draft-ietf-avtcore-rtp-circuit-breakers-04.txt   draft-ietf-avtcore-rtp-circuit-breakers-05.txt 
AVTCORE Working Group C. S. Perkins AVTCORE Working Group C. S. 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: July 17, 2014 January 13, 2014 Expires: August 18, 2014 February 14, 2014
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-04 draft-ietf-avtcore-rtp-circuit-breakers-05
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
skipping to change at page 1, line 44 skipping to change at page 1, line 44
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 July 17, 2014. This Internet-Draft will expire on August 18, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Provisions Relating to IETF Documents Provisions Relating to IETF Documents
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publication of this document. Please review these documents publication of this document. Please review these documents
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to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 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 . 5 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6
4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7 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. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 12
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12 4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 13
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 13
7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 13 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 14
8. Impact of Explicit Congestion Notification (ECN) . . . . . . 14 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 15
9. Security Considerations . . . . . . . . . . . . . . . . . . . 14 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 15
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 9. Security Considerations . . . . . . . . . . . . . . . . . . . 15
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16
12.1. Normative References . . . . . . . . . . . . . . . . . . 15 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 16
12.2. Informative References . . . . . . . . . . . . . . . . . 15 12.1. Normative References . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17 12.2. Informative References . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 18
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
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persists, it is a sign of a more ongoing and significant problem. persists, it is a sign of a more ongoing and significant problem.
Accordingly, if a sender of RTP data packets receives two or more Accordingly, if a sender of RTP data packets receives two or more
consecutive RTCP SR or RR packets from the same receiver, and those consecutive RTCP SR or RR packets from the same receiver, and those
packets correspond to its transmission and have a non-increasing packets correspond to its transmission and have a non-increasing
extended highest sequence number received field (i.e., the sender extended highest sequence number received field (i.e., the sender
receivers at least three RTCP SR or RR packets that report the same receivers at least three RTCP SR or RR packets that report the same
value in the extended highest sequence number received field for an value in the extended highest sequence number received field for an
SSRC, but the sender has sent RTP data packets for that SSRC that SSRC, but the sender has sent RTP data packets for that SSRC that
would have caused an increase in the reported value of the extended would have caused an increase in the reported value of the extended
highest sequence number received if they had reached the receiver), highest sequence number received if they had reached the receiver),
then that sender SHOULD cease transmission (see Section 4.4). then that sender SHOULD cease transmission (see Section 4.5).
The reason for waiting for two or more consecutive RTCP packets with The reason for waiting for two or more consecutive RTCP packets with
a non-increasing extended highest sequence number is to give enough a non-increasing extended highest sequence number is to give enough
time for transient reception problems to resolve themselves, but to time for transient reception problems to resolve themselves, but to
stop problem flows quickly enough to avoid causing serious ongoing stop problem flows quickly enough to avoid causing serious ongoing
network congestion. A single RTCP report showing no reception could network congestion. A single RTCP report showing no reception could
be caused by a transient fault, and so will not cease transmission. be caused by a transient fault, and so will not cease transmission.
Waiting for more than two consecutive RTCP reports before stopping a Waiting for more than two consecutive RTCP reports before stopping a
flow might avoid some false positives, but could lead to problematic flow might avoid some false positives, but could lead to problematic
flows running for a long time period (potentially tens of seconds, flows running for a long time period (potentially tens of seconds,
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In addition to media timeouts, as were discussed in Section 4.1, an In addition to media timeouts, as were discussed in Section 4.1, an
RTP session has the possibility of an RTCP timeout. This can occur RTP session has the possibility of an RTCP timeout. This can occur
when RTP data packets are being sent, but there are no RTCP reports 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 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 receiver to send RTCP reports, or a failure of the return path that
is preventing those RTCP reporting from being delivered. In either is preventing those RTCP reporting from being delivered. In either
case, it is not safe to continue transmission, since the sender has case, it is not safe to continue transmission, since the sender has
no way of knowing if it is causing congestion. Accordingly, an RTP no way of knowing if it is causing congestion. Accordingly, an RTP
sender that has not received any RTCP SR or RTCP RR packets reporting sender that has not received any RTCP SR or RTCP RR packets reporting
on the SSRC it is using for three or more RTCP reporting intervals on the SSRC it is using for three or more RTCP reporting intervals
SHOULD cease transmission (see Section 4.4). When calculating the SHOULD cease transmission (see Section 4.5). When calculating the
timeout, the fixed minimum RTCP reporting interval SHOULD be used timeout, the fixed minimum RTCP reporting interval SHOULD be used
(based on the rationale in Section 6.2 of RFC 3550 [RFC3550]). (based on the rationale in Section 6.2 of RFC 3550 [RFC3550]).
The choice of three RTCP reporting intervals as the timeout is made The choice of three RTCP reporting intervals as the timeout is made
following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that
participants in an RTP session will timeout and remove an RTP sender participants in an RTP session will timeout and remove an RTP sender
from the list of active RTP senders if no RTP data packets have been from the list of active RTP senders if no RTP data packets have been
received from that RTP sender within the last two RTCP reporting received from that RTP sender within the last two RTCP reporting
intervals. Using a timeout of three RTCP reporting intervals is intervals. Using a timeout of three RTCP reporting intervals is
therefore large enough that the other participants will have timed therefore large enough that the other participants will have timed
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throughput. The time scale on which TCP throughput is measured is throughput. The time scale on which TCP throughput is measured is
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, Padhye et al. [Padhye] showed For a long-lived TCP Reno connection, it has been shown that the TCP
that the throughput can be estimated using the following equation: throughput can be estimated using the following equation [Padhye]:
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.
p is the loss event rate, between 0 and 1.0, of the number of loss 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. events as a fraction of the number of packets transmitted.
t_RTO is the TCP retransmission timeout value in seconds, t_RTO is the TCP retransmission timeout value in seconds, generally
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 that are 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 [Mathis]:
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. Measurements have shown that the
simplified TCP throughput equation is effective as an RTP circuit
breaker for multimedia flows sent to hosts on residential networks
using ADSL and cable modem links [Singh]. The data shows that the
full TCP throughput equation tends to be more sensitive to packet
loss and triggers the RTP circuit breaker earlier than the simplified
equation. Implementations that desire this extra sensitivity MAY use
the full TCP throughput equation in the RTP circuit breaker. Initial
measurements in LTE networks have shown that the extra sensitivity is
helpful in that environment, with the full TCP throughput equation
giving a more balanced circuit breaker response than the simplified
TCP equation [Sarker]; other networks might see similar behaviour.
Given this TCP equation, two parameters need to be estimated and No matter what TCP throughput equation is chosen, two parameters need
reported to the sender in order to calculate the throughput: the to be estimated and reported to the sender in order to calculate the
round trip time, R, and the loss event rate, p (the packet size, s, throughput: the round trip time, R, and the loss event rate, p (the
is known to the sender). The round trip time can be estimated from packet size, s, is known to the sender). The round trip time can be
RTCP SR and RR packets. This is done too infrequently for accurate estimated from RTCP SR and RR packets. This is done too infrequently
statistics, but is the best that can be done with the standard RTCP for accurate statistics, but is the best that can be done with the
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, not caring about when the losses occurred). fraction of packets lost, not caring about when the losses occurred).
Using the loss fraction in place of the loss event rate can Using the loss fraction in place of the loss event rate can
overestimate the loss. We believe that this overestimate will not be overestimate the loss. We believe that this overestimate will not be
significant, given that we are only interested in order of magnitude significant, given that we are only interested in order of magnitude
comparison ([Floyd] section 3.2.1 shows that the difference is small comparison ([Floyd] section 3.2.1 shows that the difference is small
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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). Systems that usually SHOULD cease transmission (see Section 4.5). Systems that usually
send at a high data rate, but that can reduce their data rate send at a high data rate, but that can reduce their data rate
significantly (i.e., by at least a factor of ten), MAY first reduce significantly (i.e., by at least a factor of ten), MAY first reduce
their sending rate to this lower value to see if this resolves the their sending rate to this lower value to see if this resolves the
congestion, but MUST then cease transmission if the problem does not congestion, but MUST then cease transmission if the problem does not
resolve itself within a further two RTCP reporting intervals (see resolve itself within a further two RTCP reporting intervals (see
Section 4.4). An example of this might be a video conferencing Section 4.5). An example of this might be a video conferencing
system that backs off to sending audio only, before completely system that backs off to sending audio only, before completely
dropping the call. If such a reduction in sending rate resolves the dropping the call. If such a reduction in sending rate resolves the
congestion problem, the sender MAY gradually increase the rate at congestion problem, the sender MAY gradually increase the rate at
which it sends data after a reasonable amount of time has passed, which it sends data after a reasonable amount of time has passed,
provided it takes care not to cause the problem to recur 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 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 RTCP reporting interval (as specified in Section 6.2 of RFC 3550
[RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP [RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP
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If there are more media streams that can be reported in a single RTCP If there are more media streams that can be reported in a single RTCP
SR or RR packet, or if the size of a complete RTCP SR or RR packet SR or RR packet, or if the size of a complete RTCP SR or RR packet
exceeds the network MTU, then the receiver will report on a subset of exceeds the network MTU, then the receiver will report on a subset of
sources in each reporting interval, with the subsets selected round- sources in each reporting interval, with the subsets selected round-
robin across multiple intervals so that all sources are eventually robin across multiple intervals so that all sources are eventually
reported [RFC3550]. When generating such round-robin RTCP reports, reported [RFC3550]. When generating such round-robin RTCP reports,
priority SHOULD be given to reports on sources that have high packet priority SHOULD be given to reports on sources that have high packet
loss rates, to ensure that senders are aware of network congestion loss rates, to ensure that senders are aware of network congestion
they are causing (this is an update to [RFC3550]). they are causing (this is an update to [RFC3550]).
4.4. Ceasing Transmission 4.4. RTP/AVP Circuit Breaker #4: Media Usability
Applications that use RTP are generally tolerant to some amount of
packet loss. How much packet loss can be tolerated will depend on
the application, media codec, and the amount of error correction and
packet loss concealment that is applied. There is an upper bound on
the amount of loss can be corrected, however, beyond which the media
becomes unusable. Similarly, many applications have some upper bound
on the media capture to play-out latency that can be tolerated before
the application becomes unusable. The latency bound will depend on
the application, but typical values can range from the order of a few
hundred milliseconds for voice telephony and interactive conferencing
applications, up to several seconds for some video-on-demand systems.
As a final circuit breaker, applications SHOULD monitor the reported
packet loss and delay to estimate whether the media is suitable for
the intended purpose. If the packet loss rate and/or latency is such
that the media has become unusable for the application, and has
remained unusable for a significant time period, then the application
SHOULD cease transmission. This memo intentionally does not define a
bound on the packet loss rate or latency that will result in unusable
media, nor does it specify what time period is deemed significant, as
these are highly application dependent.
Sending media that suffers from such high packet loss or latency that
it is unusable at the receiver is both wasteful of resources, and of
no benefit to the user of the application. It also is highly likely
to be congesting the network, and disrupting other applications. As
such, the congestion circuit breaker will almost certainly trigger to
stop flows where the media would be unusable due to high packet loss
or latency. However, in pathological scenarios where the congestion
circuit breaker does not stop the flow, it is desirable that the RTP
application cease sending useless traffic. The role of the media
usability circuit breaker is to protect the network in such cases.
4.5. 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
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[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, July
2006. 2006.
12.2. 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", Proceedings of the ACM SIGCOMM conference,
347059.347397, August 2000. 2000, DOI 10.1145/347059.347397, August 2000.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
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:
Grouping RTCP Reception Statistics and Other Feedback", Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-01 (work
in progress), July 2013. in progress), January 2014.
[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
macroscopic behavior of the TCP congestion avoidance
algorithm", ACM SIGCOMM Computer Communication 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", Proc. ACM SIGCOMM 1998, DOI 10.1145/ Validation", Proceedings of the ACM SIGCOMM conference,
285237.285291, August 1998. 1998, DOI 10.1145/285237.285291, 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", RFC of Explicit Congestion Notification (ECN) to IP", 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,
skipping to change at page 17, line 17 skipping to change at page 18, line 25
Reporting", RFC 7002, September 2013. Reporting", RFC 7002, September 2013.
[RFC7003] Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol [RFC7003] Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Burst/Gap Discard (RTCP) Extended Report (XR) Block for Burst/Gap Discard
Metric Reporting", RFC 7003, September 2013. Metric Reporting", RFC 7003, September 2013.
[RFC7097] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol [RFC7097] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
(RTCP) Extended Report (XR) for RLE of Discarded Packets", (RTCP) Extended Report (XR) for RLE of Discarded Packets",
RFC 7097, January 2014. RFC 7097, January 2014.
[Sarker] Sarker, Z., Singh, V., and C.S. Perkins, "An Evaluation of
RTP Circuit Breaker Performance on LTE Networks",
Proceedings of the IEEE Infocom workshop on Communication
and Networking Techniques for Contemporary Video, 2014,
April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C.S. Perkins,
"Circuit Breakers for Multimedia Congestion Control",
Proceedings of the International Packet Video 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
Varun Singh Varun Singh
Aalto University Aalto University
School of Electrical Engineering School of Electrical Engineering
Otakaari 5 A Otakaari 5 A
Espoo, FIN 02150 Espoo, FIN 02150
Finland Finland
Email: varun@comnet.tkk.fi Email: varun@comnet.tkk.fi
URI: http://www.netlab.tkk.fi/~varun/ URI: http://www.netlab.tkk.fi/~varun/
 End of changes. 22 change blocks. 
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