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Versions: (draft-perkins-rmcat-rtp-cc-feedback) 00 01 02 03

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Informational                         November 14, 2016
Expires: May 18, 2017


     RTP Control Protocol (RTCP) Feedback for Congestion Control in
                   Interactive Multimedia Conferences
                  draft-ietf-rmcat-rtp-cc-feedback-03

Abstract

   This memo discusses the types of congestion control feedback that it
   is possible to send using the RTP Control Protocol (RTCP), and their
   suitability of use in implementing congestion control for unicast
   multimedia applications.

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
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   This Internet-Draft will expire on May 18, 2017.

Copyright Notice

   Copyright (c) 2016 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
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   described in the Simplified BSD License.



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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Possible Models for RTCP Feedback . . . . . . . . . . . . . .   2
   3.  What Feedback is Achievable With RTCP?  . . . . . . . . . . .   4
     3.1.  Scenario 1: Voice Telephony . . . . . . . . . . . . . . .   4
     3.2.  Scenario 2: Point-to-Point Video Conference . . . . . . .   6
     3.3.  Scenario 3: Group Video Conference  . . . . . . . . . . .  10
     3.4.  Scenario 4: Screen Sharing  . . . . . . . . . . . . . . .  10
   4.  Discussion and Conclusions  . . . . . . . . . . . . . . . . .  10
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  10
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  11
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  11
   8.  Informative References  . . . . . . . . . . . . . . . . . . .  11
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  12

1.  Introduction

   The coming deployment of WebRTC systems raises the prospect that high
   quality video conferencing will see extremely wide use.  To ensure
   the stability of the network in the face of this use, WebRTC systems
   will need to use some form of congestion control for their RTP-based
   media traffic.  To develop such congestion control, it is necessary
   to understand the sort of congestion feedback that can be provided
   within the framework of RTP [RFC3550] and the RTP Control Protocol
   (RTCP).  It then becomes possible to determine if this is sufficient
   for congestion control, or if some form of RTP extension is needed.

   This memo considers the congestion feedback that can be sent using
   RTCP under the RTP/SAVPF profile [RFC5124] (the secure version of the
   RTP/AVPF profile [RFC4585]).  This profile was chosen as it forms the
   basis for media transport in WebRTC [I-D.ietf-rtcweb-rtp-usage]
   systems.  Nothing in this memo is specific to the secure version of
   the profile, or to WebRTC, however.

2.  Possible Models for RTCP Feedback

   Several questions need to be answered when providing RTCP reception
   quality feedback for congestion control purposes.  These include:

   o  How often is feedback needed?

   o  How much overhead is acceptable?

   o  How much, and what, data does each report contain?

   The key question is how often does the receiver need to send feedback
   on the reception quality it is experiencing, and hence the congestion



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   state of the network?  Traditional congestion control protocols, such
   as TCP, send acknowledgements with every packet (or, at least, every
   couple of packets).  That is straight-forward and low overhead when
   traffic is bidirectional and acknowledgements can be piggybacked onto
   return path data packets.  It can also be acceptable, and can have
   reasonable overhead, to send separate acknowledgement packets when
   those packets are much smaller than data packets.  It becomes a
   problem, however, when there is no return traffic on which to
   piggyback acknowledgements, and when acknowledgements are similar in
   size to data packets; this can be the case for some forms of media
   traffic, especially for voice over IP (VoIP) flows, but less so for
   video.

   When considering multimedia traffic, it might make sense to consider
   less frequent feedback.  For example, it might be possible to send a
   feedback packet once per video frame, or every few frames, or once
   per network round trip time (RTT).  This could still give
   sufficiently frequent feedback for the congestion control loop to be
   stable and responsive while keeping the overhead reasonable when the
   feedback cannot be piggybacked onto returning data.  In this case, it
   is important to note that RTCP can send much more detailed feedback
   than simple acknowledgements.  For example, if it were useful, it
   could be possible to use an RTCP extended report (XR) packet
   [RFC3611] to send feedback once per RTT comprising a bitmap of lost
   and received packets, with reception times, over that RTT.  As long
   as feedback is sent frequently enough that the control loop is
   stable, and the sender is kept informed when data leaves the network
   (to provide an equivalent to ACK clocking in TCP), it is not
   necessary to report on every packet at the instant it is received
   (indeed, it is unlikely that a video codec can react instantly to a
   rate change anyway, and there is little point in providing feedback
   more often than the codec can adapt).

   The amount of overhead due to congestion control feedback that is
   considered acceptable has to be determined.  RTCP data is sent in
   separate packets to RTP data, and this has some cost in terms of
   additional header overhead compared to protocols that piggyback
   feedback on return path data packets.  The RTP standards have long
   said that a 5% overhead for RTCP traffic generally acceptable, while
   providing the ability to change this fraction.  Is this still the
   case for congestion control feedback?  Or is there a desire to either
   see more responsive feedback and congestion control, possibility with
   a higher overhead, or is lower overhead wanted, accepting that this
   might reduce responsiveness of the congestion control algorithm?

   Finally, the details of how much, and what, data is to be sent in
   each report will affect the frequency and/or overhead of feedback.
   There is a fundamental trade-off that the more frequently feedback



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   packets are sent, the less data can be included in each packet to
   keep the overhead constant.  Does the congestion control need high
   rate but simple feedback (e.g., like TCP acknowledgements), or is it
   acceptable to send more complex feedback less often?

3.  What Feedback is Achievable With RTCP?

3.1.  Scenario 1: Voice Telephony

   In many ways, point-to-point voice telephony is the simplest scenario
   for congestion control, since there is only a single media stream to
   control.  It's complicated, however, by severe bandwidth constraints
   on the feedback, to keep the overhead manageable.

   Assume a two-party point-to-point voice-over-IP call, using RTP over
   UDP/IP.  A rate adaptive speech codec, such as Opus, is used, encoded
   into RTP packets in frames of duration Tf seconds (Tf = 20ms in many
   cases, but values up to 60ms are not uncommon).  The congestion
   control algorithm requires feedback every Nr frames, i.e., every Nr *
   Tf seconds, to ensure effective control.  Both parties in the call
   send speech data or comfort noise with sufficient frequency that they
   are counted as senders for the purpose of the RTCP reporting interval
   calculation.

   RTCP feedback packets can be full, compound, RTCP feedback packets,
   or non-compound RTCP packets.  A compound RTCP packet is sent once
   for every Nnc non-compound RTCP packets.

   Compound RTCP packets contain a Sender Report (SR) packet and a
   Source Description (SDES) packet, and an RTP Congestion Control
   Feedback (RC2F) packet [I-D.dt-rmcat-feedback-message].  Non-compound
   RTCP packets contain only the RC2F packet.  Since each participant
   sends only a single media stream, the extensions for RTCP report
   aggregation [I-D.ietf-avtcore-rtp-multi-stream] and reporting group
   optimisation [I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not
   used.

   Within each compound RTCP packet, the SR packet will contain a sender
   information block (28 octets) and a single reception report block (24
   octets), for a total of 52 octets.  A minimal SDES packet will
   contain a header (4 octets) and a single chunk containing an SSRC (4
   octets) and a CNAME item, and if the recommendations for choosing the
   CNAME [RFC7022] are followed, the CNAME item will comprise a 2 octet
   header, 16 octets of data, and 2 octets of padding, for a total SDES
   packet size of 28 octets.  The RC2F packets contains an XR block
   header and SSRC (8 octets), a block type and timestamp (8 octets),
   the SSRC, beginning and ending sequence numbers (8 octets), and 2*Nr
   octets of reports, for a total of 24 + 2*Nr octets.  If IPv4 is used,



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   with no IP options, the UDP/IP header will be 28 octets in size.
   This gives a total compound RTCP packet size of Sc = 132 + 2*Nr
   octets.

   The non-compound RTCP packets will comprise just the RC2F packet with
   a UDP/IP header.  It can be seen that these packets will be Snc = 48
   + 2*Nr octets in size.

   The RTCP reporting interval calculation ([RFC3550], Section 6.2) for
   a two-party session where both participants are senders, reduces to
   Trtcp = n * Srtcp/Brtcp where Srtcp = (Sc + Nnc * Snc)/(1 + Nnc) is
   the average RTCP packet size in octets, Brtcp is the bandwidth
   allocated to RTCP in octets per second, and n is the number of
   participants (n=2 in this scenario).

   To ensure a report is sent every Nr frames, it is necessary to set
   the RTCP reporting interval Trtcp = Nr * Tf, which when substituted
   into the previous gives Nr * Tf = n * Srtcp/Brtcp.

   Solving for the RTCP bandwidth, Brtcp, and expanding the definition
   of Srtcp gives Brtcp = (n * (Sc + Nnc * Snc))/(Nr * Tf * (1 + Nnc)).

   If we assume every report is a compound RTCP packet (i.e., Nnc = 0),
   the frame duration Tf = 20ms, and an RTCP report is sent for every
   second frame (i.e., 25 RTCP reports per second), this expression
   gives the needed RTCP bandwidth Brtcp = 53.1kbps.  Increasing the
   frame duration, or reducing the frequency of reports, reduces the
   RTCP bandwidth, as shown below:

              +--------------+-------------+----------------+
              | Tf (seconds) | Nr (frames) | rtcp_bw (kbps) |
              +--------------+-------------+----------------+
              |     20ms     |      2      |      53.1      |
              |     20ms     |      4      |      27.3      |
              |     20ms     |      8      |      14.5      |
              |     20ms     |      16     |      8.01      |
              |     60ms     |      2      |      17.7      |
              |     60ms     |      4      |      9.1       |
              |     60ms     |      8      |      4.8       |
              |     60ms     |      16     |      2.66      |
              +--------------+-------------+----------------+

            Table 1: Required RTCP bandwidth for VoIP feedback

   The final row of the table (60ms frames, report every 16 frames)
   sends RTCP reports once per second, giving an RTCP bandwidth of
   2.66kbps.




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   The overhead can be reduced by sending some reports in non-compound
   RTCP packets [RFC5506].  For example, if we alternate compound and
   non-compound RTCP packets, i.e., Nnc = 1, the calculation gives:

              +--------------+-------------+----------------+
              | Tf (seconds) | Nr (frames) | rtcp_bw (kbps) |
              +--------------+-------------+----------------+
              |     20ms     |      2      |      36.7      |
              |     20ms     |      4      |      19.1      |
              |     20ms     |      8      |      10.4      |
              |     20ms     |      16     |      6.0       |
              |     60ms     |      2      |      12.2      |
              |     60ms     |      4      |      6.4       |
              |     60ms     |      8      |      3.5       |
              |     60ms     |      16     |      2.0       |
              +--------------+-------------+----------------+

      Table 2: Required RTCP bandwidth for VoIP feedback (alternating
                    compound and non-compound reports)

   The RTCP bandwidth needed for 60ms frames, reporting every 16 frames
   (once per second), can be seen to drop to 2.01kbps.  This calculation
   can be repeated for other patterns of compound and non-compound RTCP
   packets, feedback frequency, and frame duration, as needed.

   Note: To achieve the RTCP transmission intervals above the RTP/SAVPF
   profile with T_rr_interval=0 is used, since even when using the
   reduced minimal transmission interval, the RTP/SAVP profile would
   only allow sending RTCP at most every 0.11s (every third frame of
   video).  Using RTP/SAVPF with T_rr_interval=0 however is capable of
   fully utilizing the configured 5% RTCP bandwidth fraction.

3.2.  Scenario 2: Point-to-Point Video Conference

   Consider a point to point video call between two end systems.  There
   will be four RTP flows in this scenario, two audio and two video,
   with all four flows being active for essentially all the time (the
   audio flows will likely use voice activity detection and comfort
   noise to reduce the packet rate during silent periods, and does not
   cause the transmissions to stop).

   Assume all four flows are sent in a single RTP session, each using a
   separate SSRC; the RTCP reports from co-located audio and video SSRCs
   at each end point are aggregated [I-D.ietf-avtcore-rtp-multi-stream];
   the optimisations in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
   are used; and congestion control feedback is sent
   [I-D.dt-rmcat-feedback-message].




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   When all members are senders, the RTCP timing rules in Section 6.2
   and 6.3 of [RFC3550] and [RFC4585] reduce to:

                 rtcp_interval = avg_rtcp_size * n / rtcp_bw

   where n is the number of members in the session, the avg_rtcp_size is
   measured in octets, and the rtcp_bw is the bandwidth available for
   RTCP, measured in octets per second (this will typically be 5% of the
   session bandwidth).

   The average RTCP size will depend on the amount of feedback that is
   sent in each RTCP packet, on the number of members in the session, on
   the size of source description (RTCP SDES) information sent, and on
   the amount of congestion control feedback sent in each packet.

   As a baseline, each RTCP packet will be a compound RTCP packet that
   contains an aggregate of a compound RTCP packet generated by the
   video SSRC and a compound RTCP packet generated by the audio SSRC.
   Since the RTCP reporting group extensions are used, one of these
   SSRCs will be a reporting SSRC, and the other will delegate its
   reports to that.

   The aggregated compound RTCP packet from the non-reporting SSRC will
   contain an RTCP SR packet, an RTCP SDES packet, and an RTCP RGRS
   packet.  The RTCP SR packet contains the 28 octet header and sender
   information, but no report blocks (since the reporting is delegated).
   The RTCP SDES packet will comprise a header (4 octets), originating
   SSRC (4 octets), a CNAME chunk, a terminating chunk, and any padding.
   If the CNAME follows [RFC7022] and [I-D.ietf-rtcweb-rtp-usage] it
   will be 18 octets in size, and will need 1 octet of padding, making
   the SDES packet 28 octets in size.  The RTCP RGRS packet will be 12
   octets in size.  This gives a total of 28 + 28 + 12 = 68 octets.

   The aggregated compound RTCP packet from the reporting SSRC will
   contain an RTCP SR packet, an RTCP SDES packet, and an RTCP XR
   congestion control feedback packet.  The RTCP SR packet will contain
   two report blocks, one for each of the remote SSRCs (the report for
   the other local SSRC is suppressed by the reporting group extension),
   for a total of 28 + (2 * 24) = 76 octets.  The RTCP SDES packet will
   comprise a header (4 octets), originating SSRC (4 octets), a CNAME
   chunk, an RGRP chunk, a terminating chunk, and any padding.  If the
   CNAME follows [RFC7022] and [I-D.ietf-rtcweb-rtp-usage] it will be 18
   octets in size.  The RGRP chunk similarly comprises 18 octets, and 3
   octets of padding are needed, for a total of 48 octets.  The RTCP XR
   congestion control feedback report comprises an 8 octet XR header, an
   8 octet RC2F header, then for each of the remote audio and video
   SSRCs, an 8 octet report header, and 2 octets per packet reported
   upon, and padding to a 4 octet boundary, if needed; that is 8 + 8 + 8



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   + (2 * Nv) + 8 + (2 * Na) where Nv is the number of video packets per
   report, and Na is the number of audio packets per report.

   The complete compound RTCP packet contains the RTCP packets from both
   the reporting and non-reporting SSRCs, an SRTP authentication tag,
   and a UDP/IPv4 header.  The size of this RTCP packet is therefore:
   252 + (2 * Nv) + (2 * Na) octets.  Since the aggregate RTCP packet
   contains reports from two SSRCs, the RTCP packet size is halved
   before use [I-D.ietf-avtcore-rtp-multi-stream].  Accordingly, we
   define Sc = (252 + (2 * Nv) + (2 * Na))/2 for this scenario.

   How many packets does the RTCP XR congestion control feedback packet
   report on?  This is obviously highly dependent on the choice of codec
   and encoding parameters, and might be quite bursty if the codec sends
   I-frames from which later frames are predicted.  For now though,
   assume constant rate media with an MTU around 1500 octets, with
   reports for both audio and video being aggregated and sent to align
   with video frames.  This gives the following, assuming Nr =1 and Nnc
   = 0 (i.e., send a compound RTCP packet for each video frame, and no
   non-compound packets), and using the calculation from Scenario 1:
   Brtcp = (n * (Sc + Nnc * Snc))/(Nr * Tf * (1 + Nnc))

   +---------+---------+--------------+--------------+-----------------+
   |   Data  |  Video  |    Video     |    Audio     |  Required RTCP  |
   |   Rate  |  Frame  | Packets per  | Packets per  |    bandwidth:   |
   |  (kbps) |   Rate  |  Report: Nv  |  Report: Na  |   Brtcp (kbps)  |
   +---------+---------+--------------+--------------+-----------------+
   |   100   |    8    |      1       |      6       |    33.3 (33%)   |
   |   200   |    16   |      1       |      3       |    65.0 (33%)   |
   |   350   |    30   |      1       |      2       |   120.1 (35%)   |
   |   700   |    30   |      2       |      2       |   121.9 (17%)   |
   |   700   |    60   |      1       |      1       |   240.0 (34%)   |
   |   1024  |    30   |      3       |      2       |   122.8 (12%)   |
   |   1400  |    60   |      2       |      1       |   241.8 (17%)   |
   |   2048  |    30   |      6       |      2       |   125.6 ( 6%)   |
   |   2048  |    60   |      3       |      1       |   243.8 (12%)   |
   |   4096  |    30   |      12      |      2       |   131.3 ( 3%)   |
   |   4096  |    60   |      6       |      1       |   294.4 ( 6%)   |
   +---------+---------+--------------+--------------+-----------------+

        Table 3: Required RTCP bandwidth, reporting on every frame

   The RTCP bandwidth needed scales inversely with Nr.  That is, it is
   halved if Nr=2 (report on every second packet), is reduced to one-
   third if Nr=3 (report on every third packet), and so on.

   The needed RTCP bandwidth scales as a percentage of the data rate
   following the ratio of the frame rate to the data rate.  As can be



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   seen from the table above, the RTCP bandwidth needed is a significant
   fraction of the media rate, if reporting on every frame for low rate
   video.  This can be solved by reporting less often at lower rates.
   For example, to report on every frame of 100kbps/8fps video requires
   the RTCP bandwidth to be 21% of the media rate; reporting every
   fourth frame (i.e., twice per second) reduces this overhead to 5%.

   Use of reduced size RTCP [RFC5506] would allow the SR and SDES
   packets to be omitted from some reports.  These "non-compound"
   (actually, compound but reduced size in this case) RTCP packets would
   contain an RTCP RGRS packet from the non-reporting SSRC, and an RTCP
   SDES RGRP packet and a congestion control feedback packet from the
   reporting SSRC.  This will be 12 + 28 + 12 + 8 + 2*Nv + 8 + 2*Na
   octets, plus UDP/IP header.  That is, Snc = (96 + 2*Nv + 2*Na)/2.
   Repeating the analysis above, but alternating compound and non-
   compound reports, i.e., setting Nnc = 1, gives:

   +---------+---------+--------------+--------------+-----------------+
   |   Data  |  Video  |    Video     |    Audio     |  Required RTCP  |
   |   Rate  |  Frame  | Packets per  | Packets per  |    bandwidth:   |
   |  (kbps) |   Rate  |  Report: Nv  |  Report: Na  |   Brtcp (kbps)  |
   +---------+---------+--------------+--------------+-----------------+
   |   100   |    8    |      1       |      6       |    23.5 (23%)   |
   |   200   |    16   |      1       |      3       |    45.5 (23%)   |
   |   350   |    30   |      1       |      2       |    84.4 (24%)   |
   |   700   |    30   |      2       |      2       |    85.3 (12%)   |
   |   700   |    60   |      1       |      1       |   166.9 (24%)   |
   |   1024  |    30   |      3       |      2       |    86.2 ( 8%)   |
   |   1400  |    60   |      2       |      1       |   168.8 (12%)   |
   |   2048  |    30   |      6       |      2       |    89.1 ( 4%)   |
   |   2048  |    60   |      3       |      1       |   170.6 ( 8%)   |
   |   4096  |    30   |      12      |      2       |    94.7 ( 2%)   |
   |   4096  |    60   |      6       |      1       |   176.3 ( 4%)   |
   +---------+---------+--------------+--------------+-----------------+

     Table 4: Required RTCP bandwidth, reporting on every frame, with
                           reduced-size reports

   The use of reduced-size RTCP gives a noticeable reduction in the
   needed RTCP bandwidth, and can be combined with reporting every few
   frames rather than every frames.  Overall, it is clear that the RTCP
   overhead can be reasonable across the range of data and frame rates,
   if RTCP is configured carefully.








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3.3.  Scenario 3: Group Video Conference

   (tbd)

3.4.  Scenario 4: Screen Sharing

   (tbd)

4.  Discussion and Conclusions

   RTCP as it is currently specified cannot be used to send per-packet
   congestion feedback.  RTCP can, however, be used to send congestion
   feedback on each frame of video sent, provided the session bandwidth
   exceeds a couple of megabits per second (the exact rate depending on
   the number of session participants, the RTCP bandwidth fraction, and
   what RTCP extensions are enabled, and how much detail of feedback is
   needed).  For lower rate sessions, the overhead of reporting on every
   frame becomes high, but can be reduced to something reasonable by
   sending reports once per N frames (e.g., every second frame), or by
   sending non-compound RTCP reports in between the regular reports.

   If it is desired to use RTCP in something close to it's current form
   for congestion feedback in WebRTC, the multimedia congestion control
   algorithm needs be designed to work with feedback sent every few
   frames, since that fits within the limitations of RTCP.  That
   feedback can be a little more complex than just an acknowledgement,
   provided care is taken to consider the impact of the extra feedback
   on the overhead, possibly allowing for a degree of semantic feedback,
   meaningful to the codec layer as well as the congestion control
   algorithm.

   The format described in [I-D.dt-rmcat-feedback-message] seems
   sufficient for the needs of congestion control feedback.  There is
   little point optimising this format: the main overhead comes from the
   UDP/IP headers and the other RTCP packets included in the compound
   packets, and can be lowered by using the [RFC5506] extensions and
   sending reports less frequently.

   Further study of the scenarios of interest is needed, to ensure that
   the analysis presented is applicable to other media topologies, and
   to sessions with different data rates and sizes of membership.

5.  Security Considerations

   An attacker that can modify or spoof RTCP congestion control feedback
   packets can manipulate the sender behaviour to cause denial of
   service.  This can be prevented by authentication and integrity




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   protection of RTCP packets, for example using the secure RTP profile
   [RFC3711][RFC5124], or by other means as discussed in [RFC7201].

6.  IANA Considerations

   There are no actions for IANA.

7.  Acknowledgements

   Thanks to Magnus Westerlund and the members of the RMCAT feedback
   design team for their feedback.

8.  Informative References

   [I-D.dt-rmcat-feedback-message]
              Sarker, Z., Perkins, C., Singh, V., and M. Ramalho, "RTP
              Control Protocol (RTCP) Feedback for Congestion Control",
              draft-dt-rmcat-feedback-message-01 (work in progress),
              October 2016.

   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
              December 2015.

   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-12 (work
              in progress), March 2016.

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
              2016.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
              "RTP Control Protocol Extended Reports (RTCP XR)",
              RFC 3611, DOI 10.17487/RFC3611, November 2003,
              <http://www.rfc-editor.org/info/rfc3611>.



Perkins                   Expires May 18, 2017                 [Page 11]


Internet-Draft    RTCP Feedback for Congestion Control     November 2016


   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,
              <http://www.rfc-editor.org/info/rfc3711>.

   [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,
              DOI 10.17487/RFC4585, July 2006,
              <http://www.rfc-editor.org/info/rfc4585>.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <http://www.rfc-editor.org/info/rfc5506>.

   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
              September 2013, <http://www.rfc-editor.org/info/rfc7022>.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
              <http://www.rfc-editor.org/info/rfc7201>.

Author's Address

   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org












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