Internet Engineering Task Force (IETF) Z. Sarker
Request for Comments: 8888 Ericsson AB
Category: Standards Track C. Perkins
ISSN: 2070-1721 University of Glasgow
V. Singh
callstats.io
M. Ramalho
AcousticComms
January 2021
RTP Control Protocol (RTCP) Feedback for Congestion Control
Abstract
An effective RTP congestion control algorithm requires more fine-
grained feedback on packet loss, timing, and Explicit Congestion
Notification (ECN) marks than is provided by the standard RTP Control
Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets.
This document describes an RTCP feedback message intended to enable
congestion control for interactive real-time traffic using RTP. The
feedback message is designed for use with a sender-based congestion
control algorithm, in which the receiver of an RTP flow sends back to
the sender RTCP feedback packets containing the information the
sender needs to perform congestion control.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8888.
Copyright Notice
Copyright (c) 2021 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.
Table of Contents
1. Introduction
2. Terminology
3. RTCP Feedback for Congestion Control
3.1. RTCP Congestion Control Feedback Report
4. Feedback Frequency and Overhead
5. Response to Loss of Feedback Packets
6. SDP Signaling
7. Relationship to RFC 6679
8. Design Rationale
9. IANA Considerations
10. Security Considerations
11. References
11.1. Normative References
11.2. Informative References
Acknowledgements
Authors' Addresses
1. Introduction
For interactive real-time traffic, such as video conferencing flows,
the typical protocol choice is the Real-time Transport Protocol (RTP)
[RFC3550] running over the User Datagram Protocol (UDP). RTP does
not provide any guarantee of Quality of Service (QoS), reliability,
or timely delivery, and expects the underlying transport protocol to
do so. UDP alone certainly does not meet that expectation. However,
the RTP Control Protocol (RTCP) [RFC3550] provides a mechanism by
which the receiver of an RTP flow can periodically send transport and
media quality metrics to the sender of that RTP flow. This
information can be used by the sender to perform congestion control.
In the absence of standardized messages for this purpose, designers
of congestion control algorithms have developed proprietary RTCP
messages that convey only those parameters needed for their
respective designs. As a direct result, the different congestion
control designs are not interoperable. To enable algorithm evolution
as well as interoperability across designs (e.g., different rate
adaptation algorithms), it is highly desirable to have a generic
congestion control feedback format.
To help achieve interoperability for unicast RTP congestion control,
this memo specifies a common RTCP feedback packet format that can be
used by Network-Assisted Dynamic Adaptation (NADA) [RFC8698], Self-
Clocked Rate Adaptation for Multimedia (SCReAM) [RFC8298], Google
Congestion Control [Google-GCC], and Shared Bottleneck Detection
[RFC8382], and, hopefully, also by future RTP congestion control
algorithms.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
In addition, the terminology defined in [RFC3550], [RFC4585], and
[RFC5506] applies.
3. RTCP Feedback for Congestion Control
Based on an analysis of NADA [RFC8698], SCReAM [RFC8298], Google
Congestion Control [Google-GCC], and Shared Bottleneck Detection
[RFC8382], the following per-RTP packet congestion control feedback
information has been determined to be necessary:
RTP Sequence Number: The receiver of an RTP flow needs to feed the
sequence numbers of the received RTP packets back to the sender,
so the sender can determine which packets were received and which
were lost. Packet loss is used as an indication of congestion by
many congestion control algorithms.
Packet Arrival Time: The receiver of an RTP flow needs to feed the
arrival time of each RTP packet back to the sender. Packet delay
and/or delay variation (jitter) is used as a congestion signal by
some congestion control algorithms.
Packet Explicit Congestion Notification (ECN) Marking: If ECN
[RFC3168] [RFC6679] is used, it is necessary to feed back the
2-bit ECN mark in received RTP packets, indicating for each RTP
packet whether it is marked not-ECT, ECT(0), ECT(1), or ECN
Congestion Experienced (ECN-CE). ("ECT" stands for "ECN-Capable
Transport".) If the path used by the RTP traffic is ECN capable,
the sender can use ECN-CE marking information as a congestion
control signal.
Every RTP flow is identified by its Synchronization Source (SSRC)
identifier. Accordingly, the RTCP feedback format needs to group its
reports by SSRC, sending one report block per received SSRC.
As a practical matter, we note that host operating system (OS)
process interruptions can occur at inopportune times. Accordingly,
recording RTP packet send times at the sender, and the corresponding
RTP packet arrival times at the receiver, needs to be done with
deliberate care. This is because the time duration of host OS
interruptions can be significant relative to the precision desired in
the one-way delay estimates. Specifically, the send time needs to be
recorded at the last opportunity prior to transmitting the RTP packet
at the sender, and the arrival time at the receiver needs to be
recorded at the earliest available opportunity.
3.1. RTCP Congestion Control Feedback Report
Congestion control feedback can be sent as part of a regular
scheduled RTCP report or in an RTP/AVPF early feedback packet. If
sent as early feedback, congestion control feedback MAY be sent in a
non-compound RTCP packet [RFC5506] if the RTP/AVPF profile [RFC4585]
or the RTP/SAVPF profile [RFC5124] is used.
Irrespective of how it is transported, the congestion control
feedback is sent as a Transport-Layer Feedback Message (RTCP packet
type 205). The format of this RTCP packet is shown in Figure 1:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| FMT=11 | PT = 205 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of RTCP packet sender |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of 1st RTP Stream |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | num_reports |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|R|ECN| Arrival time offset | ... .
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
. .
. .
. .
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of nth RTP Stream |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | num_reports |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|R|ECN| Arrival time offset | ... |
. .
. .
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Report Timestamp (32 bits) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: RTCP Congestion Control Feedback Packet Format
The first 8 octets comprise a standard RTCP header, with PT=205 and
FMT=11 indicating that this is a congestion control feedback packet,
and with the SSRC set to that of the sender of the RTCP packet.
Section 6.1 of [RFC4585] requires the RTCP header to be followed by
the SSRC of the RTP flow being reported upon. Accordingly, the RTCP
header is followed by a report block for each SSRC from which RTP
packets have been received, followed by a Report Timestamp.
Each report block begins with the SSRC of the received RTP stream on
which it is reporting. Following this, the report block contains a
16-bit packet metric block for each RTP packet with that has a sequence
number in the range begin_seq to begin_seq+num_reports inclusive
(calculated using arithmetic modulo 65536 to account for possible
sequence number wrap-around). If the number of 16-bit packet metric
blocks included in the report block is not a multiple of two, then 16
bits of zero padding MUST be added after the last packet metric
block, to align the end of the packet metric blocks with the next
32-bit boundary. The value of num_reports MAY be 0, indicating that
there are no packet metric blocks included for that SSRC. Each
report block MUST NOT include more than 16384 packet metric blocks
(i.e., it MUST NOT report on more than one quarter of the sequence
number space in a single report).
The contents of each 16-bit packet metric block comprise the R, ECN,
and ATO fields as follows:
Received (R, 1 bit): A boolean that indicates whether the packet was
received. 0 indicates that the packet was not yet received and
the subsequent 15 bits (ECN and ATO) in this 16-bit packet metric
block are also set to 0 and MUST be ignored. 1 indicates that the
packet was received and the subsequent bits in the block need to
be parsed.
ECN (2 bits): The echoed ECN mark of the packet. These bits are set
to 00 if not received or if ECN is not used.
Arrival time offset (ATO, 13 bits): The arrival time of the RTP
packet at the receiver, as an offset before the time represented
by the Report Timestamp (RTS) field of this RTCP congestion
control feedback report. The ATO field is in units of 1/1024
seconds (this unit is chosen to give exact offsets from the RTS
field) so, for example, an ATO value of 512 indicates that the
corresponding RTP packet arrived exactly half a second before the
time instant represented by the RTS field. If the measured value
is greater than 8189/1024 seconds (the value that would be coded
as 0x1FFD), the value 0x1FFE MUST be reported to indicate an over-
range measurement. If the measurement is unavailable or if the
arrival time of the RTP packet is after the time represented by
the RTS field, then an ATO value of 0x1FFF MUST be reported for
the packet.
The RTCP congestion control feedback report packet concludes with the
Report Timestamp field (RTS, 32 bits). This denotes the time instant
on which this packet is reporting and is the instant from which the
arrival time offset values are calculated. The value of the RTS
field is derived from the same clock used to generate the NTP
timestamp field in RTCP Sender Report (SR) packets. It is formatted
as the middle 32 bits of an NTP format timestamp, as described in
Section 4 of [RFC3550].
RTCP Congestion Control Feedback Packets SHOULD include a report
block for every active SSRC. The sequence number ranges reported on
in consecutive reports for a given SSRC will generally be contiguous,
but overlapping reports MAY be sent (and need to be sent in cases
where RTP packet reordering occurs across the boundary between
consecutive reports). If an RTP packet was reported as received in
one report, that packet MUST also be reported as received in any
overlapping reports sent later that cover its sequence number range.
If feedback reports covering overlapping sequence number ranges are
sent, information in later feedback reports may update any data sent
in previous reports for RTP packets included in both feedback
reports.
RTCP Congestion Control Feedback Packets can be large if they are
sent infrequently relative to the number of RTP data packets. If an
RTCP Congestion Control Feedback Packet is too large to fit within
the path MTU, its sender SHOULD split it into multiple feedback
packets. The RTCP reporting interval SHOULD be chosen such that
feedback packets are sent often enough that they are small enough to
fit within the path MTU. ([RTCP-Multimedia-Feedback] discusses how
to choose the reporting interval; specifications for RTP congestion
control algorithms can also provide guidance.)
If duplicate copies of a particular RTP packet are received, then the
arrival time of the first copy to arrive MUST be reported. If any of
the copies of the duplicated packet are ECN-CE marked, then an ECN-CE
mark MUST be reported for that packet; otherwise, the ECN mark of the
first copy to arrive is reported.
If no packets are received from an SSRC in a reporting interval, a
report block MAY be sent with begin_seq set to the highest sequence
number previously received from that SSRC and num_reports set to 0
(or the report can simply be omitted). The corresponding Sender
Report / Receiver Report (SR/RR) packet will have a non-increased
extended highest sequence number received field that will inform the
sender that no packets have been received, but it can ease processing
to have that information available in the congestion control feedback
reports too.
A report block indicating that certain RTP packets were lost is not
to be interpreted as a request to retransmit the lost packets. The
receiver of such a report might choose to retransmit such packets,
provided a retransmission payload format has been negotiated, but
there is no requirement that it do so.
4. Feedback Frequency and Overhead
There is a trade-off between speed and accuracy of reporting, and the
overhead of the reports. [RTCP-Multimedia-Feedback] discusses this
trade-off, suggests desirable RTCP feedback rates, and provides
guidance on how to configure, for example, the RTCP bandwidth
fraction to make appropriate use of the reporting block described in
this memo. Specifications for RTP congestion control algorithms can
also provide guidance.
It is generally understood that congestion control algorithms work
better with more frequent feedback. However, RTCP bandwidth and
transmission rules put some upper limits on how frequently the RTCP
feedback messages can be sent from an RTP receiver to the RTP sender.
In many cases, sending feedback once per frame is an upper bound
before the reporting overhead becomes excessive, although this will
depend on the media rate and more frequent feedback might be needed
with high-rate media flows [RTCP-Multimedia-Feedback]. Analysis
[feedback-requirements] has also shown that some candidate congestion
control algorithms can operate with less frequent feedback, using a
feedback interval range of 50-200 ms. Applications need to negotiate
an appropriate congestion control feedback interval at session setup
time, based on the choice of congestion control algorithm, the
expected media bitrate, and the acceptable feedback overhead.
5. Response to Loss of Feedback Packets
Like all RTCP packets, RTCP Congestion Control Feedback Packets might
be lost. All RTP congestion control algorithms MUST specify how they
respond to the loss of feedback packets.
RTCP packets do not contain a sequence number, so loss of feedback
packets has to be inferred based on the time since the last feedback
packet. If only a single congestion control feedback packet is lost,
an appropriate response is to assume that the level of congestion has
remained roughly the same as the previous report. However, if
multiple consecutive congestion control feedback packets are lost,
then the media sender SHOULD rapidly reduce its sending rate as this
likely indicates a path failure. The RTP circuit breaker
specification [RFC8083] provides further guidance.
6. SDP Signaling
A new "ack" feedback parameter, "ccfb", is defined for use with the
"a=rtcp-fb:" Session Description Protocol (SDP) extension to indicate
the use of the RTP Congestion Control Feedback Packet format defined
in Section 3. The ABNF definition [RFC5234] of this SDP parameter
extension is:
rtcp-fb-ack-param = <See Section 4.2 of [RFC4585]>
rtcp-fb-ack-param =/ ccfb-par
ccfb-par = SP "ccfb"
The payload type used with "ccfb" feedback MUST be the wildcard type
("*"). This implies that the congestion control feedback is sent for
all payload types in use in the session, including any Forward Error
Correction (FEC) and retransmission payload types. An example of the
resulting SDP attribute is:
a=rtcp-fb:* ack ccfb
The offer/answer rules for these SDP feedback parameters are
specified in Section 4.2 of the RTP/AVPF profile [RFC4585].
An SDP offer might indicate support for both the congestion control
feedback mechanism specified in this memo and one or more alternative
congestion control feedback mechanisms that offer substantially the
same semantics. In this case, the answering party SHOULD include
only one of the offered congestion control feedback mechanisms in its
answer. If a subsequent offer containing the same set of congestion
control feedback mechanisms is received, the generated answer SHOULD
choose the same congestion control feedback mechanism as in the
original answer where possible.
When the SDP BUNDLE extension [RFC8843] is used for multiplexing, the
"a=rtcp-fb:" attribute has multiplexing category IDENTICAL-PER-PT
[RFC8859].
7. Relationship to RFC 6679
The use of Explicit Congestion Notification (ECN) with RTP is
described in [RFC6679], which specifies how to negotiate the use of
ECN with RTP and defines an RTCP ECN Feedback Packet to carry ECN
feedback reports. It uses an SDP "a=ecn-capable-rtp:" attribute to
negotiate the use of ECN, and the "a=rtcp-fb:" attribute with the
"nack" parameter "ecn" to negotiate the use of RTCP ECN Feedback
Packets.
The RTCP ECN Feedback Packet is not useful when ECN is used with the
RTP Congestion Control Feedback Packet defined in this memo, since it
provides duplicate information. When congestion control feedback is
to be used with RTP and ECN, the SDP offer generated MUST include an
"a=ecn-capable-rtp:" attribute to negotiate ECN support, along with
an "a=rtcp-fb:" attribute with the "ack" parameter "ccfb" to indicate
that the RTP Congestion Control Feedback Packet can be used. The
"a=rtcp-fb:" attribute MAY also include the "nack" parameter "ecn" to
indicate that the RTCP ECN Feedback Packet is also supported. If an
SDP offer signals support for both RTP Congestion Control Feedback
Packets and the RTCP ECN Feedback Packet, the answering party SHOULD
signal support for one, but not both, formats in its SDP answer to
avoid sending duplicate feedback.
When using ECN with RTP, the guidelines in Section 7.2 of [RFC6679]
MUST be followed to initiate the use of ECN in an RTP session. The
guidelines in Section 7.3 of [RFC6679] regarding the ongoing use of
ECN within an RTP session MUST also be followed, with the exception
that feedback is sent using the RTCP Congestion Control Feedback
Packets described in this memo rather than using RTP ECN Feedback
Packets. Similarly, the guidance in Section 7.4 of [RFC6679] related
to detecting failures MUST be followed, with the exception that the
necessary information is retrieved from the RTCP Congestion Control
Feedback Packets rather than from RTP ECN Feedback Packets.
8. Design Rationale
The primary function of RTCP SR/RR packets is to report statistics on
the reception of RTP packets. The reception report blocks sent in
these packets contain information about observed jitter, fractional
packet loss, and cumulative packet loss. It was intended that this
information could be used to support congestion control algorithms,
but experience has shown that it is not sufficient for that purpose.
An efficient congestion control algorithm requires more fine-grained
information on per-packet reception quality than is provided by SR/RR
packets to react effectively. The feedback format defined in this
memo provides such fine-grained feedback.
Several other RTCP extensions also provide more detailed feedback
than SR/RR packets:
TMMBR: The codec control messages for the RTP/AVPF profile [RFC5104]
include a Temporary Maximum Media Stream Bit Rate Request (TMMBR)
message. This is used to convey a temporary maximum bitrate
limitation from a receiver of RTP packets to their sender. Even
though it was not designed to replace congestion control, TMMBR
has been used as a means to do receiver-based congestion control
where the session bandwidth is high enough to send frequent TMMBR
messages, especially when used with non-compound RTCP packets
[RFC5506]. This approach requires the receiver of the RTP packets
to monitor their reception, determine the level of congestion, and
recommend a maximum bitrate suitable for current available
bandwidth on the path; it also assumes that the RTP sender
can/will respect that bitrate. This is the opposite of the
sender-based congestion control approach suggested in this memo,
so TMMBR cannot be used to convey the information needed for
sender-based congestion control. TMMBR could, however, be viewed
as a complementary mechanism that can inform the sender of the
receiver's current view of an acceptable maximum bitrate.
Mechanisms that convey the receiver's estimate of the maximum
available bitrate provide similar feedback.
RTCP Extended Reports (XRs): Numerous RTCP XR blocks have been
defined to report details of packet loss, arrival times [RFC3611],
delay [RFC6843], and ECN marking [RFC6679]. It is possible to
combine several such XR blocks into a compound RTCP packet, to
report the detailed loss, arrival time, and ECN marking
information needed for effective sender-based congestion control.
However, the result has high overhead in terms of both bandwidth
and complexity, due to the need to stack multiple reports.
Transport-wide Congestion Control: The format defined in this memo
provides individual feedback on each SSRC. An alternative is to
add a header extension to each RTP packet, containing a single,
transport-wide, packet sequence number, then have the receiver
send RTCP reports giving feedback on these additional sequence
numbers [RTP-Ext-for-CC]. Such an approach increases the size of
each RTP packet by 8 octets, due to the header extension, but
reduces the size of the RTCP feedback packets, and can simplify
the rate calculation at the sender if it maintains a single rate
limit that applies to all RTP packets sent, irrespective of their
SSRC. Equally, the use of transport-wide feedback makes it more
difficult to adapt the sending rate, or respond to lost packets,
based on the reception and/or loss patterns observed on a per-SSRC
basis (for example, to perform differential rate control and
repair for audio and video flows, based on knowledge of what
packets from each flow were lost). Transport-wide feedback is
also a less natural fit with the wider RTP framework, which makes
extensive use of per-SSRC sequence numbers and feedback.
Considering these issues, we believe it appropriate to design a new
RTCP feedback mechanism to convey information for sender-based
congestion control algorithms. The new congestion control feedback
RTCP packet described in Section 3 provides such a mechanism.
9. IANA Considerations
The IANA has registered one new RTP/AVPF Transport-Layer Feedback
Message in the "FMT Values for RTPFB Payload Types" table [RFC4585]
as defined in Section 3.1:
Name: CCFB
Long name: RTP Congestion Control Feedback
Value: 11
Reference: RFC 8888
The IANA has also registered one new SDP "rtcp-fb" attribute "ack"
parameter, "ccfb", in the SDP '"ack" and "nack" Attribute Values'
registry:
Value name: ccfb
Long name: Congestion Control Feedback
Usable with: ack
Mux: IDENTICAL-PER-PT
Reference: RFC 8888
10. Security Considerations
The security considerations of the RTP specification [RFC3550], the
applicable RTP profile (e.g., [RFC3551], [RFC3711], or [RFC4585]),
and the RTP congestion control algorithm being used (e.g., [RFC8698],
[RFC8298], [Google-GCC], or [RFC8382]) apply.
A receiver that intentionally generates inaccurate RTCP congestion
control feedback reports might be able to trick the sender into
sending at a greater rate than the path can support, thereby causing
congestion on the path. This scenario will negatively impact the
quality of experience of that receiver, potentially causing both
denial of service to other traffic sharing the path and excessively
increased resource usage at the media sender. Since RTP is an
unreliable transport, a sender can intentionally drop a packet,
leaving a gap in the RTP sequence number space without causing
serious harm, to check that the receiver is correctly reporting
losses. (This needs to be done with care and some awareness of the
media data being sent, to limit impact on the user experience.)
An on-path attacker that can modify RTCP Congestion Control Feedback
Packets can change the reports to trick the sender into sending at
either an excessively high or excessively low rate, leading to denial
of service. The secure RTCP profile [RFC3711] can be used to
authenticate RTCP packets to protect against this attack.
An off-path attacker that can spoof RTCP Congestion Control Feedback
Packets can similarly trick a sender into sending at an incorrect
rate, leading to denial of service. This attack is difficult, since
the attacker needs to guess the SSRC and sequence number in addition
to the destination transport address. As with on-path attacks, the
secure RTCP profile [RFC3711] can be used to authenticate RTCP
packets to protect against this attack.
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>.
[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>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[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>.
[RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234,
DOI 10.17487/RFC5234, January 2008,
<https://www.rfc-editor.org/info/rfc5234>.
[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>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <https://www.rfc-editor.org/info/rfc6679>.
[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>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", RFC 8843,
DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[RFC8859] Nandakumar, S., "A Framework for Session Description
Protocol (SDP) Attributes When Multiplexing", RFC 8859,
DOI 10.17487/RFC8859, January 2021,
<https://www.rfc-editor.org/info/rfc8859>.
11.2. Informative References
[feedback-requirements]
"RMCAT Feedback Requirements", IETF 95, April 2016,
<https://www.ietf.org/proceedings/95/slides/slides-95-
rmcat-1.pdf>.
[Google-GCC]
Holmer, S., Lundin, H., Carlucci, G., De Cicco, L., and S.
Mascolo, "A Google Congestion Control Algorithm for Real-
Time Communication", Work in Progress, Internet-Draft,
draft-ietf-rmcat-gcc-02, 8 July 2016,
<https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02>.
[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>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013,
<https://www.rfc-editor.org/info/rfc6843>.
[RFC8298] Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", RFC 8298, DOI 10.17487/RFC8298, December
2017, <https://www.rfc-editor.org/info/rfc8298>.
[RFC8382] Hayes, D., Ed., Ferlin, S., Welzl, M., and K. Hiorth,
"Shared Bottleneck Detection for Coupled Congestion
Control for RTP Media", RFC 8382, DOI 10.17487/RFC8382,
June 2018, <https://www.rfc-editor.org/info/rfc8382>.
[RFC8698] Zhu, X., Pan, R., Ramalho, M., and S. Mena, "Network-
Assisted Dynamic Adaptation (NADA): A Unified Congestion
Control Scheme for Real-Time Media", RFC 8698,
DOI 10.17487/RFC8698, February 2020,
<https://www.rfc-editor.org/info/rfc8698>.
[RTCP-Multimedia-Feedback]
Perkins, C., "RTP Control Protocol (RTCP) Feedback for
Congestion Control in Interactive Multimedia Conferences",
Work in Progress, Internet-Draft, draft-ietf-rmcat-rtp-cc-
feedback-05, 4 November 2019,
<https://tools.ietf.org/html/draft-ietf-rmcat-rtp-cc-
feedback-05>.
[RTP-Ext-for-CC]
Holmer, S., Flodman, M., and E. Sprang, "RTP Extensions
for Transport-wide Congestion Control", Work in Progress,
Internet-Draft, draft-holmer-rmcat-transport-wide-cc-
extensions-01, 19 October 2015,
<https://tools.ietf.org/html/draft-holmer-rmcat-transport-
wide-cc-extensions-01>.
Acknowledgements
This document is based on the outcome of a design team discussion in
the RTP Media Congestion Avoidance Techniques (RMCAT) Working Group.
The authors would like to thank David Hayes, Stefan Holmer, Randell
Jesup, Ingemar Johansson, Jonathan Lennox, Sergio Mena, Nils
Ohlmeier, Magnus Westerlund, and Xiaoqing Zhu for their valuable
feedback.
Authors' Addresses
Zaheduzzaman Sarker
Ericsson AB
Torshamnsgatan 23
SE-164 83 Stockholm
Sweden
Phone: +46 10 717 37 43
Email: zaheduzzaman.sarker@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow
G12 8QQ
United Kingdom
Email: csp@csperkins.org
Varun Singh
CALLSTATS I/O Oy
Annankatu 31-33 C 42
FI-00100 Helsinki
Finland
Email: varun.singh@iki.fi
URI: https://www.callstats.io/
Michael A. Ramalho
AcousticComms Consulting
6310 Watercrest Way Unit 203
Lakewood Ranch, FL 34202-5122
United States of America
Phone: +1 732 832 9723
Email: mar42@cornell.edu
URI: http://ramalho.webhop.info/