Sending RTP Control Protocol (RTCP) Feedback for Congestion Control in Interactive Multimedia Conferences
draft-ietf-rmcat-rtp-cc-feedback-08
| Document | Type | Active Internet-Draft (rmcat WG) | |
|---|---|---|---|
| Author | Colin Perkins | ||
| Last updated | 2022-02-27 (Latest revision 2021-12-28) | ||
| Replaces | draft-perkins-rmcat-rtp-cc-feedback | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | Informational | ||
| Formats | plain text htmlized pdfized bibtex | ||
| Stream | WG state | Waiting for WG Chair Go-Ahead | |
| Document shepherd | Anna Brunstrom | ||
| IESG | IESG state | I-D Exists | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | anna.brunstrom@kau.se |
draft-ietf-rmcat-rtp-cc-feedback-08
Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Informational December 28, 2021
Expires: July 1, 2022
Sending RTP Control Protocol (RTCP) Feedback for Congestion Control in
Interactive Multimedia Conferences
draft-ietf-rmcat-rtp-cc-feedback-08
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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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on July 1, 2022.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Possible Models for RTCP Feedback . . . . . . . . . . . . . . 3
3. What Feedback is Achievable With RTCP? . . . . . . . . . . . 5
3.1. Scenario 1: Voice Telephony . . . . . . . . . . . . . . . 5
3.2. Scenario 2: Point-to-Point Video Conference . . . . . . . 8
4. Discussion and Conclusions . . . . . . . . . . . . . . . . . 11
5. Security Considerations . . . . . . . . . . . . . . . . . . . 12
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 12
8. Informative References . . . . . . . . . . . . . . . . . . . 12
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15
1. Introduction
The deployment of WebRTC systems [RFC8825] has resulted in high-
quality video conferencing seeing extremely wide use. To ensure the
stability of the network in the face of this use, WebRTC systems need
to use some form of congestion control for their RTP-based media
traffic [RFC2914], [RFC8085], [RFC8083], [RFC8834], allowing them to
adapt and adjust the media data they send to match changes in the
available network capacity. In addition to ensuring the stable
operation of the network, such adaptation is critical to ensuring a
good user experience, since it allows the sender to match the media
to the network capacity, rather than forcing the receiver to
compensate for uncontrolled packet loss when the available capacity
is exceeded.
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 unicast 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 [RFC8834] systems.
Nothing in this memo is specific to the secure version of the
profile, or to WebRTC, however. It is also assumed that the
congestion control feedback mechanism described in [RFC8888], and
common RTCP extensions for efficient feedback [RFC5506], [RFC8108],
[RFC8861], [RFC8861], and [RFC8872] are available.
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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?
Widely used transport protocols, such as TCP, send acknowledgements
frequently. For example, a TCP receiver will send an acknowledgement
at least once every 0.5 seconds or when new data equal to twice the
maximum segment size has been received [I-D.ietf-tcpm-rfc793bis]).
That has relatively 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.
Frequent acknowledgements can become a problem, however, when there
is no return traffic on which to piggyback feedback, or if separate
feedback and data packets are sent and the feedback is similar in
size to the data being acknowledged. This can be the case for some
forms of media traffic, especially for voice over IP flows, leading
to high overhead when using a transport protocol that sends frequent
feedback. Approaches like in-network filtering of acknowledgements
can reduce the feedback frequency and overhead in some cases, but
this so-called "stretch-ACK" behaviour is non-standard and not
guaranteed.
Accordingly, when implementing congestion control for RTP-based
multimedia traffic, it might make sense to give the option of sending
congestion feedback less often than does TCP. 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
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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).
Reducing the feedback frequency compared to TCP will reduce feedback
overhead but will lead multimedia flows to adapt to congestion more
slowly than TCP, raising concerns about inter-flow fairness. Similar
concerns are noted in [RFC5348], and accordingly the congestion
control algorithm described therein aims for "reasonable" fairness
and a sending rate that is "generally within a factor of two" of that
TCP would achieve under the same conditions. It is to be noted,
however, that TCP exhibits inter-flow unfairness when flows with
differing round-trip times compete, and stretch acknowledgements due
to in-network traffic manipulation are not uncommon and also raise
fairness concerns. Implementations need to balance potential
unfairness against feedback overhead.
Generating and processing feedback consumes resources at the sender
and receiver. The feedback packets also incur forwarding costs,
contribute to link utilization, and can affect the timing of other
traffic on the network. This can affect performance on some types of
network, that can be impacted by the rate, timing, and size of
feedback packets, as well as by the overall volume of feedback bytes.
The amount of overhead due to congestion control feedback that is
considered acceptable has to be determined. RTCP feedback 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? Is there a desire to provide
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
rate but simple feedback (e.g., like TCP acknowledgements), or is it
acceptable to send more complex feedback less often? Is it useful
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for the congestion control to receive frequent feedback, perhaps to
provide more accurate round-trip time estimates, or to provide
robustness in case feedback packets are lost, even if the media
sending rate cannot quickly be changed? Or is low-rate feedback,
resulting in slowly responsive changes the sending rate, acceptable?
Different combinations of congestion control algorithm and media
codec might require different trade-offs, and the correct trade-off
for interactive, self-paced, real-time multimedia traffic might not
be the same as that for TCP congestion control.
3. What Feedback is Achievable With RTCP?
The following sections illustrate how the RTCP congestion control
feedback report [RFC8888] can be used in different scenarios, and
illustrate the overheads of this approach.
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 [RFC5506]. A compound RTCP packet is
sent once for every Nnc non-compound RTCP packets.
Compound RTCP packets contain a Sender Report (SR) packet, a Source
Description (SDES) packet, and an RTP Congestion Control Feedback
(CCFB) packet [RFC8888]. Non-compound RTCP packets contain only the
CCFB packet. Since each participant sends only a single RTP media
stream, the extensions for RTCP report aggregation [RFC8108] and
reporting group optimisation [RFC8861] 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
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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 CCFB packets contains an RTCP header
and SSRC (8 octets), a report timestamp (4 octets), the SSRC,
beginning and ending sequence numbers (8 octets), and 2*Nr octets of
reports, for a total of 20 + 2*Nr octets. The compound Secure RTCP
packet will include 4 octets of trailer followed by an 80 bit (10
octet) authentication tag if HMAC-SHA1 authentication is used. If
IPv4 is used, with no IP options, the UDP/IP header will be 28 octets
in size. This gives a total compound RTCP packet size of Sc = 142 +
2*Nr octets.
The non-compound RTCP packets will comprise just the CCFB packet,
SRTCP trailer and authentication tag, and a UDP/IP header. It can be
seen that these packets will be Snc = 62 + 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 in the RTP session
(in this scenario, n = 2).
To ensure an RTCP report containing congestion control feedback is
sent after every Nr frames of audio, 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 this to give 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 gives an RTCP
feedback bandwidth, Brtcp = 57kbps. Increasing the frame duration,
or reducing the frequency of reports, will reduce the RTCP bandwidth
as shown in Table 1.
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+--------------+-------------+----------------+
| Tf (seconds) | Nr (frames) | rtcp_bw (kbps) |
+--------------+-------------+----------------+
| 0.020 | 2 | 57.0 |
| 0.020 | 4 | 29.3 |
| 0.020 | 8 | 15.4 |
| 0.020 | 16 | 8.5 |
| 0.060 | 2 | 19.0 |
| 0.060 | 4 | 9.8 |
| 0.060 | 8 | 5.1 |
| 0.060 | 16 | 2.8 |
+--------------+-------------+----------------+
Table 1: RTCP bandwidth needed for VoIP feedback
The final row of Table 1 (60ms frames, report every 16 frames) sends
RTCP reports once per second, giving an RTCP bandwidth overhead of
2.8kbps.
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 the
results shown in Table 2.
+--------------+-------------+----------------+
| Tf (seconds) | Nr (frames) | rtcp_bw (kbps) |
+--------------+-------------+----------------+
| 0.020 | 2 | 41.4 |
| 0.020 | 4 | 21.5 |
| 0.020 | 8 | 11.5 |
| 0.020 | 16 | 6.5 |
| 0.060 | 2 | 13.8 |
| 0.060 | 4 | 7.2 |
| 0.060 | 8 | 3.8 |
| 0.060 | 16 | 2.2 |
+--------------+-------------+----------------+
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 2.2kbps. 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
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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, but this 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 the co-located audio and video
SSRCs at each end point are aggregated [RFC8108], the optimisations
in [RFC8861] are used, and RTCP congestion control feedback is sent
[RFC8888].
When all members are senders, the RTCP reporting interval calculation
in Section 6.2 and 6.3 of [RFC3550] and [RFC4585] reduces to:
Trtcp = n * Srtcp / Brtcp
where n is the number of members in the session, Srtcp is the average
RTCP packet size in octets, and the Brtcp is the RTCP bandwidth in
octets per second.
The average RTCP packet size, Srtcp, depends on the amount of
feedback 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.
When the RTCP reporting group extensions are used, one of these SSRCs
will be a reporting SSRC, to which the other SSRC will have delegated
its reports. No non-compound RTCP packets are sent.
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 [RFC8834] it will be 18 octets in
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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
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 [RFC8834] 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 congestion control
feedback (CCFB) report comprises an 8 octet RTCP header and SSRC, a 4
octet report timestamp, and 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 + 4 + 8
+ (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 SRTCP trailer and
authentication tag, and a UDP/IPv4 header. The size of this RTCP
packet is therefore: 262 + (2 * Nv) + (2 * Na) octets. Since the
aggregate RTCP packet contains reports from two SSRCs, the RTCP
packet size is halved before use [RFC8108]. Accordingly, the size of
the RTCP packets is:
Srtcp = (262 + (2 * Nv) + (2 * Na)) / 2
How many RTP packets does the RTCP XR congestion control feedback
packet included in these compound RTCP packets report on? That is,
what are the values of Nv and Na? This depends on the RTCP reporting
interval, Trtcp, the video bit rate and frame rate, Rf, the audio bit
rate and framing interval, and whether the receiver chooses to send
congestion control feedback in each RTCP packet it sends.
To simplify the calculation, assume it is desired to send one RTCP
report for each frame of video received (i.e., Trtcp = 1 / Rf) and to
include a congestion control feedback packet in each report. Assume
that video has constant bit rate and frame rate, and that each frame
of packet has to fit into a 1500 octet MTU. Further, assume that the
audio takes negligible bandwidth, and that the audio framing interval
can be varied within reasonable bounds, so that an integral number of
audio frames align with video frame boundaries.
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Table 3 shows the resulting values of Nv and Na, the number of video
and audio packets covered by each congestion control feedback report,
for a range of data rates and video frame rates, assuming congestion
control feedback is sent once per video frame. The table also shows
the result of inverting the RTCP reporting interval calculation to
find the corresponding RTCP bandwidth, Brtcp. The RTCP bandwidth is
given in kbps and as a fraction of the data rate.
It can be seen that, for example, with a date rate of 1024 kbps and
video sent at 30 frames-per-second, the RTCP congestion control
feedback report sent for each video frame will include reports on 3
video packets and 2 audio packets. The RTCP bandwidth needed to
sustain this reporting rate is 127.5kbps (12% of the data rate).
This assumes an audio framing interval of 16.67ms, so that two audio
packets are sent for each video frame.
+---------+----------+--------------+--------------+----------------+
| Data | Video | Video | Audio | Required RTCP |
| Rate | Frame | Packets per | Packets per | bandwidth: |
| (kbps) | Rate: Rf | Report: Nv | Report: Na | Brtcp (kbps) |
+---------+----------+--------------+--------------+----------------+
| 100 | 8 | 1 | 6 | 34.5 (34%) |
| 200 | 16 | 1 | 3 | 67.5 (33%) |
| 350 | 30 | 1 | 2 | 125.6 (35%) |
| 700 | 30 | 2 | 2 | 126.6 (18%) |
| 700 | 60 | 1 | 1 | 249.4 (35%) |
| 1024 | 30 | 3 | 2 | 127.5 (12%) |
| 1400 | 60 | 2 | 1 | 251.2 (17%) |
| 2048 | 30 | 6 | 2 | 130.3 ( 6%) |
| 2048 | 60 | 3 | 1 | 253.1 (12%) |
| 4096 | 30 | 12 | 2 | 135.9 ( 3%) |
| 4096 | 60 | 6 | 1 | 258.8 ( 6%) |
+---------+----------+--------------+--------------+----------------+
Table 3: Required RTCP bandwidth, reporting on every frame
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 the SRTCP trailer and authentication tag, and a UDP/IP
header. That is, the size of the non-compound packets would be (110
+ 2*Nv + 2*Na)/2 octets. Repeating the analysis above, but
alternating compound and non-compound reports gives results as shown
in Table 4.
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+---------+----------+--------------+--------------+----------------+
| Data | Video | Video | Audio | Required RTCP |
| Rate | Frame | Packets per | Packets per | bandwidth: |
| (kbps) | Rate: Rf | Report: Nv | Report: Na | Brtcp (kbps) |
+---------+----------+--------------+--------------+----------------+
| 100 | 8 | 1 | 6 | 24.1 (24%) |
| 200 | 16 | 1 | 3 | 46.8 (23%) |
| 350 | 30 | 1 | 2 | 86.7 (24%) |
| 700 | 30 | 2 | 2 | 87.7 (12%) |
| 700 | 60 | 1 | 1 | 171.6 (24%) |
| 1024 | 30 | 3 | 2 | 88.6 ( 8%) |
| 1400 | 60 | 2 | 1 | 173.4 (12%) |
| 2048 | 30 | 6 | 2 | 91.4 ( 4%) |
| 2048 | 60 | 3 | 1 | 175.3 ( 8%) |
| 4096 | 30 | 12 | 2 | 97.0 ( 2%) |
| 4096 | 60 | 6 | 1 | 180.9 ( 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.
4. Discussion and Conclusions
Practical systems will generally send some non-media traffic on the
same path as the media traffic. This can include STUN/TURN packets
to keep-alive NAT bindings [RFC8445], WebRTC Data Channel packets
[RFC8831], etc. Such traffic also needs congestion control, but the
means by which this is achieved is out of scope of this memo.
RTCP as it is currently specified cannot be used to send per-packet
congestion feedback with reasonable overhead.
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.
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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. The provided
feedback will be more detailed than just an acknowledgement, however,
and will provide a loss bitmap, relative arrival time, and received
ECN marks, for each packet sent. This will allow congestion control
that is effective, if slowly responsive, to be implemented (there is
guidance on providing effective congestion control in Section 3.1 of
[RFC8085]).
The format described in [RFC8888] 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.
The use of header compression [RFC2508], [RFC3545], [RFC5795] can
also be beneficial.
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
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, Ingemar Johansson, Gorry Fairhurst, and
the members of the RMCAT feedback design team for their feedback.
8. Informative References
[I-D.ietf-tcpm-rfc793bis]
Eddy, W., "Transmission Control Protocol (TCP)
Specification", draft-ietf-tcpm-rfc793bis-20 (work in
progress), January 2021.
Perkins Expires July 1, 2022 [Page 12]
Internet-Draft RTCP Feedback for Congestion Control December 2021
[RFC2508] Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP
Headers for Low-Speed Serial Links", RFC 2508,
DOI 10.17487/RFC2508, February 1999,
<https://www.rfc-editor.org/info/rfc2508>.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41,
RFC 2914, DOI 10.17487/RFC2914, September 2000,
<https://www.rfc-editor.org/info/rfc2914>.
[RFC3545] Koren, T., Casner, S., Geevarghese, J., Thompson, B., and
P. Ruddy, "Enhanced Compressed RTP (CRTP) for Links with
High Delay, Packet Loss and Reordering", RFC 3545,
DOI 10.17487/RFC3545, July 2003,
<https://www.rfc-editor.org/info/rfc3545>.
[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, <https://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,
<https://www.rfc-editor.org/info/rfc3611>.
[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,
<https://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,
<https://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, <https://www.rfc-editor.org/info/rfc5124>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008,
<https://www.rfc-editor.org/info/rfc5348>.
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[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, <https://www.rfc-editor.org/info/rfc5506>.
[RFC5795] Sandlund, K., Pelletier, G., and L-E. Jonsson, "The RObust
Header Compression (ROHC) Framework", RFC 5795,
DOI 10.17487/RFC5795, March 2010,
<https://www.rfc-editor.org/info/rfc5795>.
[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, <https://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,
<https://www.rfc-editor.org/info/rfc7201>.
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/info/rfc8083>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
[RFC8831] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
<https://www.rfc-editor.org/info/rfc8831>.
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[RFC8834] Perkins, C., Westerlund, M., and J. Ott, "Media Transport
and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
January 2021, <https://www.rfc-editor.org/info/rfc8834>.
[RFC8861] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session:
Grouping RTP Control Protocol (RTCP) Reception Statistics
and Other Feedback", RFC 8861, DOI 10.17487/RFC8861,
January 2021, <https://www.rfc-editor.org/info/rfc8861>.
[RFC8872] Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
and R. Even, "Guidelines for Using the Multiplexing
Features of RTP to Support Multiple Media Streams",
RFC 8872, DOI 10.17487/RFC8872, January 2021,
<https://www.rfc-editor.org/info/rfc8872>.
[RFC8888] Sarker, Z., Perkins, C., Singh, V., and M. Ramalho, "RTP
Control Protocol (RTCP) Feedback for Congestion Control",
RFC 8888, DOI 10.17487/RFC8888, January 2021,
<https://www.rfc-editor.org/info/rfc8888>.
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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