTP/AVP Circuit Breaker #3: Congestion
4.3. RTP/AVP断路器#3:阻塞

If RTP data packets are being sent and the corresponding RTCP SR or RR packets show non-zero packet loss fraction and increasing extended highest sequence number received, then those RTP data packets are arriving at the receiver, but some degree of congestion is occurring. The RTP/AVP profile [RFC3551] states that:

如果正在发送RTP数据包,并且相应的RTCP SR或RR数据包显示非零的数据包丢失分数并增加接收到的扩展最高序列号,则这些RTP数据包将到达接收器,但会发生某种程度的拥塞。RTP/AVP配置文件[RFC3551]规定:

If best-effort service is being used, RTP receivers SHOULD monitor packet loss to ensure that the packet loss rate is within acceptable parameters. Packet loss is considered acceptable if a TCP flow across the same network path and experiencing the same network conditions would achieve an average throughput, measured on a reasonable timescale, that is not less than [the throughput] the RTP flow is achieving. This condition can be satisfied by implementing congestion control mechanisms to adapt the transmission rate (or the number of layers subscribed for a layered multicast session), or by arranging for a receiver to leave the session if the loss rate is unacceptably high.


The comparison to TCP cannot be specified exactly, but is intended as an "order-of-magnitude" comparison in timescale and throughput. The timescale on which TCP throughput is measured is the round-trip time of the connection. In essence, this requirement states that it is not acceptable to deploy an application (using RTP or


any other transport protocol) on the best-effort Internet which consumes bandwidth arbitrarily and does not compete fairly with TCP within an order of magnitude.


The phase "order of magnitude" in the above means within a factor of ten, approximately. In order to implement this, it is necessary to estimate the throughput a bulk TCP connection would achieve over the path. For a long-lived TCP Reno connection, it has been shown that the TCP throughput, X, in bytes per second, can be estimated as follows [Padhye]:

上述中的相位“数量级”是指大约在10倍以内。为了实现这一点,需要估计批量TCP连接在路径上实现的吞吐量。对于长寿命的TCP Reno连接,已证明TCP吞吐量X(以字节/秒为单位)可按以下方式估算[Padhye]:

      X = -------------------------------------------------------------
          Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p)))
      X = -------------------------------------------------------------
          Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p)))

This is the same approach to estimated TCP throughput that is used in [RFC5348]. Under conditions of low packet loss, the second term on the denominator is small, so this formula can be approximated with reasonable accuracy as follows [Mathis]:


      X = ----------------
      X = ----------------

It is RECOMMENDED that this simplified throughput equation be used since the reduction in accuracy is small, and it is much simpler to calculate than the full equation. Measurements have shown that the simplified TCP throughput equation is effective as an RTP circuit breaker for multimedia flows sent to hosts on residential networks using Asymmetric Digital Subscriber Line (ADSL) and cable modem links [Singh]. The data shows that the full TCP throughput equation tends to be more sensitive to packet loss and triggers the RTP circuit breaker earlier than the simplified equation. Implementations that desire this extra sensitivity MAY use the full TCP throughput equation in the RTP circuit breaker. Initial measurements in LTE networks have shown that the extra sensitivity is helpful in that environment, with the full TCP throughput equation giving a more balanced circuit breaker response than the simplified TCP equation [Sarker]; other networks might see similar behavior.


No matter what TCP throughput equation is chosen, two parameters need to be estimated and reported to the sender in order to calculate the throughput: the round-trip time, Tr, and the loss event rate, p (the packet size, s, is known to the sender). The round-trip time can be estimated from RTCP SR and RR packets. This is done too infrequently for accurate statistics but is the best that can be done with the standard RTCP mechanisms.

无论选择何种TCP吞吐量方程,都需要估计两个参数并向发送方报告,以便计算吞吐量:往返时间Tr和丢失事件率p(发送方已知数据包大小s)。往返时间可根据RTCP SR和RR数据包估算。这对于准确的统计数据来说太少了,但这是标准RTCP机制所能做到的最好的。

Report blocks in RTCP SR or RR packets contain the packet loss fraction, rather than the loss event rate, so p cannot be reported (TCP typically treats the loss of multiple packets within a single RTT as one loss event, but RTCP RR packets report the overall fraction of packets lost and do not report when the packet losses occurred). Using the loss fraction in place of the loss event rate can overestimate the loss. We believe that this overestimate will not be significant given that we are only interested in order of magnitude comparison (Section 3.2.1 of [Floyd] shows that the difference is small for steady-state conditions and random loss, but using the loss fraction is more conservative in the case of bursty loss).

RTCP SR或RR数据包中的报告块包含数据包丢失分数,而不是丢失事件率,因此无法报告p(TCP通常将单个RTT中多个数据包的丢失视为一个丢失事件,但RTCP RR数据包报告数据包丢失的总分数,并且在发生数据包丢失时不报告)。使用损失分数代替损失事件率可能会高估损失。我们认为,由于我们只对数量级比较感兴趣,因此这种高估不会太大(Floyd的第3.2.1节表明,对于稳态条件和随机损失,差异很小,但在突发性损失的情况下,使用损失分数更为保守)。

The congestion circuit breaker is therefore as follows. When a sender that is transmitting at least one RTP packet every max(Tdr, Tr) seconds receives an RTCP SR or RR packet that contains a report block for an SSRC it is using, the sender MUST record the value of the fraction lost field from the report block, and the time since the last report block was received, for that SSRC. If more than CB_INTERVAL (see below) report blocks have been received for that SSRC, the sender MUST calculate the average fraction lost over the last CB_INTERVAL reporting intervals and then estimate the TCP throughput that would be achieved over the path using the chosen TCP throughput equation and the measured values of the round-trip time, Tr, the loss event rate, p (approximated by the average fraction lost, as is described below), and the packet size, s. The estimate of the TCP throughput, X, is then compared with the actual sending rate of the RTP stream. If the actual sending rate of the RTP stream is more than 10 * X, then the congestion circuit breaker is triggered.

因此,阻塞断路器如下所示。当每max(Tdr,Tr)秒发送至少一个RTP数据包的发送方收到一个RTCP SR或RR数据包,该数据包包含其正在使用的SSRC的报告块时,发送方必须记录该SSRC的报告块丢失分数字段的值,以及自上次收到报告块以来的时间。如果该SSRC已收到超过CB_间隔(见下文)的报告块,则发送方必须计算上一个CB_间隔报告间隔内的平均损失分数,然后使用所选TCP吞吐量方程和往返时间Tr的测量值估计路径上可实现的TCP吞吐量,丢失事件率p(由平均丢失分数近似,如下所述)和分组大小s。然后将TCP吞吐量X的估计值与RTP流的实际发送速率进行比较。如果RTP流的实际发送速率大于10*X,则触发拥塞断路器。

The average fraction lost is calculated based on the sum (over the last CB_INTERVAL reporting intervals) of the fraction lost in each reporting interval that is then multiplied by the duration of the corresponding reporting interval and then divided by the total duration of the last CB_INTERVAL reporting intervals. The CB_INTERVAL parameter is set to:


         ceil(3*min(max(10*G*Tf, 10*Tr, 3*Tdr), max(15, 3*Td))/(3*Tdr))
         ceil(3*min(max(10*G*Tf, 10*Tr, 3*Tdr), max(15, 3*Td))/(3*Tdr))

The parameters that feed into CB_INTERVAL are chosen to give the congestion control algorithm time to react to congestion. They give at least three RTCP reports, ten round trip times, and ten groups of frames to adjust the rate to reduce the congestion to a reasonable level. It is expected that a responsive congestion control algorithm


will begin to respond with the next group of frames after it receives indication of congestion, so CB_INTERVAL ought to be a much longer interval than the congestion response.


If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used, and the T_rr_interval parameter is used to reduce the frequency of regular RTCP reports, then the value of Tdr in the above expression for the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval, Tdr).


The CB_INTERVAL parameter is calculated on joining the session, and recalculated on receipt of each RTCP packet, after checking whether the media timeout circuit breaker or the congestion circuit breaker has been triggered.


To ensure a timely response to persistent congestion, implementations SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than 5 seconds. Similarly, implementations that use the RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to values larger than 4 seconds (the reduced limit for T_rr_interval follows Section 6.1.3 of [RFC8108]).


The rationale for enforcing a minimum sending rate below which the congestion circuit breaker will not trigger is to avoid spurious circuit breaker triggers when the number of packets sent per RTCP reporting interval is small, and hence, the fraction lost samples are subject to measurement artifacts. The bound of at least one packet every max(Tdr, Tr) seconds is derived from the one packet per RTT minimum sending rate of TCP [RFC8085], which is adapted for use with RTP where the RTCP reporting interval is decoupled from the network RTT.


When the congestion circuit breaker is triggered, the sender SHOULD cease transmission (see Section 4.5). However, if the sender is able to reduce its sending rate by a factor of (approximately) ten, then it MAY first reduce its sending rate by this factor (or some larger amount) to see if that resolves the congestion. If the sending rate is reduced in this way and the congestion circuit breaker triggers again after the next CB_INTERVAL RTCP reporting intervals, the sender MUST then cease transmission. An example of such a rate reduction might be a video conferencing system that backs off to sending audio only before completely dropping the call. If such a reduction in sending rate resolves the congestion problem, the sender MAY gradually increase the rate at which it sends data after a reasonable amount of time has passed, provided it takes care not to cause the problem to recur ("reasonable" is intentionally not defined here since it depends on the application, media codec, and congestion control algorithm).

当阻塞断路器触发时,发送器应停止传输(见第4.5节)。然而,如果发送方能够将其发送速率降低(大约)十倍,那么它可以首先将其发送速率降低这个系数(或更大的量),以查看这是否解决了拥塞。如果发送速率以这种方式降低,并且拥塞断路器在下一个CB_INTERVAL RTCP报告间隔后再次触发,则发送方必须停止传输。这种降低速率的一个例子可能是一个视频会议系统,该系统在完全放弃呼叫之前才停止发送音频。如果发送速率的这种降低解决了拥塞问题,则发送方可以在经过合理的时间后逐渐提高其发送数据的速率,前提是它注意不使问题再次发生(“合理”)有意不在此处定义,因为它取决于应用程序、媒体编解码器和拥塞控制算法)。

The RTCP reporting interval of the media sender does not affect how quickly the congestion circuit breaker can trigger. The timing is based on the RTCP reporting interval of the receiver that generates the SR/RR packets from which the loss rate and RTT estimate are derived (note that RTCP requires all participants in a session to have similar reporting intervals, else the participant timeout rules in [RFC3550] will not work, so this interval is likely similar to that of the sender). If the incoming RTCP SR or RR packets are using a reduced minimum RTCP reporting interval (as specified in Section 6.2 of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP reporting interval is used when determining if the circuit breaker is triggered.

媒体发送器的RTCP报告间隔不影响拥塞断路器触发的速度。定时基于接收器的RTCP报告间隔,该接收器生成SR/RR数据包,从中导出丢失率和RTT估计值(请注意,RTCP要求会话中的所有参与者具有类似的报告间隔,否则参与者超时规则参见[RFC3550]将不起作用,因此此间隔可能与发件人的间隔相似)。如果传入RTCP SR或RR数据包使用缩短的最小RTCP报告间隔(如RFC 3550[RFC3550]第6.2节或RTP/AVPF配置文件[RFC4585]中的规定),则在确定断路器是否触发时,使用缩短的RTCP报告间隔。

If there are more media streams that can be reported in a single RTCP SR or RR packet, or if the size of a complete RTCP SR or RR packet exceeds the network MTU, then the receiver will report on a subset of sources in each reporting interval with the subsets selected round-robin across multiple intervals so that all sources are eventually reported [RFC3550]. When generating such round-robin RTCP reports, priority SHOULD be given to reports on sources that have high packet loss rates to ensure that senders are aware of network congestion they are causing (this is an update to [RFC3550]).

如果单个RTCP SR或RR数据包中可以报告更多的媒体流,或者如果完整RTCP SR或RR数据包的大小超过网络MTU,然后,接收器将在每个报告间隔内报告源子集,并在多个间隔内循环选择子集,以便最终报告所有源[RFC3550]。生成此类循环RTCP报告时,应优先考虑具有高丢包率的源上的报告,以确保发送方意识到它们造成的网络拥塞(这是对[RFC3550]的更新)。

4.4. RTP/AVP Circuit Breaker #4: Media Usability
4.4. RTP/AVP断路器#4:媒体可用性

Applications that use RTP are generally tolerant to some amount of packet loss. How much packet loss can be tolerated will depend on the application, media codec, and the amount of error correction and packet loss concealment that is applied. There is an upper bound on the amount of loss that can be corrected, however, beyond which the media becomes unusable. Similarly, many applications have some upper bound on the media capture to play-out latency that can be tolerated before the application becomes unusable. The latency bound will depend on the application, but typical values can range from the order of a few hundred milliseconds for voice telephony and interactive conferencing applications up to several seconds for some video-on-demand systems.


As a final circuit breaker, RTP senders SHOULD monitor the reported packet loss and delay to estimate whether the media is likely to be suitable for the intended purpose. If the packet loss rate and/or latency is such that the media has become unusable and has remained unusable for a significant time period, then the application SHOULD cease transmission. Similarly, receivers SHOULD monitor the quality of the media they receive, and if the quality is unusable for a significant time period, they SHOULD terminate the session. This memo intentionally does not define a bound on the packet loss rate or latency that will result in unusable media, as these are highly


application dependent. Similarly, the time period that is considered significant is application dependent but is likely on the order of seconds, or tens of seconds.


Sending media that suffers from such high packet loss or latency that it is unusable at the receiver is both wasteful of resources and is of no benefit to the user of the application. It also is highly likely to be congesting the network and disrupting other applications. As such, the congestion circuit breaker will almost certainly trigger to stop flows where the media would be unusable due to high packet loss or latency. However, in pathological scenarios where the congestion circuit breaker does not stop the flow, it is desirable to prevent the application sending unnecessary traffic that might disrupt other uses of the network. The role of the media usability circuit breaker is to protect the network in such cases.


4.5. Ceasing Transmission
4.5. 停止传播

What it means to cease transmission depends on the application. This could mean stopping a single RTP flow or it could mean that multiple bundled RTP flows are stopped. The intention is that the application will stop sending RTP data packets on a particular 5-tuple (transport protocol, source and destination ports, source and destination IP addresses) until whatever network problem that triggered the RTP circuit breaker has dissipated. RTP flows halted by the circuit breaker SHOULD NOT be restarted automatically unless the sender has received information that the congestion has dissipated or can reasonably be expected to have dissipated. What could trigger this expectation is necessarily application dependent, but could be, for example, an indication that a competing flow has finished and freed up some capacity, or for an application running on a mobile device it could indicate that the device moved to a new location so the flow would traverse a different path if it were restarted. Ideally, a human user will be involved in the decision to try to restart the flow since that user will eventually give up if the flows repeatedly trigger the circuit breaker. This will help avoid problems with automatic redial systems from congesting the network.


It is recognized that the RTP implementation in some systems might not be able to determine if a flow setup request was initiated by a human user or automatically by some scripted higher-level component of the system. These implementations MUST rate limit attempts to restart a flow on the same 5-tuple as used by a flow that triggered the circuit breaker so that the reaction to a triggered circuit breaker lasts for at least the triggering interval [RFC8084].


The RTP circuit breaker will only trigger, and cease transmission, for media flows subject to long-term persistent congestion. Such flows are likely to have poor quality and usability for some time before the circuit breaker triggers. Implementations can monitor RTCP Receiver Report blocks being returned for their media flows and might find it beneficial to use this information to provide a user interface cue that problems are occurring in advance of the circuit breaker triggering.


5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles

Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) [RFC4585] allows receivers to send early RTCP reports, in some cases, to inform the sender about particular events in the media stream. There are several use cases for such early RTCP reports, including providing rapid feedback to a sender about the onset of congestion. The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF profile that is treated the same in the context of the RTP circuit breaker. These feedback profiles are often used with non-compound RTCP reports [RFC5506] to reduce the reporting overhead.


Receiving rapid feedback about congestion events potentially allows congestion control algorithms to be more responsive and to better adapt the media transmission to the limitations of the network. It is expected that many RTP congestion control algorithms will adopt the RTP/AVPF profile or the RTP/SAVPF profile for this reason and thus define new transport-layer feedback reports that suit their requirements. Since these reports are not yet defined, and likely very specific to the details of the congestion control algorithm chosen, they cannot be used as part of the generic RTP circuit breaker.


Reduced-size RTCP reports sent under the RTP/AVPF early feedback rules that do not contain an RTCP SR or RR packet MUST be ignored by the congestion circuit breaker (they do not contain the information needed by the congestion circuit breaker algorithm) but MUST be counted as received packets for the RTCP timeout circuit breaker. Reduced-size RTCP reports sent under the RTP/AVPF early feedback rules that contain RTCP SR or RR packets MUST be processed by the congestion circuit breaker as if they were sent as regular RTCP reports and counted towards the circuit breaker conditions specified in Section 4 of this memo. This will potentially make the RTP circuit breaker trigger earlier than it would if the RTP/AVPF profile was not used.

拥塞断路器必须忽略根据RTP/AVPF早期反馈规则发送的不包含RTCP SR或RR数据包的缩减RTCP报告(它们不包含拥塞断路器算法所需的信息),但必须计为RTCP超时断路器的接收数据包。根据RTP/AVPF早期反馈规则发送的包含RTCP SR或RR数据包的缩减RTCP报告必须由拥塞断路器处理,如同它们作为常规RTCP报告发送一样,并计入本备忘录第4节规定的断路器条件。这可能会使RTP断路器触发时间早于未使用RTP/AVPF配置文件时的触发时间。

When using ECN with RTP (see Section 7), early RTCP feedback packets can contain ECN feedback reports. The count of ECN-CE-marked packets contained in those ECN feedback reports is counted towards the number

将ECN与RTP一起使用时(参见第7节),早期RTCP反馈数据包可以包含ECN反馈报告。这些ECN反馈报告中包含的ECN CE标记数据包的计数计入该数量

of lost packets reported if the ECN Feedback Report is sent in a compound RTCP packet along with an RTCP SR/RR report packet. Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback packets without an RTCP SR/RR packet MUST be ignored.

如果ECN反馈报告与RTCP SR/RR报告数据包一起以复合RTCP数据包的形式发送,则报告的数据包丢失量。必须忽略不带RTCP SR/RR数据包的ECN-CE数据包作为缩减的RTCP ECN反馈数据包发送的报告。

These rules are intended to allow the use of low-overhead RTP/AVPF feedback for generic NACK messages without triggering the RTP circuit breaker. This is expected to make such feedback suitable for RTP congestion control algorithms that need to quickly report loss events in between regular RTCP reports. The reaction to reduced-size RTCP SR/RR packets is to allow such algorithms to send feedback that can trigger the circuit breaker when desired.

这些规则旨在允许在不触发RTP断路器的情况下,对一般NACK消息使用低开销RTP/AVPF反馈。这预计将使此类反馈适用于RTP拥塞控制算法,这些算法需要在定期RTCP报告之间快速报告丢失事件。对减小的RTCP SR/RR数据包的反应是允许此类算法发送反馈,以便在需要时触发断路器。

The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval parameter that can be used to adjust the regular RTCP reporting interval. The use of the T_rr_interval parameter changes the behavior of the RTP circuit breaker, as described in Section 4.


6. Impact of RTCP Extended Reports (XR)
6. RTCP扩展报告(XR)的影响

RTCP Extended Report (XR) blocks provide additional reception quality metrics, but do not change the RTCP timing rules. Some of the RTCP XR blocks provide information that might be useful for congestion control purposes, others provide non-congestion-related metrics. With the exception of RTCP XR ECN Summary Reports (see Section 7), the presence of RTCP XR blocks in a compound RTCP packet does not affect the RTP circuit breaker algorithm. For consistency and ease of implementation, only the receiver report blocks contained in RTCP SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets are used by the RTP circuit breaker algorithm.

RTCP扩展报告(XR)块提供额外的接收质量指标,但不更改RTCP定时规则。一些RTCP XR块提供可能对拥塞控制有用的信息,其他块提供非拥塞相关度量。除RTCP XR ECN摘要报告(见第7节)外,复合RTCP数据包中存在RTCP XR块不会影响RTP断路器算法。为了一致性和易于实施,RTP断路器算法仅使用RTCP SR数据包、RTCP RR数据包或RTCP XR ECN摘要报告数据包中包含的接收器报告块。

7. Impact of Explicit Congestion Notification (ECN)
7. 显式拥塞通知(ECN)的影响

The use of ECN for RTP flows does not affect the RTCP timeout circuit breaker (Section 4.1) or the media timeout circuit breaker (Section 4.2) since these are both connectivity checks that simply determinate if any packets are being received.


At the time of this writing, there's no consensus on how the receipt of ECN feedback will impact the congestion circuit breaker (Section 4.3) or indeed whether the congestion circuit breaker ought to take ECN feedback into account. A future replacement of this memo is expected to provide guidance for implementers.


For the media usability circuit breaker (Section 4.4), ECN-CE-marked packets arrive at the receiver, and if they arrive in time, they will be decoded and rendered as normal. Accordingly, receipt of such packets ought not affect the usability of the media, and the arrival

对于媒体可用性断路器(第4.4节),ECN CE标记的数据包到达接收器,如果它们及时到达,它们将被解码并正常呈现。因此,接收此类数据包不应影响媒体的可用性和到达

of RTCP feedback indicating their receipt is not expected to impact the operation of the media usability circuit breaker.


8. Impact of Bundled Media and Layered Coding
8. 捆绑媒体和分层编码的影响

The RTP circuit breaker operates on a per-RTP session basis. An RTP sender that participates in several RTP sessions MUST treat each RTP session independently with regards to the RTP circuit breaker.


An RTP sender can generate several media streams within a single RTP session, with each stream using a different SSRC. This can happen if bundled media are in use when using simulcast or when using layered media coding. By default, each SSRC will be treated independently by the RTP circuit breaker. However, the sender MAY choose to treat the flows (or a subset thereof) as a group such that a circuit breaker trigger for one flow applies to the group of flows as a whole and either causes the entire group to cease transmission or causes the sending rate of the group to reduce by a factor of ten, depending on the RTP circuit breaker triggered. Grouping flows in this way is expected to be especially useful for layered flows sent using multiple SSRCs as it allows the layered flow to react as a whole, thus ceasing transmission on the enhancement layers first to reduce sending rate, if necessary, rather than treating each layer independently. Care needs to be taken if the different media streams sent on a single transport-layer flow use different Differentiated Services Code Point (DSCP) values [RFC7657] [WebRTC-QoS] since congestion could be experienced differently depending on the DSCP marking. Accordingly, RTP media streams with different DSCP values SHOULD NOT be considered as a group when evaluating the RTP circuit breaker conditions.

RTP发送方可以在单个RTP会话中生成多个媒体流,每个流使用不同的SSRC。如果使用同步广播或分层媒体编码时使用捆绑媒体,则可能发生这种情况。默认情况下,RTP断路器将独立处理每个SSRC。然而,发送方可以选择将流(或其子集)作为一个组来处理,使得一个流的断路器触发器应用于整个流组,并且根据触发的RTP断路器,使整个流组停止传输或使组的发送速率降低十倍。以这种方式分组流预计对于使用多个ssrc发送的分层流特别有用,因为它允许分层流作为一个整体反应,从而在必要时首先停止增强层上的传输以降低发送速率,而不是独立地处理每一层。如果在单个传输层流上发送的不同媒体流使用不同的区分服务代码点(DSCP)值[RFC7657][WebRTC QoS],则需要小心,因为根据DSCP标记,拥塞可能会不同。因此,在评估RTP断路器条件时,不应将具有不同DSCP值的RTP媒体流视为一组。

9. Security Considerations
9. 安全考虑

The security considerations of [RFC3550] apply.


If the RTP/AVPF profile is used to provide rapid RTCP feedback, the security considerations of [RFC4585] apply. If ECN feedback for RTP over UDP/IP is used, the security considerations of [RFC6679] apply.


If non-authenticated RTCP reports are used, an on-path attacker can trivially generate fake RTCP packets that indicate high packet loss rates and thus cause the circuit breaker to trigger and disrupt an RTP session. This is somewhat more difficult for an off-path attacker due to the need to guess the randomly chosen RTP SSRC value and the RTP sequence number. This attack can be avoided if RTCP packets are authenticated; authentication options are discussed in [RFC7201].

如果使用未经验证的RTCP报告,路径上的攻击者可能会生成表示高丢包率的伪RTCP数据包,从而导致断路器触发并中断RTP会话。由于需要猜测随机选择的RTP SSRC值和RTP序列号,这对于非路径攻击者来说有些困难。如果RTCP数据包经过身份验证,则可以避免这种攻击;[RFC7201]中讨论了身份验证选项。

Timely operation of the RTP circuit breaker depends on the choice of RTCP reporting interval. If the receiver has a reporting interval that is overly long, then the responsiveness of the circuit breaker decreases. In the limit, the RTP circuit breaker can be disabled for all practical purposes by configuring an RTCP reporting interval that has a duration of many minutes. This issue is not specific to the circuit breaker: long RTCP reporting intervals also prevent reception quality reports, feedback messages, codec control messages, etc., from being used. Implementations are expected to impose an upper limit on the RTCP reporting interval they are willing to negotiate (based on the session bandwidth and RTCP bandwidth fraction) when using the RTP circuit breaker, as discussed in Section 4.3.


10. References
10. 工具书类
10.1. Normative References
10.1. 规范性引用文件

[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.,“RFC中用于表示需求水平的关键词”,BCP 14,RFC 2119,DOI 10.17487/RFC2119,1997年3月<>.

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <>.

[RFC3550]Schulzrinne,H.,Casner,S.,Frederick,R.,和V.Jacobson,“RTP:实时应用的传输协议”,STD 64,RFC 3550,DOI 10.17487/RFC3550,2003年7月<>.

[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, <>.

[RFC3551]Schulzrinne,H.和S.Casner,“具有最小控制的音频和视频会议的RTP配置文件”,STD 65,RFC 3551,DOI 10.17487/RFC3551,2003年7月<>.

[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, DOI 10.17487/RFC3611, November 2003, <>.

[RFC3611]Friedman,T.,Ed.,Caceres,R.,Ed.,和A.Clark,Ed.,“RTP控制协议扩展报告(RTCP XR)”,RFC 3611,DOI 10.17487/RFC36112003年11月<>.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, DOI 10.17487/RFC4585, July 2006, <>.

[RFC4585]Ott,J.,Wenger,S.,Sato,N.,Burmeister,C.,和J.Rey,“基于实时传输控制协议(RTCP)的反馈(RTP/AVPF)的扩展RTP配置文件”,RFC 4585,DOI 10.17487/RFC4585,2006年7月<>.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 5348, DOI 10.17487/RFC5348, September 2008, <>.

[RFC5348]Floyd,S.,Handley,M.,Padhye,J.,和J.Widmer,“TCP友好速率控制(TFRC):协议规范”,RFC 5348,DOI 10.17487/RFC5348,2008年9月<>.

[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., and K. Carlberg, "Explicit Congestion Notification (ECN) for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August 2012, <>.

[RFC6679]Westerlund,M.,Johansson,I.,Perkins,C.,O'Hanlon,P.,和K.Carlberg,“UDP上RTP的显式拥塞通知(ECN)”,RFC 6679,DOI 10.17487/RFC66792012年8月<>.

10.2. Informative References
10.2. 资料性引用

[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "Equation-Based Congestion Control for Unicast Applications", ACM SIGCOMM Computer Communication Review, Volume 30, Issue 4, pages 43-56, DOI 10.1145/347059.347397, August 2000.

[Floyd]Floyd,S.,Handley,M.,Padhye,J.,和J.Widmer,“单播应用中基于方程的拥塞控制”,《ACM SIGCOMM计算机通信评论》,第30卷,第4期,第43-56页,DOI 10.1145/347059.347397,2000年8月。

[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The Macroscopic Behavior of the TCP Congestion Avoidance Algorithm", ACM SIGCOMM Computer Communication Review, Volume 27, Issue 3, pages 67-82, DOI 10.1145/263932.264023, July 1997.

[Mathis]Mathis,M.,Semke,J.,Mahdavi,J.,和T.Ott,“TCP拥塞避免算法的宏观行为”,ACM SIGCOMM计算机通信评论,第27卷,第3期,第67-82页,DOI 10.1145/263932.264023,1997年7月。

[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, "Modeling TCP Throughput: A Simple Model and its Empirical Validation", ACM SIGCOMM Computer Communication Review Volume 30, Issue 4, pages 303-314, DOI 10.1145/285237.285291, August 1998.

[Padhye]Padhye,J.,Firoiu,V.,Towsley,D.,和J.Kurose,“TCP吞吐量建模:一个简单模型及其实证验证”,ACM SIGCOMM计算机通信评论第30卷,第4期,第303-314页,DOI 10.1145/285237.285291,1998年8月。

[RFC2862] Civanlar, M. and G. Cash, "RTP Payload Format for Real-Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000, <>.

[RFC2862]Civanlar,M.和G.Cash,“实时指针的RTP有效载荷格式”,RFC 2862,DOI 10.17487/RFC2862,2000年6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.,和D.Black,“向IP添加显式拥塞通知(ECN)”,RFC 3168,DOI 10.17487/RFC3168,2001年9月<>.

[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, <>.

[RFC4103]Hellstrom,G.和P.Jones,“文本对话的RTP有效载荷”,RFC 4103,DOI 10.17487/RFC4103,2005年6月<>.

[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, February 2008, <>.

[RFC5104]Wenger,S.,Chandra,U.,Westerlund,M.,和B.Burman,“带反馈的RTP视听配置文件(AVPF)中的编解码器控制消息”,RFC 5104,DOI 10.17487/RFC5104,2008年2月<>.

[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <>.

[RFC5124]Ott,J.和E.Carrara,“基于实时传输控制协议(RTCP)的反馈扩展安全RTP配置文件(RTP/SAVPF)”,RFC 5124DOI 10.17487/RFC5124,2008年2月<>.

[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009, <>.

[RFC5450]Singer,D.和H.Desneni,“RTP流中的传输时间偏移”,RFC 5450,DOI 10.17487/RFC5450,2009年3月<>.

[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, DOI 10.17487/RFC5506, April 2009, <>.

[RFC5506]Johansson,I.和M.Westerlund,“支持缩小尺寸实时传输控制协议(RTCP):机会和后果”,RFC 5506,DOI 10.17487/RFC5506,2009年4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.和E.Blanton,“TCP拥塞控制”,RFC 5681,DOI 10.17487/RFC56812009年9月<>.

[RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Packet Delay Variation Metric Reporting", RFC 6798, DOI 10.17487/RFC6798, November 2012, <>.

[RFC6798]Clark,A.和Q.Wu,“用于数据包延迟变化度量报告的RTP控制协议(RTCP)扩展报告(XR)块”,RFC 6798,DOI 10.17487/RFC6798,2012年11月<>.

[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Delay Metric Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013, <>.

[RFC6843]Clark,A.,Gross,K.和Q.Wu,“用于延迟度量报告的RTP控制协议(RTCP)扩展报告(XR)块”,RFC 6843,DOI 10.17487/RFC6843,2013年1月<>.

[RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Loss Metric Reporting", RFC 6958, DOI 10.17487/RFC6958, May 2013, <>.

[RFC6958]Clark,A.,Zhang,S.,Zhao,J.,和Q.Wu,Ed.,“用于突发/间隙损失度量报告的RTP控制协议(RTCP)扩展报告(XR)块”,RFC 6958,DOI 10.17487/RFC6958,2013年5月<>.

[RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Discard Count Metric Reporting", RFC 7002, DOI 10.17487/RFC7002, September 2013, <>.

[RFC7002]Clark,A.,Zorn,G.和Q.Wu,“用于丢弃计数度量报告的RTP控制协议(RTCP)扩展报告(XR)块”,RFC 7002,DOI 10.17487/RFC7002,2013年9月<>.

[RFC7003] Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003, September 2013, <>.

[RFC7003]Clark,A.,Huang,R.,和Q.Wu,Ed.,“用于突发/间隙丢弃度量报告的RTP控制协议(RTCP)扩展报告(XR)块”,RFC 7003,DOI 10.17487/RFC7003,2013年9月<>.

[RFC7097] Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control Protocol (RTCP) Extended Report (XR) for RLE of Discarded Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014, <>.

[RFC7097]Ott,J.,Singh,V.,Ed.,和I.Curcio,“丢弃数据包RLE的RTP控制协议(RTCP)扩展报告(XR)”,RFC 7097,DOI 10.17487/RFC7097,2014年1月<>.

[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, <>.

[RFC7201]Westerlund,M.和C.Perkins,“保护RTP会话的选项”,RFC 7201,DOI 10.17487/RFC7201,2014年4月<>.

[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services (Diffserv) and Real-Time Communication", RFC 7657, DOI 10.17487/RFC7657, November 2015, <>.

[RFC7657]Black,D.,Ed.和P.Jones,“区分服务(Diffserv)和实时通信”,RFC 7657,DOI 10.17487/RFC7657,2015年11月<>.

[RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx) Concepts, Abstract Mechanism, and Requirements", RFC 7713, DOI 10.17487/RFC7713, December 2015, <>.

[RFC7713]Mathis,M.和B.Briscoe,“拥堵暴露(ConEx)概念、抽象机制和要求”,RFC 7713,DOI 10.17487/RFC7713,2015年12月<>.

[RFC8084] Fairhurst, G., "Network Transport Circuit Breakers", BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017, <>.

[RFC8084]Fairhurst,G.,“网络传输断路器”,BCP 208,RFC 8084,DOI 10.17487/RFC8084,2017年3月<>.

[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.和G.Shepherd,“UDP使用指南”,BCP 145,RFC 8085,DOI 10.17487/RFC8085,2017年3月<>.

[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, "Sending Multiple RTP Streams in a Single RTP Session", RFC 8108, DOI 10.17487/RFC8108, March 2017, <>.

[RFC8108]Lennox,J.,Westerlund,M.,Wu,Q.,和C.Perkins,“在单个RTP会话中发送多个RTP流”,RFC 8108,DOI 10.17487/RFC8108,2017年3月<>.

[Sarker] Sarker, Z., Singh, V., and C. Perkins, "An Evaluation of RTP Circuit Breaker Performance on LTE Networks", Proceedings of the IEEE INFOCOM Workshop on Communication and Networking Techniques for Contemporary Video, DOI 10.1109/INFCOMW.2014.6849240, April 2014.

[Sarker]Sarker,Z.,Singh,V.,和C.Perkins,“LTE网络上RTP断路器性能的评估”,IEEE INFOCOM当代视频通信和网络技术研讨会论文集,DOI 10.1109/INFCOMW.2014.6849240,2014年4月。

[Singh] Singh, V., McQuistin, S., Ellis, M., and C. Perkins, "Circuit Breakers for Multimedia Congestion Control", Proceedings of the 2013 20th International Packet Video Workshop (PV), DOI 10.1109/PV.2013.6691439, December 2013.

[Singh]Singh,V.,McQuistin,S.,Ellis,M.,和C.Perkins,“多媒体拥塞控制断路器”,2013年第20届国际分组视频研讨会论文集,DOI 10.1109/PV.2013.6691439,2013年12月。

[WebRTC-QoS] Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP Packet Markings for WebRTC QoS", Work in Progress, draft-ietf-tsvwg-rtcweb-qos-18, August 2016.

[WebRTC QoS]Jones,P.,Dhesikan,S.,Jennings,C.,和D.Druta,“用于WebRTC QoS的DSCP数据包标记”,正在进行的工作,草稿-ietf-tsvwg-rtcweb-QoS-18,2016年8月。



The authors would like to thank Bernard Aboba, Harald Alvestrand, Ben Campbell, Alissa Cooper, Spencer Dawkins, Gorry Fairhurst, Stephen Farrell, Nazila Fough, Kevin Gross, Cullen Jennings, Randell Jesup, Mirja Kuehlewind, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Simon Perreault, Eric Rescorla, Abheek Saha, Meral Shirazipour, Fabio Verdicchio, and Magnus Westerlund for their valuable feedback.


Authors' Addresses


Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom



Varun Singh CALLSTATS I/O Oy Runeberginkatu 4c A 4 Helsinki 00100 Finland

Varun Singh CALLSTATS I/O Oy Runeberginkatu 4c A 4赫尔辛基00100芬兰