|RFC 8888||Congestion Control Feedback in RTCP||January 2021|
|Sarker, et al.||Standards Track||[Page]|
- Internet Engineering Task Force (IETF)
- Standards Track
RTP Control Protocol (RTCP) Feedback for Congestion Control
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.¶
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.¶
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 Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.¶
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.¶
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.¶
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.¶
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.¶
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 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.¶
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.¶
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.¶
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¶
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.¶
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.¶
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:¶
- 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.¶
The IANA has also registered one new SDP "rtcp-fb" attribute "ack" parameter, "ccfb", in the SDP '"ack" and "nack" Attribute Values' registry:¶
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.¶
- Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <https://www.rfc-editor.org/info/rfc2119>.
- Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, DOI 10.17487/RFC3168, , <https://www.rfc-editor.org/info/rfc3168>.
- 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, , <https://www.rfc-editor.org/info/rfc3550>.
- Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, , <https://www.rfc-editor.org/info/rfc3551>.
- Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, , <https://www.rfc-editor.org/info/rfc3711>.
- 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, , <https://www.rfc-editor.org/info/rfc4585>.
- 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, , <https://www.rfc-editor.org/info/rfc5124>.
- Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, , <https://www.rfc-editor.org/info/rfc5234>.
- Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, DOI 10.17487/RFC5506, , <https://www.rfc-editor.org/info/rfc5506>.
- 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, , <https://www.rfc-editor.org/info/rfc6679>.
- Perkins, C. and V. Singh, "Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions", RFC 8083, DOI 10.17487/RFC8083, , <https://www.rfc-editor.org/info/rfc8083>.
- Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <https://www.rfc-editor.org/info/rfc8174>.
- Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", RFC 8843, DOI 10.17487/RFC8843, , <https://www.rfc-editor.org/info/rfc8843>.
- Nandakumar, S., "A Framework for Session Description Protocol (SDP) Attributes When Multiplexing", RFC 8859, DOI 10.17487/RFC8859, , <https://www.rfc-editor.org/info/rfc8859>.
- "RMCAT Feedback Requirements", IETF 95, , <https://www.ietf.org/proceedings/95/slides/slides-95-rmcat-1.pdf>.
- 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, , <https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02>.
- Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, DOI 10.17487/RFC3611, , <https://www.rfc-editor.org/info/rfc3611>.
- 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, , <https://www.rfc-editor.org/info/rfc5104>.
- 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, , <https://www.rfc-editor.org/info/rfc6843>.
- Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation for Multimedia", RFC 8298, DOI 10.17487/RFC8298, , <https://www.rfc-editor.org/info/rfc8298>.
- 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, , <https://www.rfc-editor.org/info/rfc8382>.
- 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, , <https://www.rfc-editor.org/info/rfc8698>.
- 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, , <https://tools.ietf.org/html/draft-ietf-rmcat-rtp-cc-feedback-05>.
- 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, , <https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01>.
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.¶