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Versions: (draft-perkins-rmcat-rtp-cc-feedback) 00 01 02 03 04 05

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Informational                              July 8, 2016
Expires: January 9, 2017


   Using RTP Control Protocol (RTCP) Feedback for Unicast Multimedia
                           Congestion Control
                  draft-ietf-rmcat-rtp-cc-feedback-01

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
   Task Force (IETF).  Note that other groups may also distribute
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   This Internet-Draft will expire on January 9, 2017.

Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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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.  Per-packet Feedback . . . . . . . . . . . . . . . . . . .   4
     3.2.  Per-frame Feedback  . . . . . . . . . . . . . . . . . . .   4
     3.3.  Per-RTT Feedback  . . . . . . . . . . . . . . . . . . . .   6
   4.  Discussion and Conclusions  . . . . . . . . . . . . . . . . .   7
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .   7
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   7
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   7
   8.  Informative References  . . . . . . . . . . . . . . . . . . .   7
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .   9

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
   state of the network?  Traditional congestion control protocols, such



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   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 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
   packets are sent, the less data can be included in each packet to
   keep the overhead constant.  Does the congestion control need high



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   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.  Per-packet Feedback

   RTCP packets are sent as separate packets to RTP media data, and the
   protocol includes no mechanism for piggybacking an RTCP packet onto
   an RTP data packet.  In addition, the RTCP timing rules are based on
   the size of the RTP session, the number of active senders, the RTCP
   packet size, and the configured RTCP bandwidth fraction, with
   randomisation to prevent synchronisation of reports; accordingly the
   RTCP packet transmission times are extremely unlikely to line up with
   RTP packet transmission times.  As a result, RTCP cannot be used to
   send per-packet feedback in it's current form.

   All of these issues with using RTCP for per-packet feedback could be
   resolved in an update to the RTP protocol, of course.  Such an update
   could change the RTCP timing rules, and might define a shim layer to
   allow multiplexing of RTP and RTCP into a single packet, or to extend
   the RTP header to piggyback feedback data.  This sort of change would
   be a large, and almost certainly backwards incompatible, extension to
   the RTP protocol, and is unlikely to be completed quickly, but could
   be done if there was a need.

3.2.  Per-frame Feedback

   Consider one of the simplest scenarios for WebRTC: 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 [RFC3550],
   [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].

   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





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   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,
   then for each of the remote audio and video SSRCs, a 12 octet report
   header, and 2 octets per packet reported upon, and padding to a 4
   octet boundary, if needed; that is 8 + 12 + (2 *
   video_packets_per_report) + 12 + (2 * audio_packets_per_report).  =
   32 + (2 * video_packets_per_report) + (2 * audio_packets_per_report).
   The compound RTCP packet will be 156 + (2 * video_packets_per_report)
   + (2 * audio_packets_per_report).




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   The resulting aggregate RTCP packet, containing both compound RTCP
   packets, will be sent in UDP/IPv4 with no IP options and using Secure
   RTP, which adds 20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit
   Authentication tag) = 42 octets, the avg_rtcp_size will therefore be
   42 + 68 + 156 + (2 * video_packets_per_report) + (2 *
   audio_packets_per_report).  (FIXME: this ignores padding in RTCP)
   Since the aggregate RTCP packet contains reports from two SSRCs, the
   avg_rtcp_packet size is halved before use
   [I-D.ietf-avtcore-rtp-multi-stream].  The value n is this scenario is
   4, and the rtcp_bw is assumed to be 5% of the session bandwidth.

   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, assume
   video_packets_per_second = (video_bit_rate_bps / 8) / mtu and
   video_packets_per_report = video_packets_per_seconds / fps.  For
   audio, assume 50 packets per second, with audio_packets_per_report
   based on the video frame rate (i.e., RTCP packets for the audio SSRC
   are aggregated with those from the video SSRC).

   If it is desired to send RTCP feedback packets on average 30 times
   per second, to correspond to one RTCP report every frame for 30fps
   video, one can solve the above expressions to determine the session
   bandwidth needed to give an RTCP reporting interval of 1/30 second.
   This is approximately 2.5Mbps.  That is, provided the video session
   bandwidth is greater than approximately 2.5Mbps, one can report on
   each packet arrival (with ECN marks and arrival time) for every frame
   of 30 fps video, using existing RTCP mechanisms.  This is not out of
   line with the expected session bandwidth for this type of
   application, suggesting the RTCP feedback can be used to provide per-
   frame congestion control feedback for WebRTC-style applications.

      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.

3.3.  Per-RTT Feedback

   The arguments made in Section 3.2 apply to this case as well.  The
   network RTT will usually be larger than the media framing interval,
   so sending feedback per RTT is less of a load on RTCP than sending
   feedback per frame.




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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).  RTCP can likely also be used to send feedback on a per-RTT
   basis, provided the RTT is not too low.

   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 roughly each
   frame or each RTT, rather than per packet, 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.

   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

   The security considerations of [RFC3550], [RFC4585], and [RFC5124]
   apply.

6.  IANA Considerations

   There are no actions for IANA.

7.  Acknowledgements

   Thanks to Magnus Westerlund for his feedback on Section 3.2.

8.  Informative References

   [I-D.dt-rmcat-feedback-message]
              Sarker, Z., Perkins, D., Singh, V., and D. Ramalho, "RTP
              Control Protocol (RTCP) Feedback for Congestion Control",
              draft-dt-rmcat-feedback-message-00 (work in progress),
              July 2016.






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   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, Q., and D. 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 D. 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, D., 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>.

   [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>.

   [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>.







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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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