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


   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

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   provisions of BCP 78 and BCP 79.

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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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   ( in effect on the date of
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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

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   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

   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

   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:

     X = --------------------------------------------------------------
         R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))


   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


   This improves the information, but doesn't change the dynamics of the
   congestion control loop.  Suspect the impact will actually be quite

   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]

   For precise measurement of network roundtrip delay the receiver can
   signal its end-system delay [I-D.ietf-xrblock-rtcp-xr-delay]

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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

   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

              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.

              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.

              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.

              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.

              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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              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.

              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


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   Varun Singh
   Aalto University
   School of Science and Technology
   Otakaari 5 A
   Espoo, FIN  02150


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