draft-ietf-avtcore-rtp-circuit-breakers-06.txt   draft-ietf-avtcore-rtp-circuit-breakers-07.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: January 05, 2015 July 04, 2014 Expires: April 30, 2015 October 27, 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-06 draft-ietf-avtcore-rtp-circuit-breakers-07
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
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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-
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 January 05, 2015. This Internet-Draft will expire on April 30, 2015.
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.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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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 . 6 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 . . . . . . . . 8
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. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 12 4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 13
4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 13 4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 14
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 13 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 14
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 14 6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 15
7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 15 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 15
8. Impact of Explicit Congestion Notification (ECN) . . . . . . 15 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 16
9. Security Considerations . . . . . . . . . . . . . . . . . . . 15 9. Impact of Layered Coding . . . . . . . . . . . . . . . . . . 16
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16 10. Security Considerations . . . . . . . . . . . . . . . . . . . 17
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 17
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 12. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 17
12.1. Normative References . . . . . . . . . . . . . . . . . . 16 13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 18
12.2. Informative References . . . . . . . . . . . . . . . . . 17 14. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 18 14.1. Normative References . . . . . . . . . . . . . . . . . . 18
14.2. Informative References . . . . . . . . . . . . . . . . . 18
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
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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the cumulative number of packets lost, the highest sequence number the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are network round trip time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by sent periodically, with the reporting interval being determined by
the number of SSRCs used in the session and a configured session the number of SSRCs used in the session and a configured session
bandwidth estimate (the number of SSRCs used is usually two in a bandwidth estimate (the number of SSRCs used is usually two in a
unicast session, one for each participant, but can be greater if unicast session, one for each participant, but can be greater if
the participants send multiple media streams). The interval the participants send multiple media streams). The interval
between reports sent from each receiver tends to be on the order between reports sent from each receiver tends to be on the order
of a few seconds on average, and it is randomised to avoid of a few seconds on average, although it varies with the session
synchronisation of reports from multiple receivers. RTCP RR bandwidth, and sub-second reporting intervals are possible in high
packets allow a receiver to report ongoing network congestion to bandwidth sessions, and it is randomised to avoid synchronisation
the sender. However, if a receiver detects the onset of of reports from multiple receivers. RTCP RR packets allow a
congestion partway through a reporting interval, the base RTP receiver to report ongoing network congestion to the sender.
specification contains no provision for sending the RTCP RR packet However, if a receiver detects the onset of congestion part way
early, and the receiver has to wait until the next scheduled through a reporting interval, the base RTP specification contains
reporting interval. no provision for sending the RTCP RR packet early, and the
receiver has to wait until the next scheduled reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not complex and sophisticated reception quality metrics, but do not
change the RTCP timing rules. RTCP extended reports of potential change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are the extended packet interest for congestion control purposes are the extended packet
loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097], loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097],
[RFC7003], [RFC6958]; and the extended delay metrics [RFC6843], [RFC7003], [RFC6958]; and the extended delay metrics [RFC6843],
[RFC6798]. Other RTCP Extended Reports that could be helpful for [RFC6798]. Other RTCP Extended Reports that could be helpful for
congestion control purposes might be developed in future. congestion control purposes might be developed in future.
o Rapid feedback about the occurrence of congestion events can be o Rapid feedback about the occurrence of congestion events can be
achieved using the Extended RTP Profile for RTCP-Based Feedback achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] in place of the more common RTP/AVP profile (RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124])
[RFC3551]. This modifies the RTCP timing rules to allow RTCP in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP
reports to be sent early, in some cases immediately, provided the timing rules to allow RTCP reports to be sent early, in some cases
RTCP transmission keeps within its bandwidth allocation. It also immediately, provided the RTCP transmission rate keeps within its
defines new transport-layer feedback messages, including negative bandwidth allocation. It also defines transport-layer feedback
acknowledgements (NACKs), that can be used to report on specific messages, including negative acknowledgements (NACKs), that can be
congestion events. The use of the RTP/AVPF profile is dependent used to report on specific congestion events. RTP Codec Control
on signalling. The RTP Codec Control Messages [RFC5104] extend Messages [RFC5104] extend the RTP/AVPF profile with additional
the RTP/AVPF profile with additional feedback messages that can be feedback messages that can be used to influence that way in which
used to influence that way in which rate adaptation occurs. The rate adaptation occurs, but do not further change the dynamics of
dynamics of how rapidly feedback can be sent are unchanged. how rapidly feedback can be sent. Use of the RTP/AVPF profile is
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. There are two methods: [RFC5450] represents the
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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
which claim to support RTP, but that do not implement RTCP, cannot that claim to support RTP, but that do not implement RTCP, cannot use
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.
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
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3. Receivers report the fraction of RTP data packets lost during the 3. Receivers report the fraction of RTP data packets lost during the
RTCP reporting interval, and the cumulative number of RTP packets RTCP reporting interval, and the cumulative number of RTP packets
lost over the entire RTP session. lost over the entire RTP session.
These congestion signals limit the possible circuit breakers, since These congestion signals limit the possible circuit breakers, since
they give only limited visibility into the behaviour of the network. they give only limited visibility into the 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 far 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 though, and don't give enough information to
distinguish a delay change due to routing updates from queuing delay distinguish a delay change due to routing updates from queuing delay
caused by congestion. Accordingly, we cannot use the RTT estimate caused by congestion. Accordingly, we cannot use the RTT estimate
alone as an RTP circuit breaker. 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
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4.1. RTP/AVP Circuit Breaker #1: Media Timeout 4.1. RTP/AVP Circuit Breaker #1: Media Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets If RTP data packets are being sent, but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term packets are not reaching the receiver. This could be a short-term
issue affecting only a few packets, perhaps caused by a slow-to-open issue affecting only a few packets, perhaps caused by a slow-to-open
firewall or a transient connectivity problem, but if the issue firewall or a transient connectivity problem, but if the issue
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 three 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, then that sender extended highest sequence number received field, then that sender
SHOULD cease transmission (see Section 4.5). The extended highest SHOULD cease transmission (see Section 4.5). The extended highest
sequence number received field is non-increasing if the sender sequence number received field is non-increasing if the sender
receives at least three RTCP SR or RR packets that report the same receives at least three consecutive RTCP SR or RR packets that report
value for this field, but it has sent RTP data packets that would the same value for this field, but it has sent RTP data packets that
have caused an increase in the reported value if they had reached the would have caused an increase in the reported value if they had
receiver. reached the receiver.
The reason for waiting for two or more consecutive RTCP packets with The reason for waiting for three or more consecutive RTCP packets
a non-increasing extended highest sequence number is to give enough with a non-increasing extended highest sequence number is to give
time for transient reception problems to resolve themselves, but to enough time for transient reception problems to resolve themselves,
stop problem flows quickly enough to avoid causing serious ongoing but to stop problem flows quickly enough to avoid causing serious
network congestion. A single RTCP report showing no reception could ongoing network congestion. A single RTCP report showing no
be caused by a transient fault, and so will not cease transmission. reception could be caused by a transient fault, and so will not cease
Waiting for more than two consecutive RTCP reports before stopping a transmission. Waiting for more than three consecutive RTCP reports
flow might avoid some false positives, but could lead to problematic before stopping a flow might avoid some false positives, but could
flows running for a long time period (potentially tens of seconds, lead to problematic flows running for a long time period (potentially
depending on the RTCP reporting interval) before being cut off. tens of seconds, depending on the RTCP reporting interval) before
Equally, an application that sends few packets when the packet loss being cut off. Equally, an application that sends few packets when
rate is high runs the risk that the media timeout circuit breaker the packet loss rate is high runs the risk that the media timeout
triggers inadvertently. The chosen timeout interval is a trade-off circuit breaker triggers inadvertently. The chosen timeout interval
between these extremes. is a trade-off between these extremes.
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout
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
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from the list of active RTP senders if no RTP data packets have been from the list of active RTP senders if no RTP data packets have been
received from that RTP sender within the last two RTCP reporting received from that RTP sender within the last two RTCP reporting
intervals. Using a timeout of three RTCP reporting intervals is intervals. Using a timeout of three RTCP reporting intervals is
therefore large enough that the other participants will have timed therefore large enough that the other participants will have timed
out the sender if a network problem stops the data packets it is out the sender if a network problem stops the data packets it is
sending from reaching the receivers, even allowing for loss of some sending from reaching the receivers, even allowing for loss of some
RTCP packets. RTCP packets.
If a sender is transmitting a large number of RTP media streams, such If a sender is transmitting a large number of RTP media streams, such
that the corresponding RTCP SR or RR packets are too large to fit that the corresponding RTCP SR or RR packets are too large to fit
into the network MTU, this will force the receiver to generate RTCP into the network MTU, the receiver will generate RTCP SR or RR
SR or RR packets in a round-robin manner. In this case, the sender packets in a round-robin manner. In this case, the sender SHOULD
MAY treat receipt of an RTCP SR or RR packet corresponding to an SSRC treat receipt of an RTCP SR or RR packet corresponding to any SSRC it
it sent using the same 5-tuple of source and destination IP address, sent on the same 5-tuple of source and destination IP address, port,
port, and protocol, as an indication that the receiver and return and protocol, as an indication that the receiver and return path are
path are working to prevent the RTCP timeout circuit breaker from working, preventing the RTCP timeout circuit breaker from triggering.
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 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
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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, generally 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 that are 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 the second term on
approximated as follows with reasonable accuracy [Mathis]: the denominator is small, so this formula can be approximated with
reasonable accuracy as follows [Mathis]:
s s
X = --------------- X = -----------------
R * sqrt(p*2/3) R * 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
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throughput: the round trip time, R, and the loss event rate, p (the throughput: the 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 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, not caring about when the losses occurred). fraction of packets lost, and does not report when the packet losses
Using the loss fraction in place of the loss event rate can occurred). Using the loss fraction in place of the loss event rate
overestimate the loss. We believe that this overestimate will not be can overestimate the loss. We believe that this overestimate will
significant, given that we are only interested in order of magnitude not be significant, given that we are only interested in order of
comparison ([Floyd] section 3.2.1 shows that the difference is small magnitude comparison ([Floyd] section 3.2.1 shows that the difference
for steady-state conditions and random loss, but using the loss is small for steady-state conditions and random loss, but using the
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 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 three consecutive RTCP reporting intervals, the sender
SHOULD cease transmission (see Section 4.5). Systems that usually SHOULD cease transmission (see Section 4.5).
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 Systems that usually send at a high data rate, but that can reduce
their sending rate to this lower value to see if this resolves the their data rate significantly (i.e., by at least a factor of ten),
congestion, but MUST then cease transmission if the problem does not MAY first reduce their sending rate to this lower value to see if
resolve itself within a further two RTCP reporting intervals (see this resolves the congestion, but MUST then cease transmission if the
Section 4.5). An example of this might be a video conferencing problem does not resolve itself within a further two RTCP reporting
system that backs off to sending audio only, before completely intervals (see Section 4.5). An example of this might be a video
dropping the call. If such a reduction in sending rate resolves the conferencing system that backs off to sending audio only, before
congestion problem, the sender MAY gradually increase the rate at completely dropping the call. If such a reduction in sending rate
which it sends data after a reasonable amount of time has passed, resolves the congestion problem, the sender MAY gradually increase
provided it takes care not to cause the problem to recur 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 The congestion circuit breaker depends on the fraction of RTP data
RTCP reporting interval (as specified in Section 6.2 of RFC 3550 packets lost in a reporting interval. If the number of packets sent
[RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP in the reporting interval is too low, this statistic loses meaning,
reporting interval is used when determining if the circuit breaker is and it is possible that a sampling error can give the appearance of
triggered. The RTCP reporting interval of the media sender does not high packet loss rates. Following the guidelines in [RFC5405], an
affect how quickly congestion circuit breaker can trigger. The RTP sender that sends not more than one RTP packet per RTT MAY ignore
timing is based on the RTCP reporting interval of the receiver that a single trigger of the congestion circuit breaker, on the basis that
matters (note that RTCP requires all participants in a session to the packet loss rate estimate is unreliable with so few samples.
have similar reporting intervals, else the participant timeout rules However, if the congestion circuit breaker triggers again after the
in [RFC3550] will not work). following three RTCP reporting intervals (i.e., if there have been
six or more consecutive RTCP reporting intervals where the actual
sending rate is more than ten times the estimated sending rate
derived from the TCP throughput equation), then the sender SHOULD
cease transmission (see Section 4.5).
As in Section 4.1, we use two reporting intervals to avoid triggering The RTCP reporting interval of the media sender does not affect how
the circuit breaker on transient failures. This circuit breaker is a quickly congestion circuit breaker can trigger. The timing is based
worst-case condition, and congestion control needs to be performed to on the RTCP reporting interval of the receiver that generates the SR/
keep well within this bound. It is expected that the circuit breaker RR packets from which the loss rate and RTT estimate are derived
will only be triggered if the usual congestion control fails for some (note that RTCP requires all participants in a session to have
reason. similar reporting intervals, else the participant timeout rules in
[RFC3550] will not work, so this interval is likely similar to that
of the sender). If the incoming RTCP SR or RR packets are using a
reduced minimum RTCP reporting interval (as specified in Section 6.2
of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that
reduced RTCP reporting interval is used when determining if the
circuit breaker is triggered.
As in Section 4.1 and Section 4.2, we use three 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 12, line 49 skipping to change at page 13, line 29
the application, media codec, and the amount of error correction and the application, media codec, and the amount of error correction and
packet loss concealment that is applied. There is an upper bound on packet loss concealment that is applied. There is an upper bound on
the amount of loss can be corrected, however, beyond which the media the amount of loss can be corrected, however, beyond which the media
becomes unusable. Similarly, many applications have some upper bound becomes unusable. Similarly, many applications have some upper bound
on the media capture to play-out latency that can be tolerated before 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 becomes unusable. The latency bound will depend on
the application, but typical values can range from the order of a few the application, but typical values can range from the order of a few
hundred milliseconds for voice telephony and interactive conferencing hundred milliseconds for voice telephony and interactive conferencing
applications, up to several seconds for some video-on-demand systems. applications, up to several seconds for some video-on-demand systems.
As a final circuit breaker, applications SHOULD monitor the reported As a final circuit breaker, RTP senders SHOULD monitor the reported
packet loss and delay to estimate whether the media is suitable for packet loss and delay to estimate whether the media is likely to be
the intended purpose. If the packet loss rate and/or latency is such suitable for the intended purpose. If the packet loss rate and/or
that the media has become unusable for the application, and has latency is such that the media has become unusable, and has remained
remained unusable for a significant time period, then the application unusable for a significant time period, then the application SHOULD
SHOULD cease transmission. This memo intentionally does not define a cease transmission. Similarly, receivers SHOULD monitor the quality
bound on the packet loss rate or latency that will result in unusable of the media they receive, and if the quality is unusable for a
media, nor does it specify what time period is deemed significant, as significant time period, they SHOULD terminate the session. This
these are highly application dependent. 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 Sending media that suffers from such high packet loss or latency that
it is unusable at the receiver is both wasteful of resources, and of it is unusable at the receiver is both wasteful of resources, and 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
skipping to change at page 14, line 15 skipping to change at page 14, line 46
Receiving rapid feedback about congestion events potentially allows Receiving rapid feedback about congestion events potentially allows
congestion control algorithms to be more responsive, and to better congestion control algorithms to be more responsive, and to better
adapt the media transmission to the limitations of the network. It adapt the media transmission to the limitations of the network. It
is expected that many RTP congestion control algorithms will adopt is expected that many RTP congestion control algorithms will adopt
the RTP/AVPF profile for this reason, defining new transport layer the RTP/AVPF profile for this reason, defining new transport layer
feedback reports that suit their requirements. Since these reports feedback reports that suit their requirements. Since these reports
are not yet defined, and likely very specific to the details of the are not yet defined, and likely very specific to the details of the
congestion control algorithm chosen, they cannot be used as part of congestion control algorithm chosen, they cannot be used as part of
the generic RTP circuit breaker. the generic RTP circuit breaker.
If the extension for Reduced-Size RTCP [RFC5506] is not used, early Reduced-size RTCP reports sent under the RTP/AVPF early feedback
RTCP feedback packets sent according to the RTP/AVPF profile will be rules that do not contain an RTCP SR or RR packet MUST be ignored by
compound RTCP packets that include an RTCP SR/RR packet. That RTCP the congestion circuit breaker (they do not contain the information
SR/RR packet MUST be processed as if it were sent as a regular RTCP needed by the congestion circuit breaker algorithm), but MUST be
report and counted towards the circuit breaker conditions specified counted as received packets for the RTCP timeout circuit breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that contain RTCP SR or RR packets MUST be processed by the
congestion circuit breaker as if they were sent as regular RTCP
reports, and counted towards the circuit breaker conditions specified
in Section 4 of this memo. This will potentially make the RTP in Section 4 of this memo. This will potentially make the RTP
circuit breaker fire earlier than it would if the RTP/AVPF profile circuit breaker fire earlier than it would if the RTP/AVPF profile
was not used. was not used.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that do not contain an RTCP SR or RR packet MUST be ignored by
the RTP circuit breaker (they do not contain the information used by
the circuit breaker algorithm). Reduced-size RTCP reports sent under
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
counted towards the circuit breaker conditions specified in Section 4
of this memo. This will potentially make the RTP circuit breaker
fire earlier than it would if the RTP/AVPF profile was not used.
When using ECN with RTP (see Section 8), 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 RTP/AVPF
AVPF feedback for generic NACK messages without triggering the RTP feedback for generic NACK messages without triggering the RTP circuit
circuit breaker. This is expected to make such feedback suitable for breaker. This is expected to make such feedback suitable for RTP
RTP congestion control algorithms that need to quickly report loss congestion control algorithms that need to quickly report loss events
events in between regular RTCP reports. The reaction to reduced-size in between regular RTCP reports. The reaction to reduced-size RTCP
RTCP SR/RR packets is to allow such algorithms to send feedback that SR/RR packets is to allow such algorithms to send feedback that can
can trigger the circuit breaker, when desired. trigger the circuit breaker, when desired.
6. Impact of RTCP Extended Reports (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 8), 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.
skipping to change at page 15, line 44 skipping to change at page 16, line 22
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. Security Considerations 9. Impact of Layered Coding
Layered coding is a method of encoding a single media stream into
disparate layers, such that a receiver can decode a subset of the
layers to vary the quality of the media. Layered coding is often
used to aid congestion control in group communication systems, where
a different subset of the layers is sent to each receiver, depending
on the available network capacity.
Media using layered coding can be transported within RTP in several
ways: each layer can be sent as a separate RTP session; each layer
can be sent using a separate SSRC within a single RTP session; or
each layer can be identified by some payload-specific header field,
with all layers being sent by a single SSRC within a single RTP
session. The choice depends on the features provided by the RTP
payload format for the layered encoding, and on the application
requirements.
The RTP circuit breaker operates on a per-RTP session basis. If a
layered encoding is split across multiple RTP sessions, then each
session MUST be treated independently for the RTP circuit breaker.
Within an RTP session, if an application that sends a layered media
encoding using a single SSRC, with the layers identified using some
payload-specific mechanism, then it MUST apply the RTP circuit
breaker to that layered flow as a whole, considering RTCP feedback
for the SSRC sending the layered flow and applying the RTP circuit
breaker as usual.
Within an RTP session, if the layered coding is sent using several
SSRC values within a single RTP session, the flows for those SSRCs
MAY be treated together, so that a circuit breaker trigger for any
SSRC in the layered media flow causes the entire layered flow to
either cease transmission or reduce its sending rate by a factor of
ten. The intent of this is to allow a layered flow to reduce its
sending rate by dropping higher layers if the circuit breaker fails,
rather than requiring the layer that triggered the RTP circuit
breaker to cease transmission (layers are additive in many layered
codecs, so forcing a lower layer to cease transmission while allowing
higher layers to continue is pointless).
10. 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; authentication options are discussed in [RFC7201]. authenticated; authentication options are discussed in [RFC7201].
10. IANA Considerations Timely operation of the RTP circuit breaker depends on the choice of
RTCP reporting interval. If the receiver has a reporting interval
that is overly long, then the responsiveness of the circuit breaker
decreases. In the limit, the RTP circuit breaker can be disabled for
all practical purposes by configuring an RTCP reporting interval that
is many minutes duration. This issue is not specific to the circuit
breaker: long RTCP reporting intervals also prevent reception quality
reports, feedback messages, codec control messages, etc., from being
used. Implementations SHOULD impose an upper limit on the RTCP
reporting interval they are willing to negotiate (based on the
session bandwidth and RTCP bandwidth fraction) when using the RTP
circuit breaker. An upper limit on the reporting interval on the
order of 10 seconds is a reasonable bound.
11. IANA Considerations
There are no actions for IANA. There are no actions for IANA.
11. Acknowledgements 12. Open Issues
o Should the number of RTCP reporting intervals needed to trigger
the media timeout and congestion circuit breakers scale with the
duration of the RTCP reporting interval, so the circuit breaker
triggers after a fixed duration, rather than after a fixed number
of reporting intervals?
13. 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 Gorry Fairhurst, Kevin Gross, Cullen Jennings, Randell Jesup,
Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric Rescorla,
valuable feedback. Abheek Saha, and Fabio Verdicchio, for their valuable feedback.
12. References 14. References
12.1. Normative References 14.1. 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", RFC Friendly Rate Control (TFRC): Protocol Specification", 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
skipping to change at page 17, line 5 skipping to change at page 18, line 43
[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, November Protocol Extended Reports (RTCP XR)", RFC 3611, 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, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
12.2. Informative References 14.2. 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 conference,
2000, DOI 10.1145/347059.347397, 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-03 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-04 (work
in progress), July 2014. in progress), August 2014.
[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 Review
27(3), DOI 10.1145/263932.264023, July 1997. 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 conference,
1998, DOI 10.1145/285237.285291, 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.
[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.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
2008.
[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, March 2009.
[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, April 2009.
[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, September 2009.
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