Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track V. Singh
Expires: September 5, 2012 Aalto University
March 4, 2012
RTP Congestion Control: Circuit Breakers for Unicast Sessions
draft-perkins-avtcore-rtp-circuit-breakers-00
Abstract
The Real-time Transport Protocol (RTP) is widely used for telephony,
video conferencing, and telepresence applications. These
applications are often used over best-effort UDP/IP networks. If
congestion control is not implemented then network congestion will
deteriorate the user's multimedia experience. This document does not
propose a congestion control algorithm. Instead, it specifies a
minimal set of "circuit-breakers". Circuit-breakers are conditions
under which an RTP flow should cease to transmit media to protect the
network from excessive congestion. It is expected that all RTP
applications running on best-effort networks will be able to run
without triggering these circuit breakers in normal operation.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on September 5, 2012.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . . 6
4.1. RTP/AVP Circuit Breaker #1: Timeout . . . . . . . . . . . 7
4.2. RTP/AVP Circuit Breaker #2: Congestion . . . . . . . . . . 8
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 10
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 10
7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 11
8. Session Timeout . . . . . . . . . . . . . . . . . . . . . . . 11
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
9.1. Normative References . . . . . . . . . . . . . . . . . . . 11
9.2. Informative References . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13
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1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used in
voice-over-IP, video teleconferencing, and telepresence systems.
Many of these systems run over best-effort IP networks, and can
suffer from packet loss and increased latency due to network
congestion. Designing effective RTP congestion control algorithms,
to adapt the transmission of RTP-based media to match the available
network capacity, while also maintaining the user experience, is a
difficult but important problem. Many such congestion control and
media adaptation algorithms have been proposed, but to date there is
no consensus on the correct approach, or even that a single standard
algorithm is desirable.
This memo does not attempt to propose a new RTP congestion control
algorithm. Rather, it proposes a minimal set of "circuit breakers";
conditions under which there should be general agreement that an RTP
flow is causing serious congestion, and should cease transmission.
It is expected that any future standards-track congestion control
algorithms for RTP will operate within the envelope defined by this
memo.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Background
We consider congestion control for unicast RTP traffic flows. This
is the problem of adapting the transmission of an audio/visual data
flow, encapsulated within an RTP transport session, from one sender
to one receiver so that it matches the available network bandwidth.
Such adaptation must be done in a way that limits the disruption to
the user experience caused by both packet loss and excessive rate
changes.
Congestion control for unicast RTP traffic can be implemented in one
of two places in the protocol stack. One approach is to run the RTP
traffic over a congestion controlled transport protocol, for example
over TCP, and to adapt the media encoding to match the dictates of
the transport-layer congestion control algorithm. This is safe for
the network, but may be suboptimal for the media quality unless the
transport protocol is designed to support real-time media flows. We
do not consider this class of applications further in this memo, as
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their network safety is guaranteed by the underlying transport.
Alternatively, RTP flows can be run over a non-congestion controlled
transport protocol, for example UDP, performing rate adaptation at
the application layer based on RTP Control Protocol (RTCP) feedback.
With a well-designed, network-aware, application, this allows highly
effective media quality adaptation, but there is potential to disrupt
the network operation if the application does not adapt its sending
rate in a timely and effective manner. This memo focusses on this
class of application.
Congestion control relies on monitoring the delivery of a media flow,
and responding to adapt the transmission of that flow when there are
signs that the network path is congested. Network congestion may be
detected in one of three ways: 1) a receiver may infer the onset of
congestion by observing an increase in one-way delay caused by queue
build-up within the network; 2) if Explicit Congestion Notification
(ECN) [RFC3168] is supported, the network may signal the presence of
congestion by marking packets with ECN Congestion Experienced (CE)
marks; or 3) in the extreme case, congestion will cause packet loss,
which can be detected by observing a gap in the received RTP sequence
numbers. Once the onset of congestion is observed, the receiver must
send feedback to the sender to indicate that the transmission rate
should be reduced. How the sender reduces the transmission rate is
highly dependent on the media codec being used, and is outside the
scope of this memo.
There are several ways in which a receiver may send feedback to a
media sender within the RTP framework:
o The base RTP specification [RFC3550] defines RTCP Reception Report
(RR) packets to convey reception quality feedback information, and
Sender Report (SR) packets to convey information about the media
transmission. RTCP SR packets contain data that can be used to
reconstruct media timing at a receiver, along with a count of the
total number of octets and packets sent. RTCP RR packets report
on the fraction of packets lost in the last reporting interval,
the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by
the number of participants in the session and a configured session
bandwidth estimate. The interval between reports sent from each
receiver tends to be on the order of a few seconds on average, and
it is randomised to avoid synchronisation of reports from multiple
receivers. If a receiver detects problems, the base RTP
specification contains no provisions for sending the feedback
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report early and must wait until the next scheduled reporting
interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not
change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are 1) the extended
packet loss, discard or burst/gap metrics for reacting based on
loss patterns; and 2) the end-system delay metrics for delay-based
congestion control.
o Rapid feedback about the occurrence of congestion events can be
achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585]. This modifies the RTCP timing rules to
allow RTCP reports to be sent early, in some cases immediately,
provided the average RTCP reporting interval remains unchanged.
It also defines new transport-layer feedback messages, including
negative acknowledgements (NACKs), that can be used to report on
specific congestion events. The use of the RTP/AVPF profile is
dependent on signalling, but is otherwise generally backwards
compatible, as it keeps the same average RTCP reporting interval
as the base RTP specification. The Codec control messages
[RFC5104] extend the RTP/AVPF profile with additional feedback
message that can be used to influence that way in which rate
adaptation occurs. The dynamics of how rapidly feedback can be
sent are unchanged.
o Finally, the RTP and RTCP extensions for Explicit Congestion
Notification (ECN) [I-D.ietf-avtcore-ecn-for-rtp] can be used to
provide feedback on the number of packets that received an ECN
Congestion Experienced (CE) mark. This extension builds on the
RTP/AVPF profile to allow rapid congestion feedback.
In addition to these mechanisms for providing feedback, the sender
can include an RTP header extension in each packet to record packet
transmission times. There are two methods: [RFC5450] represents the
transmission time in terms of a time-offset from the RTP timestamp of
the packet, while [RFC6051] includes an explicit NTP-format sending
timestamp (potentially more accurate, but a higher header overhead).
Accurate sending timestamps can be helpful for estimating queuing
delays, to get an early indication of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide
feedback on the senders when congestion events occur, with varying
degrees of timeliness and accuracy. The key distinction is between
systems that use only the basic RTCP mechanisms, without RTP/AVPF
rapid feedback, and those that use the RTP/AVPF extensions, and can
respond to congestion more rapidly.
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4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
The feedback mechanisms defined in [RFC3550] are the minimum that can
be required for a baseline circuit breaker mechanism suitable for all
unicast applications of RTP. Accordingly, for an RTP circuit breaker
to be useful, it should be able to detect that an RTP flow is causing
excessive congestion using only basic RTCP features, without needing
RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.
Three potential congestion signals are available from the basic RTCP
SR/RR packets, and are reported for each SSRC in the RTP session:
1. The sender can estimate the network round-trip time once per RTCP
reporting interval, based on the contents and timing of RTCP SR
and RR packets.
2. Receivers report estimated jitter (the statistical variance of
the RTP data packet inter-arrival time) calculated over the RTCP
reporting interval. Due to the nature of the jitter calculation
([RFC3550], section 6.4.4), the jitter is only meaningful for RTP
flows that send a single data packet for each RTP timestamp value
(i.e., audio flows, or video flows where each frame comprises one
RTP packet).
3. Receivers report the fraction of packets lost during the RTCP
reporting interval, and the cumulative number of packets lost
over the entire RTP session.
These congestion signals limit the possible circuit breakers, since
they give only limited visibility into the behaviour of the network.
RTT estimates are widely used in congestion control algorithms, as a
proxy for queuing delay measures in delay-based congestion control,
or to determine connection timeouts. RTT estimates derived from RTCP
SR and RR packets send according to the RTP/AVP timing rules are too
infrequent to be useful though, and don't give enough information to
distinguish a delay change due to routing updates from queuing delay
caused by congestion. Accordingly, we do not use the RTT estimate
alone as an RTP circuit breaker.
Increased jitter can be a signal of transient network congestion, but
in the highly aggregated form reported in RTCP RR packets, it offers
insufficient information to estimate the extent or persistence of
congestion. Jitter reports are a useful early warning of potential
network congestion, but provide an insufficiently strong signal to be
used as a circuit breaker.
The remaining congestion signals are the packet loss fraction and the
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cumulative number of packets lost. These are robust indicators of
congestion in networks where packet loss is primarily due to queue
overflows, although less accurate in networks where losses can be
caused by non-congestive packet corruption. TCP also uses packet
loss as a congestion signal.
Two packet loss regimes can be observed: 1) RTCP RR packets show a
non-zero packet loss fraction, while the extended highest sequence
number received continues to increment; and 2) RR packets show a loss
fraction of zero, but the extended highest sequence number received
does not increment even though the sender has been transmitting RTP
data packets. The former corresponds to the TCP congestion avoidance
state, and indicates a congested path that is still delivering data;
the latter corresponds to a TCP timeout, and is most likely due to a
path failure. We derive circuit breaker conditions for these two
loss regimes.
4.1. RTP/AVP Circuit Breaker #1: Timeout
If RTP data packets are being sent while the corresponding RTCP RR
packets report a non-increasing extended highest sequence number
received, this is an indication that those RTP data packets are not
reaching the receiver. This could be a short-term issue affecting
only a few packets, perhaps caused by a slow-to-open firewall or a
transient connectivity problem, but if the issue persists, it is a
sign of a more ongoing and significant problem. Accordingly, if a
sender of RTP data packets receives two or more consecutive RTCP RR
packets from the same receiver that correspond to its transmission
and have a non-increasing extended highest sequence number received
field, then that sender SHOULD cease transmission.
Systems that usually send at a high data rate, but which can reduce
their data rate significantly (i.e., by an order of magnitude), MAY
first reduce their sending rate to this lower value, but MUST then
cease transmission if the problem does not resolve itself within a
further two RTCP reporting intervals. An example of this might be a
video conferencing system that backs off to sending audio only,
before completely dropping the call.
The choice of two RTCP reporting intervals is to give enough time for
transient problems to resolve themselves, but to stop problem flows
quickly enough to avoid causing serious problems. A single RTCP
report showing no reception could be caused by numerous transient
faults, and so should not stop transmission. More than two RTCP
reports could avoid false positives, but would lead to problematic
flows running for a long time before being cut off.
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4.2. RTP/AVP Circuit Breaker #2: Congestion
If RTP data packets are being sent, and the corresponding RTCP RR
packets show non-zero packet loss fraction and increasing extended
highest sequence number received, then the RTP data packets are
arriving at the receiver, but some degree of congestion is occurring.
The RTP/AVP profile [RFC3551] states that:
If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable timescale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in timescale and throughput.
The timescale on which TCP throughput is measured is the round-
trip time of the connection. In essence, this requirement states
that it is not acceptable to deploy an application (using RTP or
any other transport protocol) on the best-effort Internet which
consumes bandwidth arbitrarily and does not compete fairly with
TCP within an order of magnitude.
The throughput of a long-lived TCP connection can be estimated using
the TCP throughput equation:
s
X = --------------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))
Where:
X is the transmit rate in bytes/second.
s is the packet size in bytes. If the RTP data packets vary in
size, then the average size should be used.
R is the round trip time in seconds.
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p is the loss event rate, between 0 and 1.0, of the number of loss
events as a fraction of the number of packets transmitted.
t_RTO is the TCP retransmission timeout value in seconds,
approximated by setting t_RTO = 4*R.
b is the number of packets acknowledged by a single TCP
acknowledgement ([RFC3448] recommends the use of b=1 since many
TCP implementations do not use delayed acknowledgements).
This is the same approach to estimated TCP throughput that is used in
[RFC3448]. Two parameters must be estimated in order to calculate
the throughput: the round trip time, R, and the loss event rate, p.
The round trip time can be estimated from RTCP. This is done too
infrequently for accurate statistics, but is the best that can be
done with the standard RTCP mechanisms.
RTCP RR packets contain the packet loss fraction, rather than the
loss event rate, so p cannot be reported (TCP typically treats the
loss of multiple packets within a single RTT as one loss event, but
RTCP RR packets report the overall fraction of packets lost, not
caring about when the losses occurred). Using the loss fraction in
place of the loss event rate can overestimate the loss. We believe
that this overestimate will not be significant, given that we are
only interested in order of magnitude comparison (Floyd et al,
"Equation-Based Congestion Control for Unicast Applications", Proc.
SIGCOMM 2000, section 3.2.1, show that the difference is small for
steady-state conditions and random loss, but using the loss fraction
is more conservative in the case of bursty loss).
The congestion circuit breaker is therefore: when RTCP RR packets are
received, estimate the TCP throughput using the above equation and
the measured R, p (approximated by the loss fraction), and s.
Compare this with the actual sending rate. If the actual sending
rate has been more than an order of magnitude greater than the
throughput equation estimate for two or more RTCP reporting
intervals, stop transmitting.
Again, we use two reporting intervals to avoid triggering the circuit
breaker on transient failures. This circuit breaker is a worst-case
condition, and congestion control should 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.
(tbd -- we need to base the circuit breaker condition on something,
so TCP seems a logical choice. Following TCP limits too closely is
inappropriate for many applications of RTP, though, since they have
different dynamics. Is the above lax enough to not disrupt valid
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applications, but tight enough to provide meaningful protection for
the network?)
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile
More rapid feedback allows more responsiveness. The receiver SHOULD
provide feedback more often during, or at onset of, congestion, and
provide feedback less often when there is no congestion.
(tbd -- mechanisms may probably need to be designed in conjunction
with the different classes of congestion control that can leverage
RTP/AVPF; e.g., we might need to specify limits for TFRC-like or
delay-based algorithms using RTP/AVPF feedback.)
(tbd -- a high-level question to be answered is whether we need to
specify anything different for the circuit breaker for AVPF, or if we
leave that unchanged, and focus solely on the dynamics, to ensure the
circuit breaker is never triggered.)
6. Impact of RTCP XR
(tbd)
This improves the information, but doesn't change the dynamics of the
congestion control loop. Suspect the impact will actually be quite
small.
Packets discarded [I-D.ietf-xrblock-rtcp-xr-discard] or bytes
discarded [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] due to late
arrival by the receiver may indicate congestion. Congestion control
should consider the discarded packets as if they were lost packets.
The RTCP RR reports the loss fraction over an RTCP interval which is
insufficient to distinguish between solitary or bursty losses. To
provide rough sense of duration of losses or discards, an endpoint
may use burst/gap reporting for loss
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] and discard
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]. For more accurate
reporting the receiver may use Run-length encoded (RLE) lost
[RFC3611] or discarded [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics]
packets.
For precise measurement of network roundtrip delay the receiver can
signal its end-system delay [I-D.ietf-xrblock-rtcp-xr-delay]
[RFC3611].
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A receiver may also indicate onset or end of congestion by reporting
the distribution of the inter-packet delay variation
[I-D.ietf-xrblock-rtcp-xr-pdv] [RFC3611].
7. Impact of Explicit Congestion Notification (ECN)
ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal rate at
which to send. However, it seems unwise to treat the receipt of
multiple ECN-CE marked packets as a circuit breaker, since it is
likely that ECN-capable and non-ECN-capable paths will exist for a
long time to come. Rather, consider packet loss as the circuit
breaker condition as for non-ECN flows.
8. Session Timeout
From a usability perspective, if there is no audio or video response
from the other peer, it is likely that the user may terminate the
session.
According to RFC 3550 [RFC3550], any participant that has not sent an
RTP packet within the last two RTCP interval is removed from the
sender list. To avoid timing out the specific flow, the endpoint
MUST send corresponding RTCP reports. Interactive Connectivity
Establishment (ICE) [RFC5245] recommends that the timeout MUST NOT be
less than 15 seconds.
If no RTCP RR arrives for two complete SR intervals, the sender
SHOULD cease transmission. However, if the endpoint can reduce the
media rate then it MAY first reduce the rate to the lower value, but
terminate the transmission if still no RTCP RR is received in the
next two SR intervals.
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
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Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611,
November 2003.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
9.2. Informative References
[I-D.ietf-avtcore-ecn-for-rtp]
Westerlund, M., Johansson, I., Perkins, C., and K.
Carlberg, "Explicit Congestion Notification (ECN) for RTP
over UDP", draft-ietf-avtcore-ecn-for-rtp-06 (work in
progress), February 2012.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]
Clark, A., Hunt, G., Wu, W., and R. Huang, "RTCP XR Report
Block for Burst/Gap Discard metric Reporting",
draft-ietf-xrblock-rtcp-xr-burst-gap-discard-02 (work in
progress), January 2012.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]
Clark, A., Hunt, G., Zhao, J., Wu, W., and S. Zhang, "RTCP
XR Report Block for Burst/Gap Loss metric Reporting",
draft-ietf-xrblock-rtcp-xr-burst-gap-loss-01 (work in
progress), January 2012.
[I-D.ietf-xrblock-rtcp-xr-delay]
Hunt, G., Gross, K., and A. Clark, "RTCP XR Report Block
for Delay metric Reporting",
draft-ietf-xrblock-rtcp-xr-delay-01 (work in progress),
December 2011.
[I-D.ietf-xrblock-rtcp-xr-discard]
Hunt, G., Clark, A., Zorn, G., and W. Wu, "RTCP XR Report
Block for Discard metric Reporting",
draft-ietf-xrblock-rtcp-xr-discard-01 (work in progress),
December 2011.
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[I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics]
Ott, J., Singh, V., and I. Curcio, "Real-time Transport
Control Protocol (RTCP) Extension Report (XR) for Run
Length Encoding of Discarded Packets",
draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-03 (work in
progress), February 2012.
[I-D.ietf-xrblock-rtcp-xr-pdv]
Hunt, G. and A. Clark, "RTCP XR Report Block for Packet
Delay Variation Metric Reporting",
draft-ietf-xrblock-rtcp-xr-pdv-02 (work in progress),
December 2011.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, September 2001.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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Varun Singh
Aalto University
School of Science and Technology
Otakaari 5 A
Espoo, FIN 02150
Finland
Email: varun@comnet.tkk.fi
URI: http://www.netlab.tkk.fi/~varun/
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