Internet DRAFT - draft-perkins-rmcat-rtp-cc-feedback
draft-perkins-rmcat-rtp-cc-feedback
Network Working Group C. S. Perkins
Internet-Draft University of Glasgow
Intended status: Informational July 02, 2014
Expires: January 03, 2015
Using RTP Control Protocol (RTCP) Feedback for Unicast Multimedia
Congestion Control
draft-perkins-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
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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 . . . . . . . . . . . . . . . . . 6
5. Security Considerations . . . . . . . . . . . . . . . . . . . 7
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
8. Informative References . . . . . . . . . . . . . . . . . . . 7
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 8
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 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
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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. 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. Further, assume each SSRC sends RTCP reports for all
other SSRCs in the session (i.e., the optimisations in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving
the worst case for the RTCP overhead). When all members are senders
like this, the RTCP timing rules in Sections 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,
and on the size of source description (RTCP SDES) information sent.
As a baseline, each RTCP packet will be a compound RTCP packet that
contains an RTCP SR and an RTCP SDES packet. In the scenario above,
each RTCP SR packet will contain three report blocks, once for each
of the other RTP SSRCs sending data, for a total of 100 octets (this
is 8 octets header, 20 octets sender info, and 3 * 24 octets report
blocks). The RTCP SDES packet will comprise a header (4 octets), an
originating SSRC (4 octets), a CNAME chunk, and padding. If the
CNAME follows [RFC7022] and [I-D.ietf-rtcweb-rtp-usage] it will be 19
octets in size, and require 1 octet of padding. The resulting
compound RTCP packet will be 128 octets in size. If 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), the avg_rtcp_size
will therefore be 170 octets, including the header overhead. The
value n is this scenario is 4, and the rtcp_bw is assumed to be 5% of
the session bandwidth.
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 invert the above rtcp_interval calculation to get an
rtcp_bw that gives an interval of 1/30th of a second or lower. This
corresponds to an rtcp_bw of 20400 octets per second (since 1/30 =
170 * 4 / 20400). This is 163200 bits per second, which if 5% of the
session bandwidth, gives a session bandwidth of approximately 3.3Mbps
(i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a
total data rate of approximately 3.4Mbps). That is, RTCP can report
on every frame of video provided the session bandwidth is 3.3Mbps or
larger, when every SSRC sends a report for every video frame (due to
randomisation inherent in the RTCP timing rules, the actual RTCP
transmission intervals will be within the range [0.0135, 0.0406]s,
but will maintain an average RTCP transmission interval of 0.033s).
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
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capable of fully utilizing the configured 5% RTCP bandwidth
fraction.
If additional feedback beyond the standard report block is needed,
the session bandwidth needed will increase slightly. For example,
with an additional 20 octets data being reported in each RTCP packet,
the session bandwidth needed increases to 3.5Mbps for every SSRC to
be able to report on every frame.
The above calculations highlight the baseline feasibility of RTCP
congestion control feedback, but might not be the most appropriate
usage of the RTCP bandwidth of all applications. Depending on needs,
a less frequent usage of regular RTCP compound packets, controlled by
T_rr_interval combined with using the reduced size RTCP packets, can
achieve more frequent and useful reporting. Also the optimisations
defined in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] will
reduce the amount of bandwidth consumed for reporting when each
endpoint has multiple SSRCs.
It might also seem unnecessary to assign the same fraction of the
RTCP bandwidth to reporting on the audio and video, since video is
much higher rate, and so is presumably more likely to cause
congestion. Sending audio and video in separate RTCP sessions with
their own RTCP bandwidth fraction would give essentially double the
RTCP bandwidth for each video source, since RTCP bandwidth fraction
would be shared between two reporting SSRCs, rather than between the
four reporting SSRCs in the single session case. This would hence
reduce the session bandwidth needed to allow reports on every frame.
Extensions to split RTCP bandwidth unequally between participants in
a single session could be defined to allow this to work with a single
RTP session on a single UDP port, or two standard RTP sessions could
be run on a single port, using a demultiplexing shim. RTCP already
allows for different bandwidth fractions between senders and
receivers, so this is a relatively small change to the protocol.
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.
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
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the number of session participants, the RTCP bandwidth fraction, and
whether audio and video are sent in one or two RTP sessions). 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.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work
in progress), February 2014.
[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-15 (work in progress), May
2014.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, 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.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013.
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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