draft-ietf-avtcore-rtp-circuit-breakers-02.txt   draft-ietf-avtcore-rtp-circuit-breakers-03.txt 
Network Working Group C. Perkins AVTCORE Working Group C. S. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track V. Singh Updates: 3550 (if approved) V. Singh
Expires: August 26, 2013 Aalto University Intended status: Standards Track Aalto University
February 22, 2013 Expires: January 16, 2014 July 15, 2013
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-02 draft-ietf-avtcore-rtp-circuit-breakers-03
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
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
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 August 26, 2013. This Internet-Draft will expire on January 16, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the Copyright (c) 2013 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 . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . . 6 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6
4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 8 4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . . 8 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . . 9 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 9
4.4. Ceasing Transmission . . . . . . . . . . . . . . . . . . . 12 4.4. Ceasing Transmission . . . . . . . . . . . . . . . . . . 12
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13
7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 14 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 14
8. Security Considerations . . . . . . . . . . . . . . . . . . . 14 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 14
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 9. Security Considerations . . . . . . . . . . . . . . . . . . . 14
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15
11.1. Normative References . . . . . . . . . . . . . . . . . . . 15 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
11.2. Informative References . . . . . . . . . . . . . . . . . . 15 12.1. Normative References . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17 12.2. Informative References . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
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
skipping to change at page 9, line 23 skipping to change at page 8, line 50
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
that the corresponding RTCP SR or RR packets are too large to fit
into the network MTU, this will force the receiver to generate RTCP
SR or RR packets in a round-robin manner. In this case, the sender
MAY treat receipt of an RTCP SR or RR packet corresponding to an SSRC
it sent using the same 5-tuple of source and destination IP address,
port, and protocol, as an indication that the receiver and return
path are working to prevent the RTCP timeout circuit breaker from
triggering.
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 RR If RTP data packets are being sent, and the corresponding RTCP SR or
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 RTP flow is on a reasonable time scale, that is not less than the RTP flow is
skipping to change at page 10, line 11 skipping to change at page 9, line 47
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, Padhye et al. [Padhye] showed For a long-lived TCP Reno connection, Padhye et al. [Padhye] showed
that the throughput can be estimated using the following equation: that the throughput can be estimated using the following equation:
s s
X = -------------------------------------------------------------- X = --------------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2))) R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))
where: where:
X is the transmit rate in bytes/second. X is the transmit rate in bytes/second.
s is the packet size in bytes. If data packets vary in size, then s is the packet size in bytes. If data packets vary in size, then
the average size is to be used. the average size is to be used.
R is the round trip time in seconds. R is the round trip time in seconds.
skipping to change at page 10, line 38 skipping to change at page 10, line 28
approximated by setting t_RTO = 4*R. approximated by setting t_RTO = 4*R.
b is the number of packets acknowledged by a single TCP b is the number of packets acknowledged by a single TCP
acknowledgement ([RFC3448] recommends the use of b=1 since many acknowledgement ([RFC3448] recommends the use of b=1 since many
TCP implementations do not use delayed acknowledgements). TCP implementations do not use delayed acknowledgements).
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, this formula can be [RFC3448]. Under conditions of low packet loss, this formula can be
approximated as follows with reasonable accuracy: approximated as follows with reasonable accuracy:
s s
X = --------------- X = ---------------
R * sqrt(p*2/3) R * sqrt(p*2/3)
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. calculate than the full equation.
Given this TCP equation, two parameters need to be estimated and Given this TCP equation, two parameters need to be estimated and
reported to the sender in order to calculate the throughput: the reported to the sender in order to calculate the throughput: the
round trip time, R, and the loss event rate, p (the packet size, s, round trip time, R, and the loss event rate, p (the packet size, s,
is known to the sender). The round trip time can be estimated from is known to the sender). The round trip time can be estimated from
RTCP SR and RR packets. This is done too infrequently for accurate RTCP SR and RR packets. This is done too infrequently for accurate
statistics, but is the best that can be done with the standard RTCP statistics, but is the best that can be done with the standard RTCP
mechanisms. mechanisms.
RTCP RR packets contain the packet loss fraction, rather than the Report blocks in RTCP SR or RR packets contain the packet loss
loss event rate, so p cannot be reported (TCP typically treats the fraction, rather than the loss event rate, so p cannot be reported
loss of multiple packets within a single RTT as one loss event, but (TCP typically treats the loss of multiple packets within a single
RTCP RR packets report the overall fraction of packets lost, not RTT as one loss event, but RTCP RR packets report the overall
caring about when the losses occurred). Using the loss fraction in fraction of packets lost, not caring about when the losses occurred).
place of the loss event rate can overestimate the loss. We believe Using the loss fraction in place of the loss event rate can
that this overestimate will not be significant, given that we are overestimate the loss. We believe that this overestimate will not be
only interested in order of magnitude comparison ([Floyd] section significant, given that we are only interested in order of magnitude
3.2.1 shows that the difference is small for steady-state conditions comparison ([Floyd] section 3.2.1 shows that the difference is small
and random loss, but using the loss fraction is more conservative in for steady-state conditions and random loss, but using the loss
the case of bursty loss). fraction is more conservative in the case of bursty loss).
The congestion circuit breaker is therefore: when a sender receives The congestion circuit breaker is therefore: when a sender receives
an RTCP SR or RR packet that contains a report block for an SSRC it an RTCP SR or RR packet that contains a report block for an SSRC it
is using, that sender has to check the fraction lost field in that is using, that sender has to check the fraction lost field in that
report block to determine if there is a non-zero packet loss rate. report block to determine if there is a non-zero packet loss rate.
If the fraction lost field is zero, then continue sending as normal. If the fraction lost field is zero, then continue sending as normal.
If the fraction lost is greater than zero, then estimate the TCP If the fraction lost is greater than zero, then estimate the TCP
throughput using the simplified equation above, and the measured R, p throughput using the simplified equation above, and the measured R, p
(approximated by the fraction lost), and s. Compare this with the (approximated by the fraction lost), and s. Compare this with the
actual sending rate. If the actual sending rate is more than ten actual sending rate. If the actual sending rate is more than ten
times the estimated sending rate derived from the TCP throughput times the estimated sending rate derived from the TCP throughput
equation for two consecutive RTCP reporting intervals, the sender equation for two consecutive RTCP reporting intervals, the sender
SHOULD cease transmission (see Section 4.4). If the RTP sender is SHOULD cease transmission (see Section 4.4). Systems that usually
using a reduced minimum RTCP reporting interval (as specified in send at a high data rate, but that can reduce their data rate
Section 6.2 of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), significantly (i.e., by at least a factor of ten), MAY first reduce
then that reduced RTCP reporting interval is used when determining if their sending rate to this lower value to see if this resolves the
the circuit breaker is triggered, since that interval scales with the congestion, but MUST then cease transmission if the problem does not
media data rate. resolve itself within a further two RTCP reporting intervals (see
Section 4.4). An example of this might be a video conferencing
Systems that usually send at a high data rate, but that can reduce system that backs off to sending audio only, before completely
their data rate significantly (i.e., by at least a factor of ten), dropping the call. If such a reduction in sending rate resolves the
MAY first reduce their sending rate to this lower value to see if congestion problem, the sender MAY gradually increase the rate at
this resolves the congestion, but MUST then cease transmission if the which it sends data after a reasonable amount of time has passed,
problem does not resolve itself within a further two RTCP reporting provided it takes care not to cause the problem to recur
intervals (see Section 4.4). An example of this might be a video
conferencing system that backs off to sending audio only, before
completely dropping the call. If such a reduction in sending rate
resolves the congestion problem, the sender MAY gradually increase
the rate at which it sends data after a reasonable amount of time has
passed, provided it takes care not to cause the problem to recur
("reasonable" is intentionally not defined here). ("reasonable" is intentionally not defined here).
If the incoming RTCP SR or RR packets are using a reduced minimum
RTCP reporting interval (as specified in Section 6.2 of RFC 3550
[RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP
reporting interval is used when determining if the circuit breaker is
triggered. The RTCP reporting interval of the media sender does not
affect how quickly congestion circuit breaker can trigger. The
timing is based on the RTCP reporting interval of the receiver that
matters (note that RTCP requires all participants in a session to
have similar reporting intervals, else the participant timeout rules
in [RFC3550] will not work).
As in Section 4.1, we use two reporting intervals to avoid triggering As in Section 4.1, we use two reporting intervals to avoid triggering
the circuit breaker on transient failures. This circuit breaker is a the circuit breaker on transient failures. This circuit breaker is a
worst-case condition, and congestion control needs to be performed to worst-case condition, and congestion control needs to be performed to
keep well within this bound. It is expected that the circuit breaker keep well within this bound. It is expected that the circuit breaker
will only be triggered if the usual congestion control fails for some will only be triggered if the usual congestion control fails for some
reason. reason.
If there are more media streams that can be reported in a single RTCP
SR or RR packet, or if the size of a complete RTCP SR or RR packet
exceeds the network MTU, then the receiver will report on a subset of
sources in each reporting interval, with the subsets selected round-
robin across multiple intervals so that all sources are eventually
reported [RFC3550]. When generating such round-robin RTCP reports,
priority SHOULD be given to reports on sources that have high packet
loss rates, to ensure that senders are aware of network congestion
they are causing (this is an update to [RFC3550]).
4.4. Ceasing Transmission 4.4. 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
skipping to change at page 13, line 19 skipping to change at page 13, line 26
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 RTP circuit breaker (they do not contain the information used by the RTP circuit breaker (they do not contain the information used by
the circuit breaker algorithm). Reduced-size RTCP reports sent under the circuit breaker algorithm). Reduced-size RTCP reports sent under
the RTP/AVPF early feedback rules that contain RTCP SR or RR packets the RTP/AVPF early feedback rules that contain RTCP SR or RR packets
MUST be processed as if they were sent as regular RTCP reports, and MUST be processed as if they were sent as regular RTCP reports, and
counted towards the circuit breaker conditions specified in Section 4 counted towards the circuit breaker conditions specified in Section 4
of this memo. This will potentially make the RTP circuit breaker of this memo. This will potentially make the RTP circuit breaker
fire earlier than it would if the RTP/AVPF profile was not used. fire earlier than it would if the RTP/AVPF profile was not used.
When using ECN with RTP (see Section 7), early RTCP feedback packets When using ECN with RTP (see Section 8), 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 early RTP/ These rules are intended to allow the use of low-overhead early RTP/
AVPF feedback for generic NACK messages without triggering the RTP AVPF feedback for generic NACK messages without triggering the RTP
circuit breaker. This is expected to make such feedback suitable for circuit breaker. This is expected to make such feedback suitable for
skipping to change at page 13, line 41 skipping to change at page 13, line 48
events in between regular RTCP reports. The reaction to reduced-size events in between regular RTCP reports. The reaction to reduced-size
RTCP SR/RR packets is to allow such algorithms to send feedback that RTCP SR/RR packets is to allow such algorithms to send feedback that
can trigger the circuit breaker, when desired. can trigger the circuit breaker, when desired.
6. Impact of RTCP XR 6. Impact of RTCP 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 7), With the exception of RTCP XR ECN Summary Reports (see Section 8),
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 Explicit Congestion Notification (ECN) 7. Impact of RTCP Reporting Groups
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 4.1) or the RTCP 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.
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.
8. 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
session. This is somewhat more difficult for an off-path attacker, session. This is somewhat more difficult for an off-path attacker,
due to the need to guess the randomly chosen RTP SSRC value and the due to the need to guess the randomly chosen RTP SSRC value and the
RTP sequence number. This attack can be avoided if RTCP packets are RTP sequence number. This attack can be avoided if RTCP packets are
authenticated, for example using the Secure RTP profile [RFC3711]. authenticated, for example using the Secure RTP profile [RFC3711].
9. IANA Considerations 10. IANA Considerations
There are no actions for IANA. There are no actions for IANA.
10. Acknowledgements 11. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, The authors would like to thank Bernard Aboba, Harald Alvestrand,
Kevin Gross, Cullen Jennings, Randell Jesup, Jonathan Lennox, Matt Kevin Gross, Cullen Jennings, Randell Jesup, Jonathan Lennox, Matt
Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their
valuable feedback. valuable feedback.
11. References 12. References
11.1. 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, March 1997.
[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", Friendly Rate Control (TFRC): Protocol Specification", RFC
RFC 3448, January 2003. 3448, January 2003.
[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, July 2003.
[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. July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, Protocol Extended Reports (RTCP XR)", RFC 3611, November
November 2003. 2003.
[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, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
July 2006. 2006.
11.2. 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", Proc. ACM SIGCOMM 2000, DOI 10.1145/ Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/
347059.347397, August 2000. 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-00 (work
in progress), July 2013.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard] [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]
Clark, A., Huang, R., and W. Wu, "RTP Control Clark, A., Huang, R., and W. Wu, "RTP Control
Protocol(RTCP) Extended Report (XR) Block for Burst/Gap Protocol(RTCP) Extended Report (XR) Block for Burst/Gap
Discard metric Reporting", Discard metric Reporting", draft-ietf-xrblock-rtcp-xr-
draft-ietf-xrblock-rtcp-xr-burst-gap-discard-10 (work in burst-gap-discard-14 (work in progress), April 2013.
progress), January 2013.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]
Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Loss metric Reporting", Loss metric Reporting", draft-ietf-xrblock-rtcp-xr-burst-
draft-ietf-xrblock-rtcp-xr-burst-gap-loss-08 (work in gap-loss-12 (work in progress), April 2013.
progress), January 2013.
[I-D.ietf-xrblock-rtcp-xr-discard]
Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count metric
Reporting", draft-ietf-xrblock-rtcp-xr-discard-11 (work in
progress), December 2012.
[I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics]
Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
(RTCP) Extended Reports (XR) for Run Length Encoding (RLE) (RTCP) Extended Reports (XR) for Run Length Encoding (RLE)
of Discarded Packets", of Discarded Packets", draft-ietf-xrblock-rtcp-xr-discard-
draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-05 (work in rle-metrics-06 (work in progress), July 2013.
progress), December 2012.
[I-D.ietf-xrblock-rtcp-xr-discard]
Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count metric
Reporting", draft-ietf-xrblock-rtcp-xr-discard-15 (work in
progress), June 2013.
[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", Proc. ACM SIGCOMM 1998, DOI 10.1145/ Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/
285237.285291, August 1998. 285237.285291, August 1998.
[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
RFC 3168, September 2001. 3168, September 2001.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, March 2004.
[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, February 2008.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
 End of changes. 31 change blocks. 
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