draft-ietf-avtcore-rtp-circuit-breakers-10.txt   draft-ietf-avtcore-rtp-circuit-breakers-11.txt 
AVTCORE Working Group C. S. Perkins AVTCORE Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Updates: 3550 (if approved) V. Singh Updates: 3550 (if approved) V. Singh
Intended status: Standards Track Aalto University Intended status: Standards Track Aalto University
Expires: September 24, 2015 March 23, 2015 Expires: April 18, 2016 October 16, 2015
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-10 draft-ietf-avtcore-rtp-circuit-breakers-11
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 the applications, then network congestion will is not implemented in the applications, then network congestion will
deteriorate the user's multimedia experience. This document does not deteriorate the user's multimedia experience. This document does not
propose a congestion control algorithm; instead, it defines a minimal propose a congestion control algorithm; instead, it defines a minimal
set of RTP "circuit-breakers". Circuit-breakers are conditions under set of RTP circuit-breakers. Circuit-breakers are conditions under
which an RTP sender needs to stop transmitting media data in order to which an RTP sender needs to stop transmitting media data in order 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 severe congestion, all RTP applications running on in the absence of severe congestion, all RTP applications running on
best-effort IP networks will be able to run without triggering these best-effort IP networks will be able to run without triggering these
circuit breakers. Any future RTP congestion control specification circuit breakers. Any future RTP congestion control specification
will be expected to operate within the constraints defined by these will be expected to operate within the constraints defined by these
circuit breakers. circuit breakers.
Status of This Memo Status of This Memo
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- 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 Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 24, 2015. This Internet-Draft will expire on April 18, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the Copyright (c) 2015 IETF Trust and the persons identified as the
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 7
4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7 4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout . . . . . . . . 9
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8 4.2. RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . . 10
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 9 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 11
4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 13 4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 15
4.5. Choice of Circuit Breaker Interval . . . . . . . . . . . 14 4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 16
4.6. Ceasing Transmission . . . . . . . . . . . . . . . . . . 15 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 16 6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 18
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 17 7. Impact of Explicit Congestion Notification (ECN) . . . . . . 18
7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 17 8. Impact of Bundled Media and Layered Coding . . . . . . . . . 18
8. Impact of Explicit Congestion Notification (ECN) . . . . . . 18 9. Security Considerations . . . . . . . . . . . . . . . . . . . 19
9. Impact of Bundled Media and Layered Coding . . . . . . . . . 18 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
10. Security Considerations . . . . . . . . . . . . . . . . . . . 18 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 20
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 20
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 19 12.1. Normative References . . . . . . . . . . . . . . . . . . 20
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 19 12.2. Informative References . . . . . . . . . . . . . . . . . 20
13.1. Normative References . . . . . . . . . . . . . . . . . . 19 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 23
13.2. Informative References . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22
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. Rather, it proposes a minimal set of "RTP circuit algorithm. Instead, we propose a small set of RTP circuit breakers.
breakers"; conditions under which there is general agreement that an These are conditions under which there is general agreement that an
RTP flow is causing serious congestion, and ought to cease RTP flow is causing serious congestion, and hence ought to cease
transmission. The RTP circuit breakers proposed in this memo are a transmission. The RTP circuit breakers proposed in this memo are a
specific instance of the general class of network transport circuit specific instance of the general class of network transport circuit
breakers [I-D.ietf-tsvwg-circuit-breaker], designed to act as a breakers [I-D.ietf-tsvwg-circuit-breaker], designed to act as a
protection mechanism of last resort to avoid persistent congestion. protection mechanism of last resort to avoid persistent congestion.
It is expected that future standards-track congestion control It is expected that future standards-track congestion control
algorithms for RTP will operate within the envelope defined by this algorithms for RTP will operate within the envelope defined by this
memo. memo.
2. Terminology 2. Background
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
This interpretation of these key words applies only when written in
ALL CAPS. Mixed- or lower-case uses of these key words are not to be
interpreted as carrying special significance in this memo.
3. 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 matches the available network bandwidth. to one receiver, so that it matches the available network bandwidth.
Such adaptation needs to be done in a way that limits the disruption Such adaptation needs to be done in a way that limits the disruption
to the user experience caused by both packet loss and excessive rate to the user experience caused by both packet loss and excessive rate
changes. Congestion control for multicast flows is outside the scope changes. Congestion control for multicast flows is outside the scope
of this memo. Multicast traffic needs different solutions, since the of this memo. Multicast traffic needs different solutions, since the
available bandwidth estimator for a group of receivers will differ available bandwidth estimator for a group of receivers will differ
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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: 1) a receiver can infer the onset of
congestion by observing an increase in one-way delay caused by queue congestion by observing an increase in one-way delay caused by queue
build-up within the network; 2) if Explicit Congestion Notification build-up within the network; 2) if Explicit Congestion Notification
(ECN) [RFC3168] is supported, the network can signal the presence of (ECN) [RFC3168] is supported, the network can signal the presence of
congestion by marking packets using ECN Congestion Experienced (CE) congestion by marking packets using ECN Congestion Experienced (CE)
marks; or 3) in the extreme case, congestion will cause packet loss marks; or 3) in the extreme case, congestion will cause packet loss
that can be detected by observing a gap in the received RTP sequence that can be detected by observing a gap in the received RTP sequence
numbers. Once the onset of congestion is observed, the receiver has numbers.
to send feedback to the sender to indicate that the transmission rate
needs to be reduced. How the sender reduces the transmission rate is Once the onset of congestion is observed, the receiver has to send
feedback to the sender to indicate that the transmission rate needs
to be reduced. How the sender reduces the transmission rate is
highly dependent on the media codec being used, and is outside the 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 Reception 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
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dependent on signalling. dependent on signalling.
o Finally, Explicit Congestion Notification (ECN) for RTP over UDP o Finally, Explicit Congestion Notification (ECN) for RTP over UDP
[RFC6679] can be used to provide feedback on the number of packets [RFC6679] can be used to provide feedback on the number of packets
that received an ECN Congestion Experienced (CE) mark. This RTCP that received an ECN Congestion Experienced (CE) mark. This RTCP
extension builds on the RTP/AVPF profile to allow rapid congestion extension builds on the RTP/AVPF profile to allow rapid 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. There are two methods: [RFC5450] represents the transmission times [RFC5450]. Accurate transmission timestamps can
transmission time in terms of a time-offset from the RTP timestamp of be helpful for estimating queuing delays, to get an early indication
the packet, while [RFC6051] includes an explicit NTP-format sending of the onset of congestion.
timestamp (potentially more accurate, but a higher header overhead).
Accurate sending timestamps can be helpful for estimating queuing
delays, to get an early indication of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide 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
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
This interpretation of these key words applies only when written in
ALL CAPS. Mixed- or lower-case uses of these key words are not to be
interpreted as carrying special significance in this memo.
The definition of the RTP circuit breaker is specified in terms of
the following variables:
o Td is the deterministic RTCP reporting interval, as defined in
Section 6.3.1 of [RFC3550].
o Tdr is the sender's estimate of the deterministic RTCP reporting
interval, Td, calculated by a receiver of the data it is sending.
Tdr is not known at the sender, but can be estimated by executing
the algorithm in Section 6.2 of [RFC3550] using the average RTCP
packet size seen at the sender, the number of members reported in
the receiver's SR/RR report blocks, and whether the receiver is
sending SR or RR packets. Tdr is recalculated when each new RTCP
SR/RR report is received, but the media timeout circuit breaker
(see Section 4.2) is only reconsidered when Tdr increases.
o Tr is the network round-trip time, calculated by the sender using
the algorithm in Section 6.4.1 of [RFC3550] and smoothed using an
exponentially weighted moving average as Tr = (0.8 * Tr) + (0.2 *
Tr_new) where Tr_new is the latest RTT estimate obtained from an
RTCP report. The weight is chosen so old estimates decay over k
intervals.
o k is the non-reporting threshold (see Section 4.2).
o Tf is the media framing interval at the sender. For applications
sending at a constant frame rate, Tf is the inter-frame interval.
For applications that switch between a small set of possible frame
rates, for example when sending speech with comfort noise, where
comfort noise frames are sent less often than speech frames, Tf is
set to the longest of the inter-frame intervals of the different
frame rates. For applications that send periodic frames but
dynamically vary their frame rate, Tf is set to the largest inter-
frame interval used in the last 10 seconds. For applications that
send less than one frame every 10 seconds, or that have no concept
of periodic frames (e.g., text conversation [RFC4103], or pointer
events [RFC2862]), Tf is set to the time interval since the
previous frame when each frame is sent.
o G is the frame group size. That is, the number of frames that are
coded together based on a particular sending rate setting. If the
codec used by the sender can change its rate on each frame, G = 1;
otherwise G is set to the number of frames before the codec can
adjust to the new rate. For codecs that have the concept of a
group-of-pictures (GoP), G is likely the GoP length.
o T_rr_interval is the minimal interval between RTCP reports, as
defined in Section 3.4 of [RFC4585]; it is only meaningful for
implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF
profile [RFC5124].
o X is estimated throughput a TCP connection would achieve over a
path, in bytes per second.
o s is the size of RTP packets being sent, in bytes. If the RTP
packets being sent vary in size, then the average size over the
packet comprising the last 4 * G frames MUST be used (this is
intended to be comparable to the four loss intervals used in
[RFC5348]).
o p is the loss event rate, between 0.0 and 1.0, that would be seen
by a TCP connection over a particular path. When used in the RTP
congestion circuit breaker, this is approximated as described in
Section 4.3.
o t_RTO is the retransmission timeout value that would be used by a
TCP connection over a particular path, in seconds. This MUST be
approximated using t_RTO = 4 * Tr when used as part of the RTP
congestion circuit breaker.
o b is the number of packets that are acknowledged by a single TCP
acknowledgement. Following [RFC3448], it is RECOMMENDED that the
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, cannot use that claim to support RTP, but that do not implement RTCP, cannot use
the circuit breaker mechanisms described in this memo. Such the circuit breaker mechanisms described in this memo. Such
implementations SHOULD NOT be used on networks that might be subject implementations SHOULD NOT be used on networks that might be subject
to congestion unless equivalent mechanisms are defined using some to congestion unless equivalent mechanisms are defined using some
non-RTCP feedback channel to report congestion and signal circuit non-RTCP feedback channel to report congestion and signal circuit
breaker conditions. breaker conditions. The RTCP timeout circuit breaker (Section 4.1)
will trigger if an implementation of this memo attempts to interwork
with an endpoint that does not support RTCP. Implementations that
sometimes need to interwork with endpoints that do not support RTCP
need to disable the RTP circuit breakers if they don't receive some
confirmation via signalling that the remote endpoint implements RTCP
(the presence of an SDP "a=rtcp:" attribute in an answer might be
such an indication).
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 synchronisation source (SSRC) SR/RR packets and are reported for each synchronisation source (SSRC)
in the RTP session: 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
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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 behaviour 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 though, and don't give enough information to infrequent to be useful for congestion control, and don't give enough
distinguish a delay change due to routing updates from queuing delay information to distinguish a delay change due to routing updates from
caused by congestion. Accordingly, we cannot use the RTT estimate queuing delay caused by congestion. Accordingly, we cannot use the
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, these cumulative number of packets lost. If considered carefully, these
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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: Media Timeout 4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term
issue affecting only a few packets, perhaps caused by a slow-to-open
firewall or a transient connectivity problem, but if the issue
persists, it is a sign of a more ongoing and significant problem.
Accordingly, if a sender of RTP data packets receives CB_INTERVAL or
more consecutive RTCP SR or RR packets from the same receiver (see
Section 4.5), and those packets correspond to its transmission and
have a non-increasing extended highest sequence number received
field, then that sender SHOULD cease transmission (see Section 4.6).
The extended highest sequence number received field is non-increasing
if the sender receives at least CB_INTERVAL consecutive RTCP SR or RR
packets that report the same value for this field, but it has sent
RTP data packets, at a rate of at least one per RTT, that would have
caused an increase in the reported value if they had reached the
receiver.
The rationale for waiting for CB_INTERVAL or more consecutive RTCP
packets with a non-increasing extended highest sequence number is to
give enough time for transient reception problems to resolve
themselves, but to stop problem flows quickly enough to avoid causing
serious ongoing network congestion. A single RTCP report showing no
reception could be caused by a transient fault, and so will not cease
transmission. Waiting for more than CB_INTERVAL consecutive RTCP
reports before stopping a flow might avoid some false positives, but
could lead to problematic flows running for a long time period
(potentially tens of seconds, depending on the RTCP reporting
interval) before being cut off. Equally, an application that sends
few packets when the packet loss rate is high runs the risk that the
media timeout circuit breaker triggers inadvertently. The chosen
timeout interval is a trade-off between these extremes.
The rationale for enforcing a minimum sending rate below which the
media timeout circuit breaker will not trigger is to avoid spurious
circuit breaker triggers when the number of packets sent per RTCP
reporting interval is small (e.g., a telephony application sends only
two RTP comfort noise packets during a five second RTCP reporting
interval, and both are lost; this is 100% packet loss, but it seems
extreme to terminate the RTP session). The one packet per RTT bound
derives from [RFC5405].
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout An RTCP timeout can occur when RTP data packets are being sent, but
there are no RTCP reports returned from the receiver. This is either
due to a failure of the receiver to send RTCP reports, or a failure
of the return path that is preventing those RTCP reporting from being
delivered. In either case, it is not safe to continue transmission,
since the sender has no way of knowing if it is causing congestion.
In addition to media timeouts, as were discussed in Section 4.1, an An RTP sender that has not received any RTCP SR or RTCP RR packets
RTP session has the possibility of an RTCP timeout. This can occur reporting on the SSRC it is using, for a time period of at least
when RTP data packets are being sent, but there are no RTCP reports three times its deterministic RTCP reporting interval, Td, without
returned from the receiver. This is either due to a failure of the the randomization factor, and using the fixed minimum interval of
receiver to send RTCP reports, or a failure of the return path that Tmin=5 seconds, SHOULD cease transmission (see Section 4.5). The
is preventing those RTCP reporting from being delivered. In either rationale for this choice of timeout is as described in Section 6.2
case, it is not safe to continue transmission, since the sender has of [RFC3550] ("so that implementations which do not use the reduced
no way of knowing if it is causing congestion. Accordingly, an RTP value for transmitting RTCP packets are not timed out by other
sender that has not received any RTCP SR or RTCP RR packets reporting participants prematurely"), as updated by Section 6.1.4 of
on the SSRC it is using for three or more of its RTCP reporting [I-D.ietf-avtcore-rtp-multi-stream] to account for the use of the
intervals SHOULD cease transmission (see Section 4.6). When RTP/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124].
calculating the timeout, the deterministic RTCP reporting interval,
Td, without the randomization factor, and using the fixed minimum
interval of Tmin=5 seconds, MUST be used. The rationale for this
choice of timeout is as described in Section 6.2 of [RFC3550] ("so
that implementations which do not use the reduced value for
transmitting RTCP packets are not timed out by other participants
prematurely"), as updated by Section 6.1.4 of
[I-D.ietf-avtcore-rtp-multi-stream] to account for the use of the RTP
/AVPF 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 [I-D.ietf-avtcore-rtp-multi-stream]).
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
skipping to change at page 9, line 43 skipping to change at page 10, line 35
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, preventing the RTCP timeout circuit breaker from triggering.
4.2. RTP/AVP Circuit Breaker #2: Media Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term
issue affecting only a few RTP packets, perhaps caused by a slow to
open firewall or a transient connectivity problem, but if the issue
persists, it is a sign of a more ongoing and significant problem (a
"media timeout").
The time needed to declare a media timeout depends on the parameters
Tdr, Tr, Tf, and on the non-reporting threshold k. The value of k is
chosen so that when Tdr is large compared to Tr and Tf, receipt of at
least k RTCP reports with non-increasing extended highest sequence
number received gives reasonable assurance that the forward path has
failed, and that the RTP data packets have not been lost by chance.
The RECOMMENDED value for k is 5 reports.
When Tdr < Tf, then RTP data packets are being sent at a rate less
than one per RTCP reporting interval of the receiver, so the extended
highest sequence number received can be expected to be non-increasing
for some receiver RTCP reporting intervals. Similarly, when Tdr <
Tr, some receiver RTCP reporting intervals might pass before the RTP
data packets arrive at the receiver, also leading to reports where
the extended highest sequence number received is non-increasing.
Both issues require the media timeout interval to be scaled relative
to the threshold, k.
The media timeout RTP circuit breaker is therefore as follows. When
starting sending, calculate MEDIA_TIMEOUT using:
MEDIA_TIMEOUT = ceil(k * max(Tf, Tr, Tdr) / Tdr)
When a sender receives an RTCP packet indicating that the media it's
sending is being received, then it cancels the media timeout circuit
breaker. If it is still sending, then it MUST calculate a new value
for MEDIA_TIMEOUT, and set a new media timeout circuit breaker.
If a sender receives an RTCP packet indicating that its media was not
received, it MUST calculate a new value for MEDIA_TIMEOUT. If the
new value is larger than the previous, is replaces MEDIA_TIMEOUT with
the new value, extending the media timeout circuit breaker; otherwise
it keeps the original value of MEDIA_TIMEOUT. This process is known
as reconsidering the media timeout circuit breaker.
If MEDIA_TIMEOUT consecutive RTCP packets are received indicating
that the media being sent was not received, and the media timeout
circuit breaker has not been cancelled, then the media timeout
circuit breaker triggers. When the media timeout circuit breaker
triggers, the sender SHOULD cease transmission (see Section 4.5).
When stopping sending an RTP stream, a sender MUST cancel the
corresponding media timeout circuit breaker.
4.3. RTP/AVP Circuit Breaker #3: Congestion 4.3. RTP/AVP 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
skipping to change at page 10, line 30 skipping to change at page 12, line 30
the round-trip time of the connection. In essence, this the round-trip time of the connection. In essence, this
requirement states that it is not acceptable to deploy an requirement states that it is not acceptable to deploy an
application (using RTP or any other transport protocol) on the application (using RTP or any other transport protocol) on the
best-effort Internet which consumes bandwidth arbitrarily and does best-effort Internet which consumes bandwidth arbitrarily and does
not compete fairly with TCP within an order of magnitude. not compete fairly with 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 TCP connection would achieve over the path. estimate the throughput a TCP connection would achieve over the path.
For a long-lived TCP Reno connection, it has been shown that the TCP For a long-lived TCP Reno connection, it has been shown that the TCP
throughput can be estimated using the following equation [Padhye]: throughput, X, in bytes per second, can be estimated using [Padhye]:
s
X = --------------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))
where:
X is the transmit rate in bytes/second.
s is the packet size in bytes. If data packets vary in size, then
the average size is to be used.
R is the round trip time in seconds.
p is the loss event rate, between 0 and 1.0, of the number of loss
events as a fraction of the number of packets transmitted.
t_RTO is the TCP retransmission timeout value in seconds, generally
approximated by setting t_RTO = 4*R.
b is the number of packets that are acknowledged by a single TCP s
acknowledgement; [RFC3448] recommends the use of b=1 since many X = -------------------------------------------------------------
TCP implementations do not use delayed acknowledgements. 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
[RFC3448]. Under conditions of low packet loss the second term on [RFC3448]. 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 = ----------------
R * sqrt(2*b*p/3) Tr*sqrt(2*b*p/3)
It is RECOMMENDED that this simplified throughout equation be used, It is RECOMMENDED that this simplified throughout 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 ADSL and cable modem links [Singh]. The data shows that the
full TCP throughput equation tends to be more sensitive to packet full TCP throughput equation tends to be more sensitive to packet
loss and triggers the RTP circuit breaker earlier than the simplified loss and triggers the RTP circuit breaker earlier than the simplified
equation. Implementations that desire this extra sensitivity MAY use equation. Implementations that desire this extra sensitivity MAY use
the full TCP throughput equation in the RTP circuit breaker. Initial the full TCP throughput equation in the RTP circuit breaker. Initial
measurements in LTE networks have shown that the extra sensitivity is measurements in LTE networks have shown that the extra sensitivity is
helpful in that environment, with the full TCP throughput equation helpful in that environment, with the full TCP throughput equation
giving a more balanced circuit breaker response than the simplified giving a more balanced circuit breaker response than the simplified
TCP equation [Sarker]; other networks might see similar behaviour. TCP equation [Sarker]; other networks might see similar behaviour.
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, R, 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 does 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 ([Floyd] section 3.2.1 shows that the difference
is small for steady-state conditions and random loss, but using the is small for steady-state conditions and random loss, but using the
loss fraction is more conservative in the case of bursty loss). 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: when a sender that is
transmitting more than one RTP packet per RTT receives an RTCP SR or transmitting at least one RTP packet every max(Tdr, Tr) seconds
RR packet that contains a report block for an SSRC it is using, the receives an RTCP SR or RR packet that contains a report block for an
sender MUST record the value of the fraction lost field in the report SSRC it is using, the sender MUST record the value of the fraction
block and the time since the last report block was received for that lost field in the report block and the time since the last report
SSRC. If more than CB_INTERVAL (see Section 4.5) report blocks have block was received for that SSRC. If more than CB_INTERVAL (see
been received for that SSRC, the sender MUST calculate the average below) report blocks have been received for that SSRC, the sender
fraction lost over the last CB_INTERVAL reporting intervals, and then MUST calculate the average fraction lost over the last CB_INTERVAL
estimate the TCP throughput that would be achieved over the path reporting intervals, and then estimate the TCP throughput that would
using the chosen TCP throughput equation and the measured values of be achieved over the path using the chosen TCP throughput equation
the round-trip time, R, the loss event rate, p (as approximated by and the measured values of the round-trip time, Tr, the loss event
the average fraction lost), and the packet size, s. This estimate of rate, p (approximated by the average fraction lost, as is described
the TCP throughput is then compared with the actual sending rate. If below), and the packet size, s. The estimate of the TCP throughput,
the actual sending rate is more than ten times the TCP throughput X, is then compared with the actual sending rate of the RTP stream.
estimate, then the congestion circuit breaker is triggered. If the actual sending rate of the RTP stream 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 multiplied by the duration of the corresponding
reporting interval, divided by the total duration of the last reporting interval, divided by the total duration of the last
CB_INTERVAL reporting intervals. CB_INTERVAL reporting intervals. The CB_INTERVAL parameter is set
to:
CB_INTERVAL =
ceil(3*min(max(10*G*Tf, 10*Tr, 3*Tdr), max(15, 3*Td))/(3*Tdr))
The parameters that feed into CB_INTERVAL are chosen to give the
congestion control algorithm time to react to congestion. They give
at least three RTCP reports, ten round trip times, and ten groups of
frames to adjust the rate to reduce the congestion to a reasonable
level. It is expected that a responsive congestion control algorithm
will begin to respond with the next group of frames after it receives
indication of congestion, so CB_INTERVAL ought to be a much longer
interval than the congestion response.
If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used,
and the T_rr_interval parameter is used to reduce the frequency of
regular RTCP reports, then the value Tdr in the above expression for
the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval,
Tdr).
The CB_INTERVAL parameter is calculated on joining the session, and
recalculated on receipt of each RTCP packet, after checking whether
the media timeout circuit breaker or the congestion circuit breaker
has been triggered.
To ensure a timely response to persistent congestion, implementations
SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than
5 seconds. Similarly, implementations that use the RTP/AVPF profile
[RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure
T_rr_interval to values larger than 4 seconds (the reduced limit for
T_rr_interval follows Section 6.1.3 of
[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 one packet per RTT bound subject to measurement artefacts. The bound of at least one packet
derives from [RFC5405]. 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
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 4.6). 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
skipping to change at page 13, line 13 skipping to change at page 15, line 27
RR packets from which the loss rate and RTT estimate are derived RR packets from which the loss rate and RTT estimate are derived
(note that RTCP requires all participants in a session to have (note that RTCP requires all participants in a session to have
similar reporting intervals, else the participant timeout rules in similar reporting intervals, else the participant timeout rules in
[RFC3550] will not work, so this interval is likely similar to that [RFC3550] will not work, so this interval is likely similar to that
of the sender). If the incoming RTCP SR or RR packets are using a of the sender). If the incoming RTCP SR or RR packets are using a
reduced minimum RTCP reporting interval (as specified in Section 6.2 reduced minimum RTCP reporting interval (as specified in Section 6.2
of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that
reduced RTCP reporting interval is used when determining if the reduced RTCP reporting interval is used when determining if the
circuit breaker is triggered. circuit breaker is triggered.
As in Section 4.1 and Section 4.2, we use CB_INTERVAL reporting
intervals to avoid triggering the circuit breaker on transient
failures. This circuit breaker is a worst-case condition, and
congestion control needs to be performed to keep well within this
bound. It is expected that the circuit breaker will only be
triggered if the usual congestion control fails for some reason.
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]).
skipping to change at page 14, line 20 skipping to change at page 16, line 29
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 of
no benefit to the user of the application. It also is highly likely no benefit to the user of the application. It also is highly likely
to be congesting the network, and disrupting other applications. As to be congesting the network, and disrupting other applications. As
such, the congestion circuit breaker will almost certainly trigger to such, the congestion circuit breaker will almost certainly trigger to
stop flows where the media would be unusable due to high packet loss stop flows where the media would be unusable due to high packet loss
or latency. However, in pathological scenarios where the congestion or latency. However, in pathological scenarios where the congestion
circuit breaker does not stop the flow, it is desirable that the RTP circuit breaker does not stop the flow, it is desirable that the RTP
application cease sending useless traffic. The role of the media application cease sending useless traffic. The role of the media
usability circuit breaker is to protect the network in such cases. usability circuit breaker is to protect the network in such cases.
4.5. Choice of Circuit Breaker Interval 4.5. Ceasing Transmission
The CB_INTERVAL parameter determines the number of consecutive RTCP
reporting intervals that need to suffer congestion before the media
timeout circuit breaker (see Section 4.1) or the congestion circuit
breaker (see Section 4.3) triggers. It determines the sensitivity
and responsiveness of these circuit breakers.
The CB_INTERVAL parameter is set to min(floor(3+(2.5/Td)), 30) RTCP
reporting intervals, where Td is the deterministic calculated RTCP
interval described in section 6.3.1 of [RFC3550]. This expression
gives an CB_INTERVAL that varies as follows:
Td | CB_INTERVAL | Time to trigger
--------------+------------------------------+-----------------
0.016 seconds | 30 RTCP reporting intervals | 0.48 seconds
0.033 seconds | 30 RTCP reporting intervals | 0.99 seconds
0.100 seconds | 28 RTCP reporting intervals | 2.80 seconds
0.500 seconds | 8 RTCP reporting intervals | 4.00 seconds
1.000 seconds | 5 RTCP reporting intervals | 5.00 seconds
2.000 seconds | 4 RTCP reporting intervals | 8.00 seconds
5.000 seconds | 3 RTCP reporting intervals | 15.00 seconds
10.000 seconds | 3 RTCP reporting intervals | 30.00 seconds
If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used,
and the T_rr_interval parameter is used to reduce the frequency of
regular RTCP reports, then the value Td in the above expression for
the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval, Td).
The CB_INTERVAL parameter is calculated on joining the session, and
recalculated on receipt of each RTCP packet, after checking whether
the media timeout circuit breaker or the congestion circuit breaker
has been triggered.
To ensure a timely response to persistent congestion, implementations
SHOULD NOT configure the RTCP bandwidth such that Td is larger than 5
seconds. Similarly, implementations that use the RTP/AVPF profile
[RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure
T_rr_interval to values larger than 4 seconds (the reduced limit for
T_rr_interval follows Section 6.1.3 of
[I-D.ietf-avtcore-rtp-multi-stream]).
Rationale: If the CB_INTERVAL was always set to the same number of
RTCP reporting intervals, this would cause higher rate RTP sessions
to trigger the RTP circuit breaker after a shorter time interval than
lower rate sessions, because the RTCP reporting interval scales based
on the RTP session bandwidth. This is felt to penalise high rate RTP
sessions too aggressively. Conversely, scaling CB_INTERVAL according
to the inverse of the RTCP reporting interval, so the RTP circuit
breaker triggers after a constant time interval, doesn't sufficiently
protect the network from congestion caused by high-rate flows. The
chosen expression for CB_INTERVAL seeks a balance between these two
extremes. It causes higher rate RTP sessions subject to persistent
congestion to trigger the RTP circuit breaker after a shorter time
interval than do lower rate RTP sessions, while also making the RTP
circuit breaker for such sessions less sensitive by requiring the
congestion to persist for longer numbers of RTCP reporting intervals.
4.6. Ceasing Transmission
What it means to cease transmission depends on the application, but What it means to cease transmission depends on the application, but
the intention is that the application will stop sending RTP data the intention is that the application will stop sending RTP data
packets to a particular destination 3-tuple (transport protocol, packets to a particular destination 3-tuple (transport protocol,
destination port, IP address), until the user makes an explicit destination port, IP address), until the user makes an explicit
attempt to restart the call. It is important that a human user is attempt to restart the call. It is important that a human user is
involved in the decision to try to restart the call, since that user involved in the decision to try to restart the call, since that user
will eventually give up if the calls repeatedly trigger the circuit will eventually give up if the calls repeatedly trigger the circuit
breaker. This will help avoid problems with automatic redial systems breaker. This will help avoid problems with automatic redial systems
from congesting the network. Accordingly, RTP flows halted by the from congesting the network. Accordingly, RTP flows halted by the
skipping to change at page 16, line 50 skipping to change at page 17, line 41
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 8), 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 report is sent in of lost packets reported if the ECN Feedback Report report is sent in
an compound RTCP packet along with an RTCP SR/RR report packet. an compound RTCP packet along with an RTCP SR/RR report packet.
Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback
packets without an RTCP SR/RR packet MUST be ignored. packets 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
skipping to change at page 17, line 28 skipping to change at page 18, line 19
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. behaviour 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 provided non-congestion-related metrics. control purposes, others provided non-congestion-related metrics.
With the exception of RTCP XR ECN Summary Reports (see Section 8), 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 reception 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 RTCP Reporting Groups 7. Impact of Explicit Congestion Notification (ECN)
An optimisation for grouping RTCP reception statistics and other
feedback in RTP sessions with large numbers of participants is given
in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one
SSRC to act as a representative that sends reports on behalf of other
SSRCs that are co-located in the same endpoint and see identical
reception quality. When running the circuit breaker algorithms, an
endpoint MUST treat a reception report from the representative of the
reporting group as if a reception report was received from all
members of that group.
8. Impact of Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the media timeout RTP The use of ECN for RTP flows does not affect the media timeout RTP
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker circuit breaker (Section 4.2) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply (Section 4.1), since these are both connectivity checks that simply
determinate if any packets are being received. determinate if any packets are being received.
ECN-CE marked packets SHOULD be treated as if it were lost for the ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal media purposes of congestion control, when determining the optimal media
sending rate for an RTP flow. If an RTP sender has negotiated ECN sending rate for an RTP flow. If an RTP sender has negotiated ECN
support for an RTP session, and has successfully initiated ECN use on support for an RTP session, and has successfully initiated ECN use on
the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
be treated as if they were lost when calculating if the congestion- be treated as if they were lost when calculating if the congestion-
based RTP circuit breaker (Section 4.3) has been met. The count of based RTP circuit breaker (Section 4.3) has been met. The count of
ECN-CE marked RTP packets is returned in RTCP XR ECN summary report ECN-CE marked RTP packets is returned in RTCP XR ECN summary report
packets if support for ECN has been initiated for an RTP session. packets if support for ECN has been initiated for an RTP session.
9. 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 the sending
rate of the group to reduce by a factor of ten, depending on the RTP rate of the group to reduce by a factor of ten, depending on the RTP
circuit breaker triggered. Grouping flows in this way is expected to circuit breaker triggered. Grouping flows in this way is expected to
be especially useful for layered flows sent using multiple SSRCs, as be especially useful for layered flows sent using multiple SSRCs, as
it allows the layered flow to react as a whole, ceasing transmission it allows the layered flow to react as a whole, ceasing transmission
on the enhancement layers first to reduce sending rate if necessary, on the enhancement layers first to reduce sending rate if necessary,
rather than treating each layer independently. rather than treating each layer independently.
10. 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 disrupting an RTP rates, causing the circuit breaker to trigger and disrupting an RTP
skipping to change at page 19, line 21 skipping to change at page 19, line 43
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 is many minutes duration. This issue is not specific to the circuit
breaker: long RTCP reporting intervals also prevent reception quality breaker: long RTCP reporting intervals also prevent reception quality
reports, feedback messages, codec control messages, etc., from being reports, feedback messages, codec control messages, etc., from being
used. Implementations are expected to impose an upper limit on the used. Implementations are expected to impose an upper limit on the
RTCP reporting interval they are willing to negotiate (based on the RTCP reporting interval they are willing to negotiate (based on the
session bandwidth and RTCP bandwidth fraction) when using the RTP session bandwidth and RTCP bandwidth fraction) when using the RTP
circuit breaker, as discussed in Section 4.5. circuit breaker, as discussed in Section 4.3.
11. IANA Considerations 10. IANA Considerations
There are no actions for IANA. There are no actions for IANA.
12. Acknowledgements 11. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, The authors would like to thank Bernard Aboba, Harald Alvestrand,
Gorry Fairhurst, Nazila Fough, Kevin Gross, Cullen Jennings, Randell Gorry Fairhurst, Nazila Fough, Kevin Gross, Cullen Jennings, Randell
Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Simon
Rescorla, Abheek Saha, Fabio Verdicchio, and Magnus Westerlund for Perreault, Eric Rescorla, Abheek Saha, Fabio Verdicchio, and Magnus
their valuable feedback. Westerlund for their valuable feedback.
13. References 12. References
13.1. Normative References 12.1. 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, March 1997. Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP [RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC Friendly Rate Control (TFRC): Protocol Specification",
3448, January 2003. RFC 3448, DOI 10.17487/RFC3448, January 2003,
<http://www.rfc-editor.org/info/rfc3448>.
[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, July 2003. Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
Protocol Extended Reports (RTCP XR)", RFC 3611, November "RTP Control Protocol Extended Reports (RTCP XR)",
2003. RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
2006. DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
13.2. Informative References 12.2. 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 conference, Applications", Proceedings of the ACM SIGCOMM
2000, DOI 10.1145/347059.347397, August 2000. conference, 2000, DOI 10.1145/347059.347397, August 2000.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work
in progress), February 2015.
[I-D.ietf-avtcore-rtp-multi-stream] [I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session", "Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-07 (work in progress), draft-ietf-avtcore-rtp-multi-stream-09 (work in progress),
March 2015. September 2015.
[I-D.ietf-tsvwg-circuit-breaker] [I-D.ietf-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers", Fairhurst, G., "Network Transport Circuit Breakers",
draft-ietf-tsvwg-circuit-breaker-00 (work in progress), draft-ietf-tsvwg-circuit-breaker-05 (work in progress),
September 2014. October 2015.
[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 Review algorithm", ACM SIGCOMM Computer Communication
27(3), DOI 10.1145/263932.264023, July 1997. Review 27(3), 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 conference, Validation", Proceedings of the ACM SIGCOMM
1998, DOI 10.1145/285237.285291, August 1998. conference, 1998, DOI 10.1145/285237.285291, August 1998.
[RFC2862] Civanlar, M. and G. Cash, "RTP Payload Format for Real-
Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000,
<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", RFC of Explicit Congestion Notification (ECN) to IP",
3168, September 2001. RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<http://www.rfc-editor.org/info/rfc4103>.
[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, February 2008. with Feedback (AVPF)", RFC 5104, DOI 10.17487/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, February 2008. (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines [RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November for Application Designers", BCP 145, RFC 5405,
2008. 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, March 2009. RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009,
<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, April 2009. and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
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
Control", RFC 5681, September 2009. Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<http://www.rfc-editor.org/info/rfc5681>.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[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, August 2012. for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>.
[RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended [RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
Report (XR) Block for Packet Delay Variation Metric Report (XR) Block for Packet Delay Variation Metric
Reporting", RFC 6798, November 2012. Reporting", RFC 6798, DOI 10.17487/RFC6798, November 2012,
<http://www.rfc-editor.org/info/rfc6798>.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol [RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric (RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, January 2013. Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013,
<http://www.rfc-editor.org/info/rfc6843>.
[RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, "RTP Control [RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Control Protocol (RTCP) Extended Report (XR) Block for
Loss Metric Reporting", RFC 6958, May 2013. Burst/Gap Loss Metric Reporting", RFC 6958,
DOI 10.17487/RFC6958, May 2013,
<http://www.rfc-editor.org/info/rfc6958>.
[RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol [RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count Metric (RTCP) Extended Report (XR) Block for Discard Count Metric
Reporting", RFC 7002, September 2013. Reporting", RFC 7002, DOI 10.17487/RFC7002, September
2013, <http://www.rfc-editor.org/info/rfc7002>.
[RFC7003] Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol [RFC7003] Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control
(RTCP) Extended Report (XR) Block for Burst/Gap Discard Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Metric Reporting", RFC 7003, September 2013. Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003,
September 2013, <http://www.rfc-editor.org/info/rfc7003>.
[RFC7097] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol [RFC7097] Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control
(RTCP) Extended Report (XR) for RLE of Discarded Packets", Protocol (RTCP) Extended Report (XR) for RLE of Discarded
RFC 7097, January 2014. Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014,
<http://www.rfc-editor.org/info/rfc7097>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, April 2014. Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[Sarker] Sarker, Z., Singh, V., and C.S. 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, 2014,
April 2014. April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C.S. 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 Workshop, Proceedings of the International Packet Video
2013, DOI 10.1109/PV.2013.6691439, December 2013. Workshop, 2013, DOI 10.1109/PV.2013.6691439, December
2013.
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
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