draft-ietf-avtcore-rtp-circuit-breakers-18.txt   rfc8083.txt 
AVTCORE Working Group C. Perkins Internet Engineering Task Force (IETF) C. Perkins
Internet-Draft University of Glasgow Request for Comments: 8083 University of Glasgow
Updates: 3550 (if approved) V. Singh Updates: 3550 V. Singh
Intended status: Standards Track callstats.io Category: Standards Track callstats.io
Expires: February 19, 2017 August 18, 2016 ISSN: 2070-1721 March 2017
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-18
Abstract Abstract
The Real-time Transport Protocol (RTP) is widely used in telephony, The Real-time Transport Protocol (RTP) is widely used in telephony,
video conferencing, and telepresence applications. Such applications video conferencing, and telepresence applications. Such applications
are often run on best-effort UDP/IP networks. If congestion control are often run on best-effort UDP/IP networks. If congestion control
is not implemented in these applications, then network congestion can is not implemented in these applications, then network congestion can
lead to uncontrolled packet loss, and a resulting deterioration of lead to uncontrolled packet loss and a resulting deterioration of the
the user's multimedia experience. The congestion control algorithm user's multimedia experience. The congestion control algorithm acts
acts as a safety measure, stopping RTP flows from using excessive as a safety measure by stopping RTP flows from using excessive
resources, and protecting the network from overload. At the time of resources and protecting the network from overload. At the time of
this writing, however, while there are several proprietary solutions, this writing, however, while there are several proprietary solutions,
there is no standard algorithm for congestion control of interactive there is no standard algorithm for congestion control of interactive
RTP flows. RTP flows.
This document does not propose a congestion control algorithm. It This document does not propose a congestion control algorithm. It
instead defines a minimal set of RTP circuit breakers: conditions instead defines a minimal set of RTP circuit breakers: conditions
under which an RTP sender needs to stop transmitting media data, to under which an RTP sender needs to stop transmitting media data to
protect the network from excessive congestion. It is expected that, protect the network from excessive congestion. It is expected that,
in the absence of long-lived excessive congestion, RTP applications in the absence of long-lived excessive congestion, RTP applications
running on best-effort IP networks will be able to operate without running on best-effort IP networks will be able to operate without
triggering these circuit breakers. To avoid triggering the RTP triggering these circuit breakers. To avoid triggering the RTP
circuit breaker, any standards-track congestion control algorithms circuit breaker, any Standards Track congestion control algorithms
defined for RTP will need to operate within the envelope set by these defined for RTP will need to operate within the envelope set by these
RTP circuit breaker algorithms. RTP circuit breaker algorithms.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on February 19, 2017. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8083.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 7 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 8
4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout . . . . . . . . 10 4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout . . . . . . . . 10
4.2. RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . . 11 4.2. RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . . 11
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 12 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 12
4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 16 4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 16
4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 17 4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 17
5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 17 5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 18
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 19 6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 19
7. Impact of Explicit Congestion Notification (ECN) . . . . . . 19 7. Impact of Explicit Congestion Notification (ECN) . . . . . . 19
8. Impact of Bundled Media and Layered Coding . . . . . . . . . 19 8. Impact of Bundled Media and Layered Coding . . . . . . . . . 20
9. Security Considerations . . . . . . . . . . . . . . . . . . . 20 9. Security Considerations . . . . . . . . . . . . . . . . . . . 20
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 21
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 21 10.1. Normative References . . . . . . . . . . . . . . . . . . 21
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 21 10.2. Informative References . . . . . . . . . . . . . . . . . 22
12.1. Normative References . . . . . . . . . . . . . . . . . . 21 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 25
12.2. Informative References . . . . . . . . . . . . . . . . . 22 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 24
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used in The Real-time Transport Protocol (RTP) [RFC3550] is widely used in
voice-over-IP, video teleconferencing, and telepresence systems. voice-over-IP, video teleconferencing, and telepresence systems.
Many of these systems run over best-effort UDP/IP networks, and can Many of these systems run over best-effort UDP/IP networks and can
suffer from packet loss and increased latency if network congestion suffer from packet loss and increased latency if network congestion
occurs. Designing effective RTP congestion control algorithms, to occurs. Designing effective RTP congestion control algorithms to
adapt the transmission of RTP-based media to match the available adapt the transmission of RTP-based media to match the available
network capacity, while also maintaining the user experience, is a network capacity while also maintaining the user experience is a
difficult but important problem. Many such congestion control and difficult but important problem. Many such congestion control and
media adaptation algorithms have been proposed, but to date there is media adaptation algorithms have been proposed, but to date there is
no consensus on the correct approach, or even that a single standard no consensus on the correct approach or even that a single standard
algorithm is desirable. algorithm is desirable.
This memo does not attempt to propose a new RTP congestion control This memo does not attempt to propose a new RTP congestion control
algorithm. Instead, we propose a small set of RTP circuit breakers: algorithm. Instead, we propose a small set of RTP circuit breakers:
mechanisms that terminate RTP flows in conditions under which there mechanisms that terminate RTP flows in conditions under which there
is general agreement that serious network congestion is occurring. is general agreement that serious network congestion is occurring.
The RTP circuit breakers proposed in this memo are a specific The RTP circuit breakers proposed in this memo are a specific
instance of the general class of network transport circuit breakers instance of the general class of network transport circuit breakers
[I-D.ietf-tsvwg-circuit-breaker], designed to act as a protection [RFC8084] designed to act as a protection mechanism of last resort to
mechanism of last resort to avoid persistent excessive congestion. avoid persistent excessive congestion. To avoid triggering the RTP
To avoid triggering the RTP circuit breaker, any standards-track circuit breaker, any Standards Track congestion control algorithms
congestion control algorithms defined for RTP will need to operate defined for RTP will need to operate within the envelope set by the
within the envelope set by the RTP circuit breaker algorithms defined RTP circuit breaker algorithms defined by this memo.
by this memo.
2. Background 2. Background
We consider congestion control for unicast RTP traffic flows. This We consider congestion control for unicast RTP traffic flows. This
is the problem of adapting the transmission of an audio/visual data is the problem of adapting the transmission of an audio/visual data
flow, encapsulated within an RTP transport session, from one sender flow, encapsulated within an RTP transport session, from one sender
to one receiver, so that it does not use more capacity than is to one receiver so that it does not use more capacity than is
available along the network path. Such adaptation needs to be done available along the network path. Such adaptation needs to be done
in a way that limits the disruption to the user experience caused by in a way that limits the disruption to the user experience caused by
both packet loss and excessive rate changes. Congestion control for both packet loss and excessive rate changes. Congestion control for
multicast flows is outside the scope of this memo. Multicast traffic multicast flows is outside the scope of this memo. Multicast traffic
needs different solutions, since the available capacity estimator for needs different solutions since the available capacity estimator for
a group of receivers will differ from that for a single receiver, and a group of receivers will differ from that for a single receiver, and
because multicast congestion control has to consider issues of because multicast congestion control has to consider issues of
fairness across groups of receivers that do not apply to unicast fairness across groups of receivers that do not apply to unicast
flows. flows.
Congestion control for unicast RTP traffic can be implemented in one 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 of two places in the protocol stack. One approach is to run the RTP
traffic over a congestion controlled transport protocol, for example traffic over a congestion-controlled transport protocol (for example,
over TCP, and to adapt the media encoding to match the dictates of over TCP), and to adapt the media encoding to match the dictates of
the transport-layer congestion control algorithm. This is safe for the transport-layer congestion control algorithm. This is safe for
the network, but can be suboptimal for the media quality unless the the network but can be suboptimal for the media quality unless the
transport protocol is designed to support real-time media flows. We transport protocol is designed to support real-time media flows. We
do not consider this class of applications further in this memo, as do not consider this class of applications further in this memo, as
their network safety is guaranteed by the underlying transport. their network safety is guaranteed by the underlying transport.
Alternatively, RTP flows can be run over a non-congestion controlled Alternatively, RTP flows can be run over a non-congestion-controlled
transport protocol, for example UDP, performing rate adaptation at transport protocol (for example, UDP) performing rate adaptation at
the application layer based on RTP Control Protocol (RTCP) feedback. the application layer based on RTP Control Protocol (RTCP) feedback.
With a well-designed, network-aware, application, this allows highly With a well-designed, network-aware application, this allows highly
effective media quality adaptation, but there is potential to cause effective media quality adaptation, but there is potential to cause
persistent congestion in the network if the application does not persistent congestion in the network if the application does not
adapt its sending rate in a timely and effective manner. We consider adapt its sending rate in a timely and effective manner. We consider
this class of applications in this memo. this class of applications in this memo.
Congestion control relies on monitoring the delivery of a media flow, Congestion control relies on monitoring the delivery of a media flow
and responding to adapt the transmission of that flow when there are and responding to adapt the transmission of that flow when there are
signs that the network path is congested. Network congestion can be 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 detected in one of three ways:
congestion by observing an increase in one-way delay caused by queue
build-up within the network; 2) if Explicit Congestion Notification 1) a receiver can infer the onset of congestion by observing an
(ECN) [RFC3168] is supported, the network can signal the presence of increase in one-way delay caused by queue build-up within the
congestion by marking packets using ECN Congestion Experienced (CE) network;
marks (this could potentially be augmented by mechanisms such as
ConEX [RFC7713], or other future protocol extensions for network 2) if Explicit Congestion Notification (ECN) [RFC3168] is supported,
signalling of congestion); or 3) in the extreme case, congestion will the network can signal the presence of congestion by marking
cause packet loss that can be detected by observing a gap in the packets using ECN Congestion Experienced (CE) marks (this could
received RTP sequence numbers. potentially be augmented by mechanisms such as Congestion
Exposure (ConEx) [RFC7713] or other future protocol extensions
for network signaling of congestion); or
3) in the extreme case, congestion will cause packet loss that can
be detected by observing a gap in the received RTP sequence
numbers.
Once the onset of congestion is observed, the receiver has to send Once the onset of congestion is observed, the receiver has to send
feedback to the sender to indicate that the transmission rate needs feedback to the sender to indicate that the transmission rate needs
to be reduced. How the sender reduces the transmission rate is to be reduced. How the sender reduces the transmission rate is
highly dependent on the media codec being used, and is outside the highly dependent on the media codec being used and is outside the
scope of this memo. scope of this memo.
There are several ways in which a receiver can send feedback to a There are several ways in which a receiver can send feedback to a
media sender within the RTP framework: media sender within the RTP framework:
o The base RTP specification [RFC3550] defines RTCP Reception Report o The base RTP specification [RFC3550] defines RTCP Receiver Report
(RR) packets to convey reception quality feedback information, and (RR) packets to convey reception quality feedback information and
Sender Report (SR) packets to convey information about the media Sender Report (SR) packets to convey information about the media
transmission. RTCP SR packets contain data that can be used to transmission. RTCP SR packets contain data that can be used to
reconstruct media timing at a receiver, along with a count of the reconstruct media timing at a receiver along with a count of the
total number of octets and packets sent. RTCP RR packets report total number of octets and packets sent. RTCP RR packets report
on the fraction of packets lost in the last reporting interval, on the fraction of packets lost in the last reporting interval,
the cumulative number of packets lost, the highest sequence number the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are network Round-Trip Time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by sent periodically, with the reporting interval being determined by
the number of SSRCs used in the session and a configured session the number of Synchronization Sources (SSRCs) used in the session
bandwidth estimate (the number of synchronisation sources (SSRCs) and a configured session bandwidth estimate (the number of SSRCs)
used is usually two in a unicast session, one for each used is usually two in a unicast session, one for each
participant, but can be greater if the participants send multiple participant, but can be greater if the participants send multiple
media streams). The interval between reports sent from each media streams). The interval between reports sent from each
receiver tends to be on the order of a few seconds on average, receiver is on the order of a few seconds on average; although it
although it varies with the session bandwidth, and sub-second varies with the session bandwidth, it is randomized to avoid
reporting intervals are possible in high bandwidth sessions, and synchronization of reports from multiple receivers. The interval
it is randomised to avoid synchronisation of reports from multiple can be less than a second in a high-bandwidth session. RTCP RR
receivers. RTCP RR packets allow a receiver to report ongoing packets allow a receiver to report ongoing network congestion to
network congestion to the sender. However, if a receiver detects the sender. However, if a receiver detects the onset of
the onset of congestion part way through a reporting interval, the congestion part way through a reporting interval, the base RTP
base RTP specification contains no provision for sending the RTCP specification contains no provision for sending the RTCP RR packet
RR packet early, and the receiver has to wait until the next early, and the receiver has to wait until the next scheduled
scheduled reporting interval. reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not complex and sophisticated reception quality metrics but do not
change the RTCP timing rules. RTCP extended reports of potential change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are the extended packet interest for congestion control purposes are the extended packet
loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097], loss, discard, and burst metrics [RFC3611] [RFC7002] [RFC7097]
[RFC7003], [RFC6958]; and the extended delay metrics [RFC6843], [RFC7003] [RFC6958] as well as the extended delay metrics
[RFC6798]. Other RTCP Extended Reports that could be helpful for [RFC6843] [RFC6798]. Other RTCP Extended Reports that could be
congestion control purposes might be developed in future. helpful for congestion control purposes might be developed in
future.
o Rapid feedback about the occurrence of congestion events can be o Rapid feedback about the occurrence of congestion events can be
achieved using the Extended RTP Profile for RTCP-Based Feedback achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124]) (RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124])
in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP 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 timing rules to allow RTCP reports to be sent early, in some cases
immediately, provided the RTCP transmission rate keeps within its immediately, provided the RTCP transmission rate keeps within its
bandwidth allocation. It also defines transport-layer feedback bandwidth allocation. It also defines transport-layer feedback
messages, including negative acknowledgements (NACKs), that can be messages, including Negative Acknowledgements (NACKs), that can be
used to report on specific congestion events. RTP Codec Control used to report on specific congestion events. RTP Codec Control
Messages [RFC5104] extend the RTP/AVPF profile with additional Messages [RFC5104] extend the RTP/AVPF profile with additional
feedback messages that can be used to influence that way in which feedback messages that can be used to influence the way in which
rate adaptation occurs, but do not further change the dynamics of rate adaptation occurs but do not further change the dynamics of
how rapidly feedback can be sent. Use of the RTP/AVPF profile is how rapidly feedback can be sent. Use of the RTP/AVPF profile is
dependent on signalling. dependent on signaling.
o Finally, Explicit Congestion Notification (ECN) for RTP over UDP o Finally, ECN for RTP over UDP [RFC6679] can be used to provide
[RFC6679] can be used to provide feedback on the number of packets feedback on the number of packets that received an ECN-CE mark.
that received an ECN Congestion Experienced (CE) mark. This RTCP This RTCP extension builds on the RTP/AVPF profile to allow rapid
extension builds on the RTP/AVPF profile to allow rapid congestion congestion feedback when ECN is supported.
feedback when ECN is supported.
In addition to these mechanisms for providing feedback, the sender In addition to these mechanisms for providing feedback, the sender
can include an RTP header extension in each packet to record packet can include an RTP header extension in each packet to record packet
transmission times [RFC5450]. Accurate transmission timestamps can transmission times [RFC5450]. Accurate transmission timestamps can
be helpful for estimating queuing delays, to get an early indication be helpful for estimating queuing delays to get an early indication
of the onset of congestion. of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide Taken together, these various mechanisms allow receivers to provide
feedback on the senders when congestion events occur, with varying feedback on the senders when congestion events occur, with varying
degrees of timeliness and accuracy. The key distinction is between degrees of timeliness and accuracy. The key distinction is between
systems that use only the basic RTCP mechanisms, without RTP/AVPF systems that use only the basic RTCP mechanisms, without RTP/AVPF
rapid feedback, and those that use the RTP/AVPF extensions to respond rapid feedback, and those that use the RTP/AVPF extensions to respond
to congestion more rapidly. to congestion more rapidly.
3. Terminology 3. Terminology
skipping to change at page 6, line 29 skipping to change at page 6, line 40
interpreted as carrying special significance in this memo. interpreted as carrying special significance in this memo.
The definition of the RTP circuit breaker is specified in terms of The definition of the RTP circuit breaker is specified in terms of
the following variables: the following variables:
o Td is the deterministic RTCP reporting interval, as defined in o Td is the deterministic RTCP reporting interval, as defined in
Section 6.3.1 of [RFC3550]. Section 6.3.1 of [RFC3550].
o Tdr is the sender's estimate of the deterministic RTCP reporting o Tdr is the sender's estimate of the deterministic RTCP reporting
interval, Td, calculated by a receiver of the data it is sending. 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 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 the algorithm in Section 6.2 of [RFC3550] using the average RTCP
packet size seen at the sender, the number of members reported in packet size seen at the sender, the number of members reported in
the receiver's SR/RR report blocks, and whether the receiver is the receiver's SR/RR report blocks, and whether the receiver is
sending SR or RR packets. Tdr is recalculated when each new RTCP sending SR or RR packets. Tdr is recalculated when each new RTCP
SR/RR report is received, but the media timeout circuit breaker SR/RR report is received, but the media timeout circuit breaker
(see Section 4.2) is only reconsidered when Tdr increases. (see Section 4.2) is only reconsidered when Tdr increases.
o Tr is the network round-trip time, calculated by the sender using o Tr is the network round-trip time, which is calculated by the
the algorithm in Section 6.4.1 of [RFC3550] and smoothed using an sender using the algorithm in Section 6.4.1 of [RFC3550] and is
exponentially weighted moving average as Tr = (0.8 * Tr) + (0.2 * smoothed using an exponentially weighted moving average as
Tr_new) where Tr_new is the latest RTT estimate obtained from an Tr = (0.8 * Tr) + (0.2 * Tr_new) where Tr_new is the latest RTT
RTCP report. The weight is chosen so old estimates decay over k estimate obtained from an RTCP report. The weight is chosen so
intervals. old estimates decay over k intervals.
o k is the non-reporting threshold (see Section 4.2). o k is the non-reporting threshold (see Section 4.2).
o Tf is the media framing interval at the sender. For applications o Tf is the media framing interval at the sender. For applications
sending at a constant frame rate, Tf is the inter-frame interval. sending at a constant frame rate, Tf is the inter-frame interval.
For applications that switch between a small set of possible frame For applications that switch between a small set of possible frame
rates, for example when sending speech with comfort noise, where rates (for example, when sending speech with comfort noise, such
comfort noise frames are sent less often than speech frames, Tf is that comfort noise frames are sent less often than speech frames),
set to the longest of the inter-frame intervals of the different Tf is set to the longest of the inter-frame intervals of the
frame rates. For applications that send periodic frames but different frame rates. For applications that send periodic frames
dynamically vary their frame rate, Tf is set to the largest inter- but dynamically vary their frame rate, Tf is set to the largest
frame interval used in the last 10 seconds. For applications that inter-frame interval used in the last 10 seconds. For
send less than one frame every 10 seconds, or that have no concept applications that send less than one frame every 10 seconds, or
of periodic frames (e.g., text conversation [RFC4103], or pointer that have no concept of periodic frames (e.g., text conversation
events [RFC2862]), Tf is set to the time interval since the [RFC4103], or pointer events [RFC2862]), when each frame is sent,
previous frame when each frame is sent. Tf is set to the time interval since the previous frame.
o G is the frame group size. That is, the number of frames that are 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 coded together based on a particular sending rate setting. If the
codec used by the sender can change its rate on each frame, G = 1; codec used by the sender can change its rate on each frame, then G
otherwise G is set to the number of frames before the codec can = 1; otherwise, G is set to the number of frames before the codec
adjust to the new rate. For codecs that have the concept of a can adjust to the new rate. For codecs that have the concept of a
group-of-pictures (GoP), G is likely the GoP length. Group of Pictures (GOP), G is likely the GOP length.
o T_rr_interval is the minimal interval between RTCP reports, as o T_rr_interval is the minimal interval between RTCP reports, as
defined in Section 3.4 of [RFC4585]; it is only meaningful for defined in Section 3.4 of [RFC4585]; it is only meaningful for
implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF
profile [RFC5124]. profile [RFC5124].
o X is the estimated throughput a TCP connection would achieve over o X is the estimated throughput a TCP connection would achieve over
a path, in bytes per second. a path, in bytes per second.
o s is the size of RTP packets being sent, in bytes. If the RTP o s is the size of RTP packets being sent, in bytes. If the RTP
skipping to change at page 8, line 5 skipping to change at page 8, line 21
acknowledgement. Following [RFC5348], it is RECOMMENDED that the acknowledgement. Following [RFC5348], it is RECOMMENDED that the
value b = 1 is used as part of the RTP congestion circuit breaker. value b = 1 is used as part of the RTP congestion circuit breaker.
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
The feedback mechanisms defined in [RFC3550] and available under the The feedback mechanisms defined in [RFC3550] and available under the
RTP/AVP profile [RFC3551] are the minimum that can be assumed for a RTP/AVP profile [RFC3551] are the minimum that can be assumed for a
baseline circuit breaker mechanism that is suitable for all unicast baseline circuit breaker mechanism that is suitable for all unicast
applications of RTP. Accordingly, for an RTP circuit breaker to be 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 useful, it needs to be able to detect that an RTP flow is causing
excessive congestion using only basic RTCP features, without needing excessive congestion using only basic RTCP features without needing
RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports. RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.
RTCP is a fundamental part of the RTP protocol, and the mechanisms RTCP is a fundamental part of the RTP protocol, and the mechanisms
described here rely on the implementation of RTCP. Implementations described here rely on the implementation of RTCP. Implementations
that claim to support RTP, but that do not implement RTCP, will be that claim to support RTP, but that do not implement RTCP, will be
unable to use the circuit breaker mechanisms described in this memo. unable to use the circuit breaker mechanisms described in this memo.
Such implementations SHOULD NOT be used on networks that might be Such implementations SHOULD NOT be used on networks that might be
subject to congestion unless equivalent mechanisms are defined using subject to congestion unless equivalent mechanisms are defined using
some non-RTCP feedback channel to report congestion and signal some non-RTCP feedback channel to report congestion and signal
circuit breaker conditions. circuit breaker conditions.
The RTCP timeout circuit breaker (Section 4.1) will trigger if an The RTCP timeout circuit breaker (Section 4.1) will trigger if an
implementation of this memo attempts to interwork with an endpoint implementation of this memo attempts to interwork with an endpoint
that does not support RTCP. Implementations that sometimes need to that does not support RTCP. Implementations that sometimes need to
interwork with endpoints that do not support RTCP need to disable the interwork with endpoints that do not support RTCP need to disable the
RTP circuit breakers if they don't receive some confirmation via RTP circuit breakers if they don't receive some confirmation via
signalling that the remote endpoint implements RTCP (the presence of signaling that the remote endpoint implements RTCP (the presence of a
an SDP "a=rtcp:" attribute in an answer might be such an indication). Session Description Protocol (SDP) "a=rtcp:" attribute in an answer
The RTP Circuit Breaker SHOULD NOT be disabled on networks that might might be such an indication). The RTP circuit breaker SHOULD NOT be
be subject to congestion, unless equivalent mechanisms are defined disabled on networks that might be subject to congestion unless
using some non-RTCP feedback channel to report congestion and signal equivalent mechanisms are defined using some non-RTCP feedback
circuit breaker conditions [I-D.ietf-tsvwg-circuit-breaker]. channel to report congestion and signal circuit breaker conditions
[RFC8084].
Three potential congestion signals are available from the basic RTCP Three potential congestion signals are available from the basic RTCP
SR/RR packets and are reported for each SSRC in the RTP session: SR/RR packets and are reported for each SSRC in the RTP session:
1. The sender can estimate the network round-trip time once per RTCP 1. The sender can estimate the network round-trip time once per RTCP
reporting interval, based on the contents and timing of RTCP SR reporting interval based on the contents and timing of RTCP SR
and RR packets. and RR packets.
2. Receivers report a jitter estimate (the statistical variance of 2. Receivers report a jitter estimate (the statistical variance of
the RTP data packet inter-arrival time) calculated over the RTCP the RTP data packet inter-arrival time) calculated over the RTCP
reporting interval. Due to the nature of the jitter calculation reporting interval. Due to the nature of the jitter calculation
([RFC3550], section 6.4.4), the jitter is only meaningful for RTP (Section 6.4.4. of [RFC3550]), the jitter is only meaningful for
flows that send a single data packet for each RTP timestamp value RTP flows that send a single data packet for each RTP timestamp
(i.e., audio flows, or video flows where each packet comprises value (i.e., audio flows, or video flows where each packet
one video frame). comprises one video frame).
3. Receivers report the fraction of RTP data packets lost during the 3. Receivers report the fraction of RTP data packets lost during the
RTCP reporting interval, and the cumulative number of RTP packets RTCP reporting interval and the cumulative number of RTP packets
lost over the entire RTP session. lost over the entire RTP session.
These congestion signals limit the possible circuit breakers, since These congestion signals limit the possible circuit breakers since
they give only limited visibility into the behaviour of the network. they give only limited visibility into the behavior of the network.
RTT estimates are widely used in congestion control algorithms, as a RTT estimates are widely used in congestion control algorithms as a
proxy for queuing delay measures in delay-based congestion control or proxy for queuing delay measures in delay-based congestion control or
to determine connection timeouts. RTT estimates derived from RTCP SR to determine connection timeouts. RTT estimates derived from RTCP SR
and RR packets sent according to the RTP/AVP timing rules are too 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 infrequent to be useful for congestion control and don't give enough
information to distinguish a delay change due to routing updates from information to distinguish a delay change due to routing updates from
queuing delay caused by congestion. Accordingly, we cannot use the queuing delay caused by congestion. Accordingly, we cannot use the
RTT estimate alone as an RTP circuit breaker. RTT estimate alone as an RTP circuit breaker.
Increased jitter can be a signal of transient network congestion, but Increased jitter can be a signal of transient network congestion, but
in the highly aggregated form reported in RTCP RR packets, it offers in the highly aggregated form reported in RTCP RR packets, it offers
insufficient information to estimate the extent or persistence of insufficient information to estimate the extent or persistence of
congestion. Jitter reports are a useful early warning of potential congestion. Jitter reports are a useful early warning of potential
network congestion, but provide an insufficiently strong signal to be network congestion but provide an insufficiently strong signal to be
used as a circuit breaker. used as a circuit breaker.
The remaining congestion signals are the packet loss fraction and the The remaining congestion signals are the packet loss fraction and the
cumulative number of packets lost. If considered carefully, and over cumulative number of packets lost. If considered carefully, and over
an appropriate time frame to distinguish transient problems from long an appropriate time frame to distinguish transient problems from long
term issues [I-D.ietf-tsvwg-circuit-breaker], these can be effective term issues [RFC8084], these can be effective indicators that
indicators that persistent excessive congestion is occurring in persistent excessive congestion is occurring in networks where packet
networks where packet loss is primarily due to queue overflows, loss is primarily due to queue overflows, although loss caused by
although loss caused by non-congestive packet corruption can distort non-congestive packet corruption can distort the result in some
the result in some networks. TCP congestion control [RFC5681] networks. TCP congestion control [RFC5681] intentionally tries to
intentionally tries to fill the router queues, and uses the resulting fill the router queues and uses the resulting packet loss as
packet loss as congestion feedback. An RTP flow competing with TCP congestion feedback. An RTP flow competing with TCP traffic will
traffic will therefore expect to see a non-zero packet loss fraction, therefore expect to see a non-zero packet loss fraction, and some
and some variation in queuing latency, in normal operation when variation in queuing latency, in normal operation when sharing a path
sharing a path with other flows, that needs to be accounted for when with other flows, which needs to be accounted for when determining
determining the circuit breaker threshold the circuit breaker threshold [RFC8084]. This behavior of TCP is
[I-D.ietf-tsvwg-circuit-breaker]. This behaviour of TCP is reflected reflected in the congestion circuit breaker below and will affect the
in the congestion circuit breaker below, and will affect the design design of any RTP congestion control protocol.
of any RTP congestion control protocol.
Two packet loss regimes can be observed: 1) RTCP RR packets show a Two packet loss regimes can be observed: 1) RTCP RR packets show a
non-zero packet loss fraction, while the extended highest sequence non-zero packet loss fraction while the extended highest sequence
number received continues to increment; and 2) RR packets show a loss number received continues to increment; and 2) RR packets show a loss
fraction of zero, but the extended highest sequence number received fraction of zero, but the extended highest sequence number received
does not increment even though the sender has been transmitting RTP does not increment even though the sender has been transmitting RTP
data packets. The former corresponds to the TCP congestion avoidance data packets. The former corresponds to the TCP congestion avoidance
state, and indicates a congested path that is still delivering data; 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 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 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 RTCP feedback is received at all, corresponding to a failure of the
reverse path. We derive circuit breaker conditions for these loss reverse path. We derive circuit breaker conditions for these loss
regimes in the following. regimes in the following.
4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout 4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout
An RTCP timeout can occur when RTP data packets are being sent, but 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 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 due to a failure of the receiver to send RTCP reports or a failure of
of the return path that is preventing those RTCP reporting from being the return path that is preventing those RTCP reporting from being
delivered. In either case, it is not safe to continue transmission, delivered. In either case, it is not safe to continue transmission
since the sender has no way of knowing if it is causing congestion. 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 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 reporting on the SSRC it is using, for a time period of at least
three times its deterministic RTCP reporting interval, Td, without three times its deterministic RTCP reporting interval, Td (where Td
the randomization factor, and using the fixed minimum interval of is calculated without the randomization factor and using the fixed
Tmin=5 seconds, SHOULD cease transmission (see Section 4.5). The minimum interval of Tmin=5 seconds), SHOULD cease transmission (see
rationale for this choice of timeout is as described in Section 6.2 Section 4.5). The rationale for this choice of timeout is as
of [RFC3550] ("so that implementations which do not use the reduced described in Section 6.2 of [RFC3550] ("so that implementations which
value for transmitting RTCP packets are not timed out by other do not use the reduced value for transmitting RTCP packets are not
participants prematurely"), as updated by Section 6.1.4 of timed out by other participants prematurely") and has been updated by
[I-D.ietf-avtcore-rtp-multi-stream] to account for the use of the Section 6.1.4 of [RFC8108] to account for the use of the RTP/AVPF
RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124]. profile [RFC4585] or the RTP/SAVPF profile [RFC5124].
To reduce the risk of premature timeout, implementations SHOULD NOT To reduce the risk of premature timeout, implementations SHOULD NOT
configure the RTCP bandwidth such that Td is larger than 5 seconds. configure the RTCP bandwidth such that Td is larger than 5 seconds.
Similarly, implementations that use the RTP/AVPF profile [RFC4585] or Similarly, implementations that use the RTP/AVPF profile [RFC4585] or
the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to
values larger than 4 seconds (the reduced limit for T_rr_interval values larger than 4 seconds (the reduced limit for T_rr_interval
follows Section 6.1.3 of [I-D.ietf-avtcore-rtp-multi-stream]). follows Section 6.1.3 of [RFC8108]).
The choice of three RTCP reporting intervals as the timeout is made The choice of three RTCP reporting intervals as the timeout is made
following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that
participants in an RTP session will timeout and remove an RTP sender 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 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 received from that RTP sender within the last two RTCP reporting
intervals. Using a timeout of three RTCP reporting intervals is intervals. Using a timeout of three RTCP reporting intervals is
therefore large enough that the other participants will have timed therefore large enough that the other participants will have timed
out the sender if a network problem stops the data packets it is out the sender if a network problem stops the data packets it is
sending from reaching the receivers, even allowing for loss of some sending from reaching the receivers, even allowing for loss of some
RTCP packets. RTCP packets.
If a sender is transmitting a large number of RTP media streams, such 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 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 into the network MTU, the receiver will generate RTCP SR or RR
packets in a round-robin manner. In this case, the sender SHOULD 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 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, 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 and protocol as an indication that the receiver and return path are
working, preventing the RTCP timeout circuit breaker from triggering. working and thus preventing the RTCP timeout circuit breaker from
triggering.
4.2. RTP/AVP Circuit Breaker #2: Media Timeout 4.2. RTP/AVP Circuit Breaker #2: Media Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets If RTP data packets are being sent but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term 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 issue affecting only a few RTP packets, perhaps caused by a slow-to-
open firewall or a transient connectivity problem, but if the issue open firewall or a transient connectivity problem, but if the issue
persists, it is a sign of a more ongoing and significant problem (a persists, it is a sign of a more ongoing and significant problem (a
"media timeout"). "media timeout").
The time needed to declare a media timeout depends on the parameters 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 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 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 least k RTCP reports with non-increasing extended highest sequence
number received gives reasonable assurance that the forward path has number received gives reasonable assurance that the forward path has
failed, and that the RTP data packets have not been lost by chance. failed and that the RTP data packets have not been lost by chance.
The RECOMMENDED value for k is 5 reports. The RECOMMENDED value for k is 5 reports.
When Tdr < Tf, then RTP data packets are being sent at a rate less 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 than one per RTCP reporting interval of the receiver, so the extended
highest sequence number received can be expected to be non-increasing highest sequence number received can be expected to be non-increasing
for some receiver RTCP reporting intervals. Similarly, when Tdr < for some receiver RTCP reporting intervals. Similarly, when
Tr, some receiver RTCP reporting intervals might pass before the RTP Tdr < Tr, some receiver RTCP reporting intervals might pass before
data packets arrive at the receiver, also leading to reports where the RTP data packets arrive at the receiver, also leading to reports
the extended highest sequence number received is non-increasing. where the extended highest sequence number received is non-
Both issues require the media timeout interval to be scaled relative increasing. Both issues require the media timeout interval to be
to the threshold, k. scaled relative to the threshold, k.
The media timeout RTP circuit breaker is therefore as follows. When The media timeout RTP circuit breaker is therefore as follows. When
starting sending, calculate MEDIA_TIMEOUT using: 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 When a sender receives an RTCP packet that indicates reception of the
media it has been sending, then it cancels the media timeout circuit 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 breaker. If it is still sending, then it MUST calculate a new value
for MEDIA_TIMEOUT, and set a new media timeout circuit breaker. for MEDIA_TIMEOUT and set a new media timeout circuit breaker.
If a sender receives an RTCP packet indicating that its media was not 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 received, it MUST calculate a new value for MEDIA_TIMEOUT. If the
new value is larger than the previous, it replaces MEDIA_TIMEOUT with new value is larger than the previous, it replaces MEDIA_TIMEOUT with
the new value, extending the media timeout circuit breaker; otherwise the new value, extending the media timeout circuit breaker;
it keeps the original value of MEDIA_TIMEOUT. This process is known otherwise, it keeps the original value of MEDIA_TIMEOUT. This
as reconsidering the media timeout circuit breaker. process is known as reconsidering the media timeout circuit breaker.
If MEDIA_TIMEOUT consecutive RTCP packets are received indicating If MEDIA_TIMEOUT consecutive RTCP packets are received indicating
that the media being sent was not received, and the media timeout that the media being sent was not received, and the media timeout
circuit breaker has not been cancelled, then the media timeout circuit breaker has not been canceled, then the media timeout circuit
circuit breaker triggers. When the media timeout circuit breaker breaker triggers. When the media timeout circuit breaker triggers,
triggers, the sender SHOULD cease transmission (see Section 4.5). the sender SHOULD cease transmission (see Section 4.5).
When stopping sending an RTP stream, a sender MUST cancel the When stopping sending an RTP stream, a sender MUST cancel the
corresponding media timeout circuit breaker. corresponding media timeout circuit breaker.
4.3. RTP/AVP Circuit Breaker #3: Congestion 4.3. RTP/AVP Circuit Breaker #3: Congestion
If RTP data packets are being sent, and the corresponding RTCP SR or If RTP data packets are being sent and the corresponding RTCP SR or
RR packets show non-zero packet loss fraction and increasing extended RR packets show non-zero packet loss fraction and increasing extended
highest sequence number received, then those RTP data packets are highest sequence number received, then those RTP data packets are
arriving at the receiver, but some degree of congestion is occurring. arriving at the receiver, but some degree of congestion is occurring.
The RTP/AVP profile [RFC3551] states that: The RTP/AVP profile [RFC3551] states that:
If best-effort service is being used, RTP receivers SHOULD monitor If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured network conditions would achieve an average throughput, measured
on a reasonable time scale, that is not less than the throughput on a reasonable timescale, that is not less than [the throughput]
the RTP flow is achieving. This condition can be satisfied by the RTP flow is achieving. This condition can be satisfied by
implementing congestion control mechanisms to adapt the implementing congestion control mechanisms to adapt the
transmission rate (or the number of layers subscribed for a transmission rate (or the number of layers subscribed for a
layered multicast session), or by arranging for a receiver to layered multicast session), or by arranging for a receiver to
leave the session if the loss rate is unacceptably high. leave the session if the loss rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in time scale and as an "order-of-magnitude" comparison in timescale and throughput.
throughput. The time scale on which TCP throughput is measured is The timescale on which TCP throughput is measured is the round-
the round-trip time of the connection. In essence, this trip time of the connection. In essence, this requirement states
requirement states that it is not acceptable to deploy an that it is not acceptable to deploy an application (using RTP or
application (using RTP or any other transport protocol) on the any other transport protocol) on the best-effort Internet which
best-effort Internet which consumes bandwidth arbitrarily and does consumes bandwidth arbitrarily and does not compete fairly with
not compete fairly with TCP within an order of magnitude. TCP within an order of magnitude.
The phase "order of magnitude" in the above means within a factor of The phase "order of magnitude" in the above means within a factor of
ten, approximately. In order to implement this, it is necessary to ten, approximately. In order to implement this, it is necessary to
estimate the throughput a bulk TCP connection would achieve over the estimate the throughput a bulk TCP connection would achieve over the
path. For a long-lived TCP Reno connection, it has been shown that path. For a long-lived TCP Reno connection, it has been shown that
the TCP throughput, X, in bytes per second, can be estimated using the TCP throughput, X, in bytes per second, can be estimated as
[Padhye]: follows [Padhye]:
s s
X = ------------------------------------------------------------- X = -------------------------------------------------------------
Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p))) 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 This is the same approach to estimated TCP throughput that is used in
[RFC5348]. Under conditions of low packet loss the second term on [RFC5348]. Under conditions of low packet loss, the second term on
the denominator is small, so this formula can be approximated with the denominator is small, so this formula can be approximated with
reasonable accuracy as follows [Mathis]: reasonable accuracy as follows [Mathis]:
s s
X = ---------------- X = ----------------
Tr*sqrt(2*b*p/3) Tr*sqrt(2*b*p/3)
It is RECOMMENDED that this simplified throughput equation be used, It is RECOMMENDED that this simplified throughput equation be used
since the reduction in accuracy is small, and it is much simpler to since the reduction in accuracy is small, and it is much simpler to
calculate than the full equation. Measurements have shown that the calculate than the full equation. Measurements have shown that the
simplified TCP throughput equation is effective as an RTP circuit simplified TCP throughput equation is effective as an RTP circuit
breaker for multimedia flows sent to hosts on residential networks breaker for multimedia flows sent to hosts on residential networks
using ADSL and cable modem links [Singh]. The data shows that the using Asymmetric Digital Subscriber Line (ADSL) and cable modem links
full TCP throughput equation tends to be more sensitive to packet [Singh]. The data shows that the full TCP throughput equation tends
loss and triggers the RTP circuit breaker earlier than the simplified to be more sensitive to packet loss and triggers the RTP circuit
equation. Implementations that desire this extra sensitivity MAY use breaker earlier than the simplified equation. Implementations that
the full TCP throughput equation in the RTP circuit breaker. Initial desire this extra sensitivity MAY use the full TCP throughput
measurements in LTE networks have shown that the extra sensitivity is equation in the RTP circuit breaker. Initial measurements in LTE
helpful in that environment, with the full TCP throughput equation networks have shown that the extra sensitivity is helpful in that
giving a more balanced circuit breaker response than the simplified environment, with the full TCP throughput equation giving a more
TCP equation [Sarker]; other networks might see similar behaviour. 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 No matter what TCP throughput equation is chosen, two parameters need
to be estimated and reported to the sender in order to calculate the 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 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 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 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 for accurate statistics but is the best that can be done with the
standard RTCP mechanisms. standard RTCP mechanisms.
Report blocks in RTCP SR or RR packets contain the packet loss Report blocks in RTCP SR or RR packets contain the packet loss
fraction, rather than the loss event rate, so p cannot be reported fraction, rather than the loss event rate, so p cannot be reported
(TCP typically treats the loss of multiple packets within a single (TCP typically treats the loss of multiple packets within a single
RTT as one loss event, but RTCP RR packets report the overall RTT as one loss event, but RTCP RR packets report the overall
fraction of packets lost, and does not report when the packet losses fraction of packets lost and do not report when the packet losses
occurred). Using the loss fraction in place of the loss event rate occurred). Using the loss fraction in place of the loss event rate
can overestimate the loss. We believe that this overestimate will can overestimate the loss. We believe that this overestimate will
not be significant, given that we are only interested in order of not be significant given that we are only interested in order of
magnitude comparison ([Floyd] section 3.2.1 shows that the difference magnitude comparison (Section 3.2.1 of [Floyd] shows that the
is small for steady-state conditions and random loss, but using the difference is small for steady-state conditions and random loss, but
loss fraction is more conservative in the case of bursty loss). using the loss fraction is more conservative in the case of bursty
loss).
The congestion circuit breaker is therefore: when a sender that is The congestion circuit breaker is therefore as follows. When a
transmitting at least one RTP packet every max(Tdr, Tr) seconds sender that is transmitting at least one RTP packet every max(Tdr,
receives an RTCP SR or RR packet that contains a report block for an Tr) seconds receives an RTCP SR or RR packet that contains a report
SSRC it is using, the sender MUST record the value of the fraction block for an SSRC it is using, the sender MUST record the value of
lost field from the report block, and the time since the last report the fraction lost field from the report block, and the time since the
block was received, for that SSRC. If more than CB_INTERVAL (see last report block was received, for that SSRC. If more than
below) report blocks have been received for that SSRC, the sender CB_INTERVAL (see below) report blocks have been received for that
MUST calculate the average fraction lost over the last CB_INTERVAL SSRC, the sender MUST calculate the average fraction lost over the
reporting intervals, and then estimate the TCP throughput that would last CB_INTERVAL reporting intervals and then estimate the TCP
be achieved over the path using the chosen TCP throughput equation throughput that would be achieved over the path using the chosen TCP
and the measured values of the round-trip time, Tr, the loss event throughput equation and the measured values of the round-trip time,
rate, p (approximated by the average fraction lost, as is described Tr, the loss event rate, p (approximated by the average fraction
below), and the packet size, s. The estimate of the TCP throughput, lost, as is described below), and the packet size, s. The estimate
X, is then compared with the actual sending rate of the RTP stream. of the TCP throughput, X, is then compared with the actual sending
If the actual sending rate of the RTP stream is more than 10 * X, rate of the RTP stream. If the actual sending rate of the RTP stream
then the congestion circuit breaker is triggered. is more than 10 * X, then the congestion circuit breaker is
triggered.
The average fraction lost is calculated based on the sum, over the The average fraction lost is calculated based on the sum (over the
last CB_INTERVAL reporting intervals, of the fraction lost in each last CB_INTERVAL reporting intervals) of the fraction lost in each
reporting interval multiplied by the duration of the corresponding reporting interval that is then multiplied by the duration of the
reporting interval, divided by the total duration of the last corresponding reporting interval and then divided by the total
CB_INTERVAL reporting intervals. The CB_INTERVAL parameter is set duration of the last CB_INTERVAL reporting intervals. The
to: CB_INTERVAL parameter is set to:
CB_INTERVAL = CB_INTERVAL =
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 The parameters that feed into CB_INTERVAL are chosen to give the
congestion control algorithm time to react to congestion. They give congestion control algorithm time to react to congestion. They give
at least three RTCP reports, ten round trip times, and ten groups of 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 frames to adjust the rate to reduce the congestion to a reasonable
level. It is expected that a responsive congestion control algorithm level. It is expected that a responsive congestion control algorithm
will begin to respond with the next group of frames after it receives 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 indication of congestion, so CB_INTERVAL ought to be a much longer
interval than the congestion response. interval than the congestion response.
If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used, 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 and the T_rr_interval parameter is used to reduce the frequency of
regular RTCP reports, then the value Tdr in the above expression for regular RTCP reports, then the value of Tdr in the above expression
the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval, for the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval,
Tdr). Tdr).
The CB_INTERVAL parameter is calculated on joining the session, and The CB_INTERVAL parameter is calculated on joining the session, and
recalculated on receipt of each RTCP packet, after checking whether recalculated on receipt of each RTCP packet, after checking whether
the media timeout circuit breaker or the congestion circuit breaker the media timeout circuit breaker or the congestion circuit breaker
has been triggered. has been triggered.
To ensure a timely response to persistent congestion, implementations To ensure a timely response to persistent congestion, implementations
SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than
5 seconds. Similarly, implementations that use the RTP/AVPF profile 5 seconds. Similarly, implementations that use the RTP/AVPF profile
[RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure [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 to values larger than 4 seconds (the reduced limit for
T_rr_interval follows Section 6.1.3 of T_rr_interval follows Section 6.1.3 of [RFC8108]).
[I-D.ietf-avtcore-rtp-multi-stream]).
The rationale for enforcing a minimum sending rate below which the The rationale for enforcing a minimum sending rate below which the
congestion circuit breaker will not trigger is to avoid spurious congestion circuit breaker will not trigger is to avoid spurious
circuit breaker triggers when the number of packets sent per RTCP circuit breaker triggers when the number of packets sent per RTCP
reporting interval is small, and hence the fraction lost samples are reporting interval is small, and hence, the fraction lost samples are
subject to measurement artefacts. The bound of at least one packet subject to measurement artifacts. The bound of at least one packet
every max(Tdr, Tr) seconds is derived from the one packet per RTT every max(Tdr, Tr) seconds is derived from the one packet per RTT
minimum sending rate of TCP [RFC5405], adapted for use with RTP where minimum sending rate of TCP [RFC8085], which is adapted for use with
the RTCP reporting interval is decoupled from the network RTT. RTP where the RTCP reporting interval is decoupled from the network
RTT.
When the congestion circuit breaker is triggered, the sender SHOULD When the congestion circuit breaker is triggered, the sender SHOULD
cease transmission (see Section 4.5). However, if the sender is able cease transmission (see Section 4.5). However, if the sender is able
to reduce its sending rate by a factor of (approximately) ten, then 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 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 amount) to see if that resolves the congestion. If the sending rate
is reduced in this way and the congestion circuit breaker triggers is reduced in this way and the congestion circuit breaker triggers
again after the next CB_INTERVAL RTCP reporting intervals, the sender again after the next CB_INTERVAL RTCP reporting intervals, the sender
MUST then cease transmission. An example of such a rate reduction MUST then cease transmission. An example of such a rate reduction
might be a video conferencing system that backs off to sending audio might be a video conferencing system that backs off to sending audio
only, before completely dropping the call. If such a reduction in only before completely dropping the call. If such a reduction in
sending rate resolves the congestion problem, the sender MAY sending rate resolves the congestion problem, the sender MAY
gradually increase the rate at which it sends data after a reasonable 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 amount of time has passed, provided it takes care not to cause the
problem to recur ("reasonable" is intentionally not defined here, problem to recur ("reasonable" is intentionally not defined here
since it depends on the application, media codec, and congestion since it depends on the application, media codec, and congestion
control algorithm). control algorithm).
The RTCP reporting interval of the media sender does not affect how The RTCP reporting interval of the media sender does not affect how
quickly congestion circuit breaker can trigger. The timing is based quickly the congestion circuit breaker can trigger. The timing is
on the RTCP reporting interval of the receiver that generates the SR/ based on the RTCP reporting interval of the receiver that generates
RR packets from which the loss rate and RTT estimate are derived the SR/RR packets from which the loss rate and RTT estimate are
(note that RTCP requires all participants in a session to have derived (note that RTCP requires all participants in a session to
similar reporting intervals, else the participant timeout rules in have similar reporting intervals, else the participant timeout rules
[RFC3550] will not work, so this interval is likely similar to that in [RFC3550] will not work, so this interval is likely similar to
of the sender). If the incoming RTCP SR or RR packets are using a that of the sender). If the incoming RTCP SR or RR packets are using
reduced minimum RTCP reporting interval (as specified in Section 6.2 a reduced minimum RTCP reporting interval (as specified in
of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that Section 6.2 of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]),
reduced RTCP reporting interval is used when determining if the then that reduced RTCP reporting interval is used when determining if
circuit breaker is triggered. the circuit breaker is triggered.
If there are more media streams that can be reported in a single RTCP 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 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 exceeds the network MTU, then the receiver will report on a subset of
sources in each reporting interval, with the subsets selected round- sources in each reporting interval with the subsets selected round-
robin across multiple intervals so that all sources are eventually robin across multiple intervals so that all sources are eventually
reported [RFC3550]. When generating such round-robin RTCP reports, reported [RFC3550]. When generating such round-robin RTCP reports,
priority SHOULD be given to reports on sources that have high packet priority SHOULD be given to reports on sources that have high packet
loss rates, to ensure that senders are aware of network congestion loss rates to ensure that senders are aware of network congestion
they are causing (this is an update to [RFC3550]). they are causing (this is an update to [RFC3550]).
4.4. RTP/AVP Circuit Breaker #4: Media Usability 4.4. RTP/AVP Circuit Breaker #4: Media Usability
Applications that use RTP are generally tolerant to some amount of Applications that use RTP are generally tolerant to some amount of
packet loss. How much packet loss can be tolerated will depend on packet loss. How much packet loss can be tolerated will depend on
the application, media codec, and the amount of error correction and the application, media codec, and the amount of error correction and
packet loss concealment that is applied. There is an upper bound on packet loss concealment that is applied. There is an upper bound on
the amount of loss that can be corrected, however, beyond which the the amount of loss that can be corrected, however, beyond which the
media becomes unusable. Similarly, many applications have some upper media becomes unusable. Similarly, many applications have some upper
bound on the media capture to play-out latency that can be tolerated bound on the media capture to play-out latency that can be tolerated
before the application becomes unusable. The latency bound will before the application becomes unusable. The latency bound will
depend on the application, but typical values can range from the depend on the application, but typical values can range from the
order of a few hundred milliseconds for voice telephony and order of a few hundred milliseconds for voice telephony and
interactive conferencing applications, up to several seconds for some interactive conferencing applications up to several seconds for some
video-on-demand systems. video-on-demand systems.
As a final circuit breaker, RTP senders SHOULD monitor the reported As a final circuit breaker, RTP senders SHOULD monitor the reported
packet loss and delay to estimate whether the media is likely to be 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 suitable for the intended purpose. If the packet loss rate and/or
latency is such that the media has become unusable, and has remained latency is such that the media has become unusable and has remained
unusable for a significant time period, then the application SHOULD unusable for a significant time period, then the application SHOULD
cease transmission. Similarly, receivers SHOULD monitor the quality cease transmission. Similarly, receivers SHOULD monitor the quality
of the media they receive, and if the quality is unusable for a of the media they receive, and if the quality is unusable for a
significant time period, they SHOULD terminate the session. This significant time period, they SHOULD terminate the session. This
memo intentionally does not define a bound on the packet loss rate or memo intentionally does not define a bound on the packet loss rate or
latency that will result in unusable media, as these are highly latency that will result in unusable media, as these are highly
application dependent. Similarly, the time period that is considered application dependent. Similarly, the time period that is considered
significant is application dependent, but is likely on the order of significant is application dependent but is likely on the order of
seconds, or tens of seconds. seconds, or tens of seconds.
Sending media that suffers from such high packet loss or latency that Sending media that suffers from such high packet loss or latency that
it is unusable at the receiver is both wasteful of resources, and of it is unusable at the receiver is both wasteful of resources and is
no benefit to the user of the application. It also is highly likely of no benefit to the user of the application. It also is highly
to be congesting the network, and disrupting other applications. As likely to be congesting the network and disrupting other
such, the congestion circuit breaker will almost certainly trigger to applications. As such, the congestion circuit breaker will almost
stop flows where the media would be unusable due to high packet loss certainly trigger to stop flows where the media would be unusable due
or latency. However, in pathological scenarios where the congestion to high packet loss or latency. However, in pathological scenarios
circuit breaker does not stop the flow, it is desirable to prevent where the congestion circuit breaker does not stop the flow, it is
the application sending unnecessary traffic that might disrupt other desirable to prevent the application sending unnecessary traffic that
uses of the network. The role of the media usability circuit breaker might disrupt other uses of the network. The role of the media
is to protect the network in such cases. usability circuit breaker is to protect the network in such cases.
4.5. Ceasing Transmission 4.5. Ceasing Transmission
What it means to cease transmission depends on the application. This What it means to cease transmission depends on the application. This
could mean stopping a single RTP flow, or it could mean that multiple could mean stopping a single RTP flow or it could mean that multiple
bundled RTP flows are stopped. The intention is that the application bundled RTP flows are stopped. The intention is that the application
will stop sending RTP data packets on a particular 5-tuple (transport will stop sending RTP data packets on a particular 5-tuple (transport
protocol, source and destination ports, source and destination IP protocol, source and destination ports, source and destination IP
addresses), until whatever network problem that triggered the RTP addresses) until whatever network problem that triggered the RTP
circuit breaker has dissipated. RTP flows halted by the circuit circuit breaker has dissipated. RTP flows halted by the circuit
breaker SHOULD NOT be restarted automatically unless the sender has breaker SHOULD NOT be restarted automatically unless the sender has
received information that the congestion has dissipated, or can received information that the congestion has dissipated or can
reasonably be expected to have dissipated. What could trigger this reasonably be expected to have dissipated. What could trigger this
expectation is necessarily application dependent, but could be, for expectation is necessarily application dependent, but could be, for
example, an indication that a competing flow has finished and freed example, an indication that a competing flow has finished and freed
up some capacity, or for an application running on a mobile device, up some capacity, or for an application running on a mobile device it
that the device moved to a new location so the flow would traverse a could indicate that the device moved to a new location so the flow
different path if it were restarted. Ideally, a human user will be would traverse a different path if it were restarted. Ideally, a
involved in the decision to try to restart the flow, since that user human user will be involved in the decision to try to restart the
will eventually give up if the flows repeatedly trigger the circuit flow since that user will eventually give up if the flows repeatedly
breaker. This will help avoid problems with automatic redial systems trigger the circuit breaker. This will help avoid problems with
from congesting the network. automatic redial systems from congesting the network.
It is recognised that the RTP implementation in some systems might It is recognized that the RTP implementation in some systems might
not be able to determine if a flow set-up request was initiated by a not be able to determine if a flow setup request was initiated by a
human user, or automatically by some scripted higher-level component human user or automatically by some scripted higher-level component
of the system. These implementations MUST rate limit attempts to 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 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 the circuit breaker so that the reaction to a triggered circuit
breaker lasts for at least the triggering interval breaker lasts for at least the triggering interval [RFC8084].
[I-D.ietf-tsvwg-circuit-breaker].
The RTP circuit breaker will only trigger, and cease transmission, The RTP circuit breaker will only trigger, and cease transmission,
for media flows subject to long-term persistent congestion. Such for media flows subject to long-term persistent congestion. Such
flows are likely to have poor quality and usability for some time flows are likely to have poor quality and usability for some time
before the circuit breaker triggers. Implementations can monitor before the circuit breaker triggers. Implementations can monitor
RTCP Reception Report blocks being returned for their media flows, RTCP Receiver Report blocks being returned for their media flows and
and might find it beneficial to use this information to provide a might find it beneficial to use this information to provide a user
user interface cue that problems are occurring, in advance of the interface cue that problems are occurring in advance of the circuit
circuit breaker triggering. breaker triggering.
5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles
Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)
[RFC4585] allows receivers to send early RTCP reports in some cases, [RFC4585] allows receivers to send early RTCP reports, in some cases,
to inform the sender about particular events in the media stream. to inform the sender about particular events in the media stream.
There are several use cases for such early RTCP reports, including There are several use cases for such early RTCP reports, including
providing rapid feedback to a sender about the onset of congestion. providing rapid feedback to a sender about the onset of congestion.
The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF 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 profile that is treated the same in the context of the RTP circuit
breaker. These feedback profiles are often used with non-compound breaker. These feedback profiles are often used with non-compound
RTCP reports [RFC5506] to reduce the reporting overhead. RTCP reports [RFC5506] to reduce the reporting overhead.
Receiving rapid feedback about congestion events potentially allows Receiving rapid feedback about congestion events potentially allows
congestion control algorithms to be more responsive, and to better congestion control algorithms to be more responsive and to better
adapt the media transmission to the limitations of the network. It adapt the media transmission to the limitations of the network. It
is expected that many RTP congestion control algorithms will adopt is expected that many RTP congestion control algorithms will adopt
the RTP/AVPF profile or the RTP/SAVPF profile for this reason, the RTP/AVPF profile or the RTP/SAVPF profile for this reason and
defining new transport layer feedback reports that suit their thus define new transport-layer feedback reports that suit their
requirements. Since these reports are not yet defined, and likely requirements. Since these reports are not yet defined, and likely
very specific to the details of the congestion control algorithm very specific to the details of the congestion control algorithm
chosen, they cannot be used as part of the generic RTP circuit chosen, they cannot be used as part of the generic RTP circuit
breaker. breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback 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 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 the congestion circuit breaker (they do not contain the information
needed by the congestion circuit breaker algorithm), but MUST be needed by the congestion circuit breaker algorithm) but MUST be
counted as received packets for the RTCP timeout circuit breaker. counted as received packets for the RTCP timeout circuit breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that contain RTCP SR or RR packets MUST be processed by the rules that contain RTCP SR or RR packets MUST be processed by the
congestion circuit breaker as if they were sent as regular RTCP congestion circuit breaker as if they were sent as regular RTCP
reports, and counted towards the circuit breaker conditions specified reports and counted towards the circuit breaker conditions specified
in Section 4 of this memo. This will potentially make the RTP in Section 4 of this memo. This will potentially make the RTP
circuit breaker trigger earlier than it would if the RTP/AVPF profile circuit breaker trigger earlier than it would if the RTP/AVPF profile
was not used. was not used.
When using ECN with RTP (see Section 7), early RTCP feedback packets When using ECN with RTP (see Section 7), early RTCP feedback packets
can contain ECN feedback reports. The count of ECN-CE marked packets can contain ECN feedback reports. The count of ECN-CE-marked packets
contained in those ECN feedback reports is counted towards the number contained in those ECN feedback reports is counted towards the number
of lost packets reported if the ECN Feedback Report is sent in a 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 compound RTCP packet along with an RTCP SR/RR report packet. Reports
of ECN-CE packets sent as reduced-size RTCP ECN feedback packets of ECN-CE packets sent as reduced-size RTCP ECN feedback packets
without an RTCP SR/RR packet MUST be ignored. without an RTCP SR/RR packet MUST be ignored.
These rules are intended to allow the use of low-overhead RTP/AVPF These rules are intended to allow the use of low-overhead RTP/AVPF
feedback for generic NACK messages without triggering the RTP circuit feedback for generic NACK messages without triggering the RTP circuit
breaker. This is expected to make such feedback suitable for RTP breaker. This is expected to make such feedback suitable for RTP
congestion control algorithms that need to quickly report loss events congestion control algorithms that need to quickly report loss events
in between regular RTCP reports. The reaction to reduced-size RTCP in between regular RTCP reports. The reaction to reduced-size RTCP
SR/RR packets is to allow such algorithms to send feedback that can SR/RR packets is to allow such algorithms to send feedback that can
trigger the circuit breaker, when desired. trigger the circuit breaker when desired.
The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval
parameter that can be used to adjust the regular RTCP reporting parameter that can be used to adjust the regular RTCP reporting
interval. The use of the T_rr_interval parameter changes the interval. The use of the T_rr_interval parameter changes the
behaviour of the RTP circuit breaker, as described in Section 4. behavior of the RTP circuit breaker, as described in Section 4.
6. Impact of RTCP Extended Reports (XR) 6. Impact of RTCP Extended Reports (XR)
RTCP Extended Report (XR) blocks provide additional reception quality RTCP Extended Report (XR) blocks provide additional reception quality
metrics, but do not change the RTCP timing rules. Some of the RTCP metrics, but do not change the RTCP timing rules. Some of the RTCP
XR blocks provide information that might be useful for congestion XR blocks provide information that might be useful for congestion
control purposes, others provide non-congestion-related metrics. control purposes, others provide non-congestion-related metrics.
With the exception of RTCP XR ECN Summary Reports (see Section 7), 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 the presence of RTCP XR blocks in a compound RTCP packet does not
affect the RTP circuit breaker algorithm. For consistency and ease affect the RTP circuit breaker algorithm. For consistency and ease
of implementation, only the reception report blocks contained in RTCP of implementation, only the receiver report blocks contained in RTCP
SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets, SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets
are used by the RTP circuit breaker algorithm. are used by the RTP circuit breaker algorithm.
7. Impact of Explicit Congestion Notification (ECN) 7. Impact of Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the RTCP timeout circuit The use of ECN for RTP flows does not affect the RTCP timeout circuit
breaker (Section 4.1) or the media timeout circuit breaker breaker (Section 4.1) or the media timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply (Section 4.2) since these are both connectivity checks that simply
determinate if any packets are being received. determinate if any packets are being received.
At the time of this writing, there's no consensus on how the receipt At the time of this writing, there's no consensus on how the receipt
of ECN feedback will impact the congestion circuit breaker of ECN feedback will impact the congestion circuit breaker
(Section 4.3) or indeed whether the congestion circuit breaker ought (Section 4.3) or indeed whether the congestion circuit breaker ought
to take ECN feedback into account. A future version of this memo is to take ECN feedback into account. A future replacement of this memo
expected to provide guidance for implementers. is expected to provide guidance for implementers.
For the media usability circuit breaker (Section 4.4), ECN-CE marked 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 packets arrive at the receiver, and if they arrive in time, they will
be decoded and rendered as normal. Accordingly, receipt of such be decoded and rendered as normal. Accordingly, receipt of such
packets ought not affect the usability of the media, and the arrival packets ought not affect the usability of the media, and the arrival
of RTCP feedback indicating their receipt is not expected to impact of RTCP feedback indicating their receipt is not expected to impact
the operation of the media usability circuit breaker. the operation of the media usability circuit breaker.
8. Impact of Bundled Media and Layered Coding 8. Impact of Bundled Media and Layered Coding
The RTP circuit breaker operates on a per-RTP session basis. An RTP The RTP circuit breaker operates on a per-RTP session basis. An RTP
sender that participates in several RTP sessions MUST treat each RTP sender that participates in several RTP sessions MUST treat each RTP
session independently with regards to the RTP circuit breaker. session independently with regards to the RTP circuit breaker.
An RTP sender can generate several media streams within a single RTP An RTP sender can generate several media streams within a single RTP
session, with each stream using a different SSRC. This can happen if session, with each stream using a different SSRC. This can happen if
bundled media are in use, when using simulcast, or when using layered bundled media are in use when using simulcast or when using layered
media coding. By default, each SSRC will be treated independently by media coding. By default, each SSRC will be treated independently by
the RTP circuit breaker. However, the sender MAY choose to treat the 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 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 trigger for one flow applies to the group of flows as a whole and
either causes the entire group to cease transmission, or the sending either causes the entire group to cease transmission or causes the
rate of the group to reduce by a factor of ten, depending on the RTP sending rate of the group to reduce by a factor of ten, depending on
circuit breaker triggered. Grouping flows in this way is expected to the RTP circuit breaker triggered. Grouping flows in this way is
be especially useful for layered flows sent using multiple SSRCs, as expected to be especially useful for layered flows sent using
it allows the layered flow to react as a whole, ceasing transmission multiple SSRCs as it allows the layered flow to react as a whole,
on the enhancement layers first to reduce sending rate if necessary, thus ceasing transmission on the enhancement layers first to reduce
rather than treating each layer independently. Care needs to be sending rate, if necessary, rather than treating each layer
taken if the different media streams sent on a single transport layer independently. Care needs to be taken if the different media streams
flow use different DSCP values [RFC7657], sent on a single transport-layer flow use different Differentiated
[I-D.ietf-tsvwg-rtcweb-qos], since congestion could be experienced Services Code Point (DSCP) values [RFC7657] [WebRTC-QoS] since
differently depending on the DSCP marking. Accordingly, RTP media congestion could be experienced differently depending on the DSCP
streams with different DSCP values SHOULD NOT be considered as a marking. Accordingly, RTP media streams with different DSCP values
group when evaluating the RTP Circuit Breaker conditions. SHOULD NOT be considered as a group when evaluating the RTP circuit
breaker conditions.
9. Security Considerations 9. Security Considerations
The security considerations of [RFC3550] apply. The security considerations of [RFC3550] apply.
If the RTP/AVPF profile is used to provide rapid RTCP feedback, the If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
security considerations of [RFC4585] apply. If ECN feedback for RTP security considerations of [RFC4585] apply. If ECN feedback for RTP
over UDP/IP is used, the security considerations of [RFC6679] apply. over UDP/IP is used, the security considerations of [RFC6679] apply.
If non-authenticated RTCP reports are used, an on-path attacker can If non-authenticated RTCP reports are used, an on-path attacker can
trivially generate fake RTCP packets that indicate high packet loss trivially generate fake RTCP packets that indicate high packet loss
rates, causing the circuit breaker to trigger and disrupt an RTP rates and thus cause the circuit breaker to trigger and disrupt an
session. This is somewhat more difficult for an off-path attacker, RTP session. This is somewhat more difficult for an off-path
due to the need to guess the randomly chosen RTP SSRC value and the attacker due to the need to guess the randomly chosen RTP SSRC value
RTP sequence number. This attack can be avoided if RTCP packets are and the RTP sequence number. This attack can be avoided if RTCP
authenticated; authentication options are discussed in [RFC7201]. packets are authenticated; authentication options are discussed in
[RFC7201].
Timely operation of the RTP circuit breaker depends on the choice of Timely operation of the RTP circuit breaker depends on the choice of
RTCP reporting interval. If the receiver has a reporting interval RTCP reporting interval. If the receiver has a reporting interval
that is overly long, then the responsiveness of the circuit breaker that is overly long, then the responsiveness of the circuit breaker
decreases. In the limit, the RTP circuit breaker can be disabled for decreases. In the limit, the RTP circuit breaker can be disabled for
all practical purposes by configuring an RTCP reporting interval that all practical purposes by configuring an RTCP reporting interval that
is many minutes duration. This issue is not specific to the circuit has a duration of many minutes. This issue is not specific to the
breaker: long RTCP reporting intervals also prevent reception quality circuit breaker: long RTCP reporting intervals also prevent reception
reports, feedback messages, codec control messages, etc., from being quality reports, feedback messages, codec control messages, etc.,
used. Implementations are expected to impose an upper limit on the from being used. Implementations are expected to impose an upper
RTCP reporting interval they are willing to negotiate (based on the limit on the RTCP reporting interval they are willing to negotiate
session bandwidth and RTCP bandwidth fraction) when using the RTP (based on the session bandwidth and RTCP bandwidth fraction) when
circuit breaker, as discussed in Section 4.3. using the RTP circuit breaker, as discussed in Section 4.3.
10. IANA Considerations
There are no actions for IANA.
11. Acknowledgements
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.
12. References 10. References
12.1. Normative References 10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <http://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>. July 2003, <http://www.rfc-editor.org/info/rfc3550>.
skipping to change at page 22, line 5 skipping to change at page 22, line 10
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008, RFC 5348, DOI 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>. <http://www.rfc-editor.org/info/rfc5348>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN) and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>. 2012, <http://www.rfc-editor.org/info/rfc6679>.
12.2. Informative References 10.2. Informative References
[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
"Equation-Based Congestion Control for Unicast "Equation-Based Congestion Control for Unicast
Applications", Proceedings of the ACM SIGCOMM Applications", ACM SIGCOMM Computer Communication
conference, 2000, DOI 10.1145/347059.347397, August 2000. Review, Volume 30, Issue 4, pages 43-56,
DOI 10.1145/347059.347397, August 2000.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and D. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers",
draft-ietf-tsvwg-circuit-breaker-15 (work in progress),
April 2016.
[I-D.ietf-tsvwg-rtcweb-qos]
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP
Packet Markings for WebRTC QoS", draft-ietf-tsvwg-rtcweb-
qos-17 (work in progress), May 2016.
[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The [Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
macroscopic behavior of the TCP congestion avoidance Macroscopic Behavior of the TCP Congestion Avoidance
algorithm", ACM SIGCOMM Computer Communication Algorithm", ACM SIGCOMM Computer Communication
Review 27(3), DOI 10.1145/263932.264023, July 1997. Review, Volume 27, Issue 3, pages 67-82,
DOI 10.1145/263932.264023, July 1997.
[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, [Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its Empirical "Modeling TCP Throughput: A Simple Model and its Empirical
Validation", Proceedings of the ACM SIGCOMM Validation", ACM SIGCOMM Computer Communication
conference, 1998, DOI 10.1145/285237.285291, August 1998. Review Volume 30, Issue 4, pages 303-314,
DOI 10.1145/285237.285291, August 1998.
[RFC2862] Civanlar, M. and G. Cash, "RTP Payload Format for Real- [RFC2862] Civanlar, M. and G. Cash, "RTP Payload Format for Real-
Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000, Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000,
<http://www.rfc-editor.org/info/rfc2862>. <http://www.rfc-editor.org/info/rfc2862>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001, RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>. <http://www.rfc-editor.org/info/rfc3168>.
skipping to change at page 23, line 15 skipping to change at page 23, line 5
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>. February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>. 2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405,
DOI 10.17487/RFC5405, November 2008,
<http://www.rfc-editor.org/info/rfc5405>.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009, RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009,
<http://www.rfc-editor.org/info/rfc5450>. <http://www.rfc-editor.org/info/rfc5450>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>. 2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
skipping to change at page 24, line 29 skipping to change at page 24, line 15
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657, (Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015, DOI 10.17487/RFC7657, November 2015,
<http://www.rfc-editor.org/info/rfc7657>. <http://www.rfc-editor.org/info/rfc7657>.
[RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx) [RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx)
Concepts, Abstract Mechanism, and Requirements", RFC 7713, Concepts, Abstract Mechanism, and Requirements", RFC 7713,
DOI 10.17487/RFC7713, December 2015, DOI 10.17487/RFC7713, December 2015,
<http://www.rfc-editor.org/info/rfc7713>. <http://www.rfc-editor.org/info/rfc7713>.
[RFC8084] Fairhurst, G., "Network Transport Circuit Breakers",
BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
<http://www.rfc-editor.org/info/rfc8084>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <http://www.rfc-editor.org/info/rfc8085>.
[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,
<http://www.rfc-editor.org/info/rfc8108>.
[Sarker] Sarker, Z., Singh, V., and C. Perkins, "An Evaluation of [Sarker] Sarker, Z., Singh, V., and C. Perkins, "An Evaluation of
RTP Circuit Breaker Performance on LTE Networks", RTP Circuit Breaker Performance on LTE Networks",
Proceedings of the IEEE Infocom workshop on Communication Proceedings of the IEEE INFOCOM Workshop on Communication
and Networking Techniques for Contemporary Video, 2014, and Networking Techniques for Contemporary Video,
April 2014. DOI 10.1109/INFCOMW.2014.6849240, April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C. Perkins, [Singh] Singh, V., McQuistin, S., Ellis, M., and C. Perkins,
"Circuit Breakers for Multimedia Congestion Control", "Circuit Breakers for Multimedia Congestion Control",
Proceedings of the International Packet Video Proceedings of the 2013 20th International Packet Video
Workshop, 2013, DOI 10.1109/PV.2013.6691439, December Workshop (PV), DOI 10.1109/PV.2013.6691439, December 2013.
2013.
[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.
Acknowledgements
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 Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
skipping to change at page 25, line 4 skipping to change at page 25, line 23
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
Varun Singh Varun Singh
Nemu Dialogue Systems Oy CALLSTATS I/O Oy
Runeberginkatu 4c A 4 Runeberginkatu 4c A 4
Helsinki 00100 Helsinki 00100
Finland Finland
Email: varun.singh@iki.fi Email: varun@callstats.io
URI: http://www.callstats.io/ URI: https://www.callstats.io/about
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