draft-ietf-rtcweb-rtp-usage-12.txt   draft-ietf-rtcweb-rtp-usage-13.txt 
RTCWEB Working Group C. S. Perkins RTCWEB Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund Intended status: Standards Track M. Westerlund
Expires: August 18, 2014 Ericsson Expires: October 25, 2014 Ericsson
J. Ott J. Ott
Aalto University Aalto University
February 14, 2014 April 23, 2014
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-12 draft-ietf-rtcweb-rtp-usage-13
Abstract Abstract
The Web Real-Time Communication (WebRTC) framework provides support The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text, for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework. memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features, the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported. profiles, and extensions need to be supported.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 18, 2014. This Internet-Draft will expire on October 25, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 6 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 8 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 12
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12 5.1. Conferencing Extensions and Topologies . . . . . . . . . 12
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 13 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 14
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 14
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 14
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 14 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 15
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 14 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 15
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 16
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 16
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 17
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 17
6.1. Negative Acknowledgements and RTP Retransmission . . . . 16 6.1. Negative Acknowledgements and RTP Retransmission . . . . 17
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 17 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 18
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 17 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 18 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20
7.2. RTCP Limitations for Congestion Control . . . . . . . . . 19 7.2. RTCP Limitations for Congestion Control . . . . . . . . . 20
7.3. Congestion Control Interoperability and Legacy Systems . 20 7.3. Congestion Control Interoperability and Legacy Systems . 22
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 23
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 21 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 24
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 24
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25
12. RTP Implementation Considerations . . . . . . . . . . . . . . 25 12. RTP Implementation Considerations . . . . . . . . . . . . . . 28
12.1. Configuration and Use of RTP Sessions . . . . . . . . . 25 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 28
12.1.1. Use of Multiple Media Flows Within an RTP Session . 25 12.1.1. Use of Multiple Media Sources Within an RTP Session 28
12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 26 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 29
12.1.3. Differentiated Treatment of Flows . . . . . . . . . 31 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 34
12.2. Source, Flow, and Participant Identification . . . . . . 32 12.2. Media Source, RTP Packet Streams, and Participant
12.2.1. Media Streams . . . . . . . . . . . . . . . . . . . 33 Identification . . . . . . . . . . . . . . . . . . . . . 35
12.2.2. Media Streams: SSRC Collision Detection . . . . . . 33 12.2.1. Media Source . . . . . . . . . . . . . . . . . . . . 36
12.2.3. Media Synchronisation Context . . . . . . . . . . . 34 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36
13. Security Considerations . . . . . . . . . . . . . . . . . . . 35 12.2.3. Media Synchronisation Context . . . . . . . . . . . 37
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35 13. Security Considerations . . . . . . . . . . . . . . . . . . . 38
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 36 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39
16. References . . . . . . . . . . . . . . . . . . . . . . . . . 36 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39
16.1. Normative References . . . . . . . . . . . . . . . . . . 36 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39
16.2. Informative References . . . . . . . . . . . . . . . . . 39 16.1. Normative References . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 41 16.2. Informative References . . . . . . . . . . . . . . . . . 42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real- for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol, time media applications. Previous work has defined the RTP protocol,
along with numerous profiles, payload formats, and other extensions. along with numerous profiles, payload formats, and other extensions.
When combined with appropriate signalling, these form the basis for When combined with appropriate signalling, these form the basis for
many teleconferencing systems. many teleconferencing systems.
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(Section 13) and IANA considerations (Section 14). (Section 13) and IANA considerations (Section 14).
2. Rationale 2. Rationale
The RTP framework comprises the RTP data transfer protocol, the RTP The RTP framework comprises the RTP data transfer protocol, the RTP
control protocol, and numerous RTP payload formats, profiles, and control protocol, and numerous RTP payload formats, profiles, and
extensions. This range of add-ons has allowed RTP to meet various extensions. This range of add-ons has allowed RTP to meet various
needs that were not envisaged by the original protocol designers, and needs that were not envisaged by the original protocol designers, and
to support many new media encodings, but raises the question of what to support many new media encodings, but raises the question of what
extensions are to be supported by new implementations. The extensions are to be supported by new implementations. The
development of the WebRTC framework provides an opportunity for us to development of the WebRTC framework provides an opportunity to review
review the available RTP features and extensions, and to define a the available RTP features and extensions, and to define a common
common baseline feature set for all WebRTC implementations of RTP. baseline feature set for all WebRTC implementations of RTP. This
This builds on the past 20 years development of RTP to mandate the builds on the past 20 years development of RTP to mandate the use of
use of extensions that have shown widespread utility, while still extensions that have shown widespread utility, while still remaining
remaining compatible with the wide installed base of RTP compatible with the wide installed base of RTP implementations where
implementations where possible. possible.
Other RTP and RTCP extensions not discussed in this document can be RTP and RTCP extensions that are not discussed in this document can
implemented by WebRTC end-points if they are beneficial for new use be implemented by WebRTC end-points if they are beneficial for new
cases. However, they are not necessary to address the WebRTC use use cases. However, they are not necessary to address the WebRTC use
cases and requirements identified to date cases and requirements identified in
[I-D.ietf-rtcweb-use-cases-and-requirements]. [I-D.ietf-rtcweb-use-cases-and-requirements].
While the baseline set of RTP features and extensions defined in this While the baseline set of RTP features and extensions defined in this
memo is targeted at the requirements of the WebRTC framework, it is memo is targeted at the requirements of the WebRTC framework, it is
expected to be broadly useful for other conferencing-related uses of expected to be broadly useful for other conferencing-related uses of
RTP. In particular, it is likely that this set of RTP features and RTP. In particular, it is likely that this set of RTP features and
extensions will be appropriate for other desktop or mobile video extensions will be appropriate for other desktop or mobile video
conferencing systems, or for room-based high-quality telepresence conferencing systems, or for room-based high-quality telepresence
applications. applications.
3. Terminology 3. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. The RFC document are to be interpreted as described in [RFC2119]. The RFC
2119 interpretation of these key words applies only when written in 2119 interpretation of these key words applies only when written in
ALL CAPS. Lower- or mixed-case uses of these key words are not to be ALL CAPS. Lower- or mixed-case uses of these key words are not to be
interpreted as carrying special significance in this memo. interpreted as carrying special significance in this memo.
We define the following terms: We define the following additional terms:
RTP Media Stream: A sequence of RTP packets, and associated RTCP WebRTC MediaStream: The MediaStream concept defined by the W3C in
packets, using a single synchronisation source (SSRC) that the WebRTC API [W3C.WD-mediacapture-streams-20130903].
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Session: As defined by [RFC3550], the endpoints belonging to the Transport-layer Flow: A uni-directional flow of transport packets
same RTP Session are those that share a single SSRC space. That that are identified by having a particular 5-tuple of source IP
is, those endpoints can see an SSRC identifier transmitted by any address, source port, destination IP address, destination port,
one of the other endpoints. An endpoint can see an SSRC either and transport protocol used.
directly in RTP and RTCP packets, or as a contributing source
(CSRC) in RTP packets from a mixer. The RTP Session scope is
hence decided by the endpoints' network interconnection topology,
in combination with RTP and RTCP forwarding strategies deployed by
endpoints and any interconnecting middle nodes.
WebRTC MediaStream: The MediaStream concept defined by the W3C in Bi-directional Transport-layer Flow: A bi-directional transport-
the API [W3C.WD-mediacapture-streams-20130903]. layer flow is a transport-layer flow that is symmetric. That is,
the transport-layer flow in the reverse direction has a 5-tuple
where the source and destination address and ports are swapped
compared to the forward path transport-layer flow, and the
transport protocol is the same.
Other terms are used according to their definitions from the RTP This document uses the terminology from
Specification [RFC3550]. [I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used
according to their definitions from the RTP Specification [RFC3550].
We especially note the following frequently used terms: RTP Packet
Stream, RTP Session, and End-point.
4. WebRTC Use of RTP: Core Protocols 4. WebRTC Use of RTP: Core Protocols
The following sections describe the core features of RTP and RTCP The following sections describe the core features of RTP and RTCP
that need to be implemented, along with the mandated RTP profiles. that need to be implemented, along with the mandated RTP profiles.
Also described are the core extensions providing essential features Also described are the core extensions providing essential features
that all WebRTC implementations need to implement to function that all WebRTC implementations need to implement to function
effectively on today's networks. effectively on today's networks.
4.1. RTP and RTCP 4.1. RTP and RTCP
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optimisations for multi-SSRC sessions defined in optimisations for multi-SSRC sessions defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED. [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED.
o Random choice of SSRC on joining a session; collision detection o Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also Section 4.8). and resolution for SSRC values (see also Section 4.8).
o Support for reception of RTP data packets containing CSRC lists, o Support for reception of RTP data packets containing CSRC lists,
as generated by RTP mixers, and RTCP packets relating to CSRCs. as generated by RTP mixers, and RTCP packets relating to CSRCs.
o Sending correct synchronisation information in the RTCP Sender o Sending correct synchronisation information in the RTCP Sender
Reports, to allow receivers to implement lip-sync, with support Reports, to allow receivers to implement lip-synchronisation;
for the rapid RTP synchronisation extensions (see Section 5.2.1) support for the rapid RTP synchronisation extensions (see
being RECOMMENDED. Section 5.2.1) is RECOMMENDED.
o Support for multiple synchronisation contexts. Participants that o Support for multiple synchronisation contexts. Participants that
send multiple simultaneous RTP media streams MAY do so as part of send multiple simultaneous RTP packet streams SHOULD do so as part
a single synchronisation context, using a single RTCP CNAME for of a single synchronisation context, using a single RTCP CNAME for
all streams and allowing receivers to play the streams out in a all streams and allowing receivers to play the streams out in a
synchronised manner, or they MAY use different synchronisation synchronised manner. For compatibility with potential future
contexts, and hence different RTCP CNAMEs, for some or all of the versions of this specification, or for interoperability with non-
streams. Receivers MUST support reception of multiple RTCP CNAMEs WebRTC devices through a gateway, receivers MUST support multiple
from each participant in an RTP session. See also Section 4.9. synchronisation contexts, indicated by the use of multiple RTCP
CNAMEs in an RTP session. This specification requires the usage
of a single CNAME when sending RTP Packet Streams in some
circumstances, see Section 4.9.
o Support for sending and receiving RTCP SR, RR, SDES, and BYE o Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types, with OPTIONAL support for other RTCP packet types packet types, with OPTIONAL support for other RTCP packet types
unless mandated by other parts of this specification; unless mandated by other parts of this specification;
implementations MUST ignore unknown RTCP packet types. Note that implementations MUST ignore unknown RTCP packet types. Note that
additional RTCP Packet types are needed by the RTP/SAVPF Profile additional RTCP Packet types are used by the RTP/SAVPF Profile
(Section 4.2) and the other RTCP extensions (Section 5). (Section 4.2) and the other RTCP extensions (Section 5).
o Support for multiple end-points in a single RTP session, and for o Support for multiple end-points in a single RTP session, and for
scaling the RTCP transmission interval according to the number of scaling the RTCP transmission interval according to the number of
participants in the session; support for randomised RTCP participants in the session; support for randomised RTCP
transmission intervals to avoid synchronisation of RTCP reports; transmission intervals to avoid synchronisation of RTCP reports;
support for RTCP timer reconsideration. support for RTCP timer reconsideration.
o Support for configuring the RTCP bandwidth as a fraction of the o Support for configuring the RTCP bandwidth as a fraction of the
media bandwidth, and for configuring the fraction of the RTCP media bandwidth, and for configuring the fraction of the RTCP
bandwidth allocated to senders, e.g., using the SDP "b=" line. bandwidth allocated to senders, e.g., using the SDP "b=" line
[RFC4566][RFC3556]. Support for the reduced minimum RTCP
reporting interval described in Section 6.2 of [RFC3550] is
RECOMMENDED.
It is known that a significant number of legacy RTP implementations, It is known that a significant number of legacy RTP implementations,
especially those targeted at VoIP-only systems, do not support all of especially those targeted at VoIP-only systems, do not support all of
the above features, and in some cases do not support RTCP at all. the above features, and in some cases do not support RTCP at all.
Implementers are advised to consider the requirements for graceful Implementers are advised to consider the requirements for graceful
degradation when interoperating with legacy implementations. degradation when interoperating with legacy implementations.
Other implementation considerations are discussed in Section 12. Other implementation considerations are discussed in Section 12.
4.2. Choice of the RTP Profile 4.2. Choice of the RTP Profile
The complete specification of RTP for a particular application domain The complete specification of RTP for a particular application domain
requires the choice of an RTP Profile. For WebRTC use, the Extended requires the choice of an RTP Profile. For WebRTC use, the Extended
Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as
extended by [RFC7007], MUST be implemented. This builds on the basic extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is
RTP/AVP profile [RFC3551], the RTP profile for RTCP-based feedback the combination of basic RTP/AVP profile [RFC3551], the RTP profile
(RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP) for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP
[RFC3711]. profile (RTP/SAVP) [RFC3711].
The RTCP-based feedback extensions [RFC4585] are needed for the The RTCP-based feedback extensions [RFC4585] are needed for the
improved RTCP timer model, that allows more flexible transmission of improved RTCP timer model. This allows more flexible transmission of
RTCP packets in response to events, rather than strictly according to RTCP packets in response to events, rather than strictly according to
bandwidth. This is vital for being able to report congestion events. bandwidth, and is vital for being able to report congestion signals
These extensions also allow saving RTCP bandwidth, and an endpoint as well as media events. These extensions also allow saving RTCP
will commonly only use the full RTCP bandwidth allocation if there bandwidth, and an end-point will commonly only use the full RTCP
are many events that require feedback. The timer rules are also bandwidth allocation if there are many events that require feedback.
needed to make use of the RTP conferencing extensions discussed in The timer rules are also needed to make use of the RTP conferencing
Section 5.1. extensions discussed in Section 5.1.
Note: The enhanced RTCP timer model defined in the RTP/AVPF Note: The enhanced RTCP timer model defined in the RTP/AVPF
profile is backwards compatible with legacy systems that implement profile is backwards compatible with legacy systems that implement
only the RTP/AVP or RTP/SAVP profile, given some constraints on only the RTP/AVP or RTP/SAVP profile, given some constraints on
parameter configuration such as the RTCP bandwidth value and "trr- parameter configuration such as the RTCP bandwidth value and "trr-
int" (the most important factor for interworking with RTP/(S)AVP int" (the most important factor for interworking with RTP/(S)AVP
end-points via a gateway is to set the trr-int parameter to a end-points via a gateway is to set the trr-int parameter to a
value representing 4 seconds). value representing 4 seconds).
The secure RTP (SRTP) profile [RFC3711] is needed to provide media The secure RTP (SRTP) profile extensions [RFC3711] are needed to
encryption, integrity protection, replay protection and a limited provide media encryption, integrity protection, replay protection and
form of source authentication. WebRTC implementations MUST NOT send a limited form of source authentication. WebRTC implementations MUST
packets using the basic RTP/AVP profile or the RTP/AVPF profile; they NOT send packets using the basic RTP/AVP profile or the RTP/AVPF
MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP profile; they MUST employ the full RTP/SAVPF profile to protect all
packets that are generated (i.e., implementations MUST use SRTP and RTP and RTCP packets that are generated (i.e., implementations MUST
SRTCP). The RTP/SAVPF profile MUST be configured using the cipher use SRTP and SRTCP). The RTP/SAVPF profile MUST be configured using
suites, DTLS-SRTP protection profiles, keying mechanisms, and other the cipher suites, DTLS-SRTP protection profiles, keying mechanisms,
parameters described in [I-D.ietf-rtcweb-security-arch]. and other parameters described in [I-D.ietf-rtcweb-security-arch].
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
The set of mandatory to implement codecs and RTP payload formats for The set of mandatory to implement codecs and RTP payload formats for
WebRTC is not specified in this memo, instead they are defined in WebRTC is not specified in this memo, instead they are defined in
separate specifications, such as [I-D.ietf-rtcweb-audio]. separate specifications, such as [I-D.ietf-rtcweb-audio].
Implementations can support any codec for which an RTP payload format Implementations can support any codec for which an RTP payload format
and associated signalling is defined. Implementation cannot assume and associated signalling is defined. Implementation cannot assume
that the other participants in an RTP session understand any RTP that the other participants in an RTP session understand any RTP
payload format, no matter how common; the mapping between RTP payload payload format, no matter how common; the mapping between RTP payload
type numbers and specific configurations of particular RTP payload type numbers and specific configurations of particular RTP payload
formats MUST be agreed before those payload types/formats can be formats MUST be agreed before those payload types/formats can be
used. In an SDP context, this can be done using the "a=rtpmap:" and used. In an SDP context, this can be done using the "a=rtpmap:" and
"a=fmtp:" attributes associated with an "m=" line. "a=fmtp:" attributes associated with an "m=" line, along with any
other SDP attributes needed to configure the RTP payload format.
Endpoints can signal support for multiple RTP payload formats, or End-points can signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, as long as multiple configurations of a single RTP payload format, as long as
each unique RTP payload format configuration uses a different RTP each unique RTP payload format configuration uses a different RTP
payload type number. As outlined in Section 4.8, the RTP payload payload type number. As outlined in Section 4.8, the RTP payload
type number is sometimes used to associate an RTP media stream with a type number is sometimes used to associate an RTP packet stream with
signalling context. This association is possible provided unique RTP a signalling context. This association is possible provided unique
payload type numbers are used in each context. For example, an RTP RTP payload type numbers are used in each context. For example, an
media stream can be associated with an SDP "m=" line by comparing the RTP packet stream can be associated with an SDP "m=" line by
RTP payload type numbers used by the media stream with payload types comparing the RTP payload type numbers used by the RTP packet stream
signalled in the "a=rtpmap:" lines in the media sections of the SDP. with payload types signalled in the "a=rtpmap:" lines in the media
If RTP media streams are being associated with signalling contexts sections of the SDP. If RTP packet streams are being associated with
based on the RTP payload type, then the assignment of RTP payload signalling contexts based on the RTP payload type, then the
type numbers MUST be unique across signalling contexts; if the same assignment of RTP payload type numbers MUST be unique across
RTP payload format configuration is used in multiple contexts, then a signalling contexts; if the same RTP payload format configuration is
different RTP payload type number has to be assigned in each context used in multiple contexts, then a different RTP payload type number
to ensure uniqueness. If the RTP payload type number is not being has to be assigned in each context to ensure uniqueness. If the RTP
used to associated RTP media streams with a signalling context, then payload type number is not being used to associate RTP packet streams
the same RTP payload type number can be used to indicate the exact with a signalling context, then the same RTP payload type number can
same RTP payload format configuration in multiple contexts. be used to indicate the exact same RTP payload format configuration
in multiple contexts. A single RTP payload type number MUST NOT be
assigned to different RTP payload formats, or different
configurations of the same RTP payload format, within a single RTP
session (note that the different "m=" lines in an SDP bundle group
[I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session).
An endpoint that has signalled support for multiple RTP payload An end-point that has signalled support for multiple RTP payload
formats SHOULD accept data in any of those payload formats at any formats SHOULD be able to accept data in any of those payload formats
time, unless it has previously signalled limitations on its decoding at any time, unless it has previously signalled limitations on its
capability. This requirement is constrained if several types of decoding capability. This requirement is constrained if several
media (e.g., audio and video) are sent in the same RTP session. In types of media (e.g., audio and video) are sent in the same RTP
such a case, a source (SSRC) is restricted to switching only between session. In such a case, a source (SSRC) is restricted to switching
the RTP payload formats signalled for the type of media that is being only between the RTP payload formats signalled for the type of media
sent by that source; see Section 4.4. To support rapid rate that is being sent by that source; see Section 4.4. To support rapid
adaptation by changing codec, RTP does not require advance signalling rate adaptation by changing codec, RTP does not require advance
for changes between RTP payload formats that were signalled during signalling for changes between RTP payload formats used by a single
session set-up. SSRC that were signalled during session set-up.
An RTP sender that changes between two RTP payload types that use An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in different RTP clock rates MUST follow the recommendations in
Section 4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers Section 4.1 of [RFC7160]. RTP receivers MUST follow the
MUST follow the recommendations in Section 4.3 of recommendations in Section 4.3 of [RFC7160] in order to support
[I-D.ietf-avtext-multiple-clock-rates], in order to support sources sources that switch between clock rates in an RTP session (these
that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate). where senders use only a single clock rate).
4.4. Use of RTP Sessions 4.4. Use of RTP Sessions
An association amongst a set of participants communicating using RTP An association amongst a set of end-points communicating using RTP is
is known as an RTP session. A participant can be involved in several known as an RTP session [RFC3550]. An end-point can be involved in
RTP sessions at the same time. In a multimedia session, each type of several RTP sessions at the same time. In a multimedia session, each
media has typically been carried in a separate RTP session (e.g., type of media has typically been carried in a separate RTP session
using one RTP session for the audio, and a separate RTP session using (e.g., using one RTP session for the audio, and a separate RTP
different transport addresses for the video). WebRTC implementations session using a different transport-layer flow for the video).
of RTP are REQUIRED to implement support for multimedia sessions in WebRTC implementations of RTP are REQUIRED to implement support for
this way, separating each session using different transport-layer multimedia sessions in this way, separating each session using
addresses (e.g., different UDP ports) for compatibility with legacy different transport-layer flows for compatibility with legacy
systems. systems.
In modern day networks, however, with the widespread use of network In modern day networks, however, with the widespread use of network
address/port translators (NAT/NAPT) and firewalls, it is desirable to address/port translators (NAT/NAPT) and firewalls, it is desirable to
reduce the number of transport-layer flows used by RTP applications. reduce the number of transport-layer flows used by RTP applications.
This can be done by sending all the RTP media streams in a single RTP This can be done by sending all the RTP packet streams in a single
session, which will comprise a single transport-layer flow (this will RTP session, which will comprise a single transport-layer flow (this
prevent the use of some quality-of-service mechanisms, as discussed will prevent the use of some quality-of-service mechanisms, as
in Section 12.1.3). Implementations are REQUIRED to support discussed in Section 12.1.3). Implementations are therefore also
transport of all RTP media streams, independent of media type, in a REQUIRED to support transport of all RTP packet streams, independent
single RTP session according to of media type, in a single RTP session using a single transport layer
[I-D.ietf-avtcore-multi-media-rtp-session]. If multiple types of flow, according to [I-D.ietf-avtcore-multi-media-rtp-session]. If
media are to be used in a single RTP session, all participants in multiple types of media are to be used in a single RTP session, all
that session MUST agree to this usage. In an SDP context, participants in that RTP session MUST agree to this usage. In an SDP
[I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal this. context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to
signal such a bundle of RTP packet streams forming a single RTP
session.
Further discussion about when different RTP session structures and Further discussion about the suitability of different RTP session
multiplexing methods are suitable can be found in structures and multiplexing methods to different scenarios are
[I-D.ietf-avtcore-multiplex-guidelines]. suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines].
4.5. RTP and RTCP Multiplexing 4.5. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate transport layer Historically, RTP and RTCP have been run on separate transport layer
addresses (e.g., two UDP ports for each RTP session, one port for RTP flows (e.g., two UDP ports for each RTP session, one port for RTP and
and one port for RTCP). With the increased use of Network Address/ one port for RTCP). With the increased use of Network Address/Port
Port Translation (NAPT) this has become problematic, since Translation (NAT/NAPT) this has become problematic, since maintaining
maintaining multiple NAT bindings can be costly. It also complicates multiple NAT bindings can be costly. It also complicates firewall
firewall administration, since multiple ports need to be opened to administration, since multiple ports need to be opened to allow RTP
allow RTP traffic. To reduce these costs and session set-up times, traffic. To reduce these costs and session set-up times, support for
support for multiplexing RTP data packets and RTCP control packets on multiplexing RTP data packets and RTCP control packets on a single
a single port for each RTP session is REQUIRED, as specified in transport-layer flow for each RTP session is REQUIRED, provided it is
negotiated in the signalling channel before use as specified in
[RFC5761]. For backwards compatibility, implementations are also [RFC5761]. For backwards compatibility, implementations are also
REQUIRED to support RTP and RTCP sent on separate transport-layer REQUIRED to support RTP and RTCP sent on separate transport-layer
addresses. flows.
Note that the use of RTP and RTCP multiplexed onto a single transport Note that the use of RTP and RTCP multiplexed onto a single
port ensures that there is occasional traffic sent on that port, even transport-layer flow ensures that there is occasional traffic sent on
if there is no active media traffic. This can be useful to keep NAT that port, even if there is no active media traffic. This can be
bindings alive, and is the recommend method for application level useful to keep NAT bindings alive, and is the recommend method for
keep-alives of RTP sessions [RFC6263]. application level keep-alives of RTP sessions [RFC6263].
4.6. Reduced Size RTCP 4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and [RFC3550] RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
requires that those compound packets start with an Sender Report (SR) requires that those compound packets start with an Sender Report (SR)
or Receiver Report (RR) packet. When using frequent RTCP feedback or Receiver Report (RR) packet. When using frequent RTCP feedback
messages under the RTP/AVPF Profile [RFC4585] these statistics are messages under the RTP/AVPF Profile [RFC4585] these statistics are
not needed in every packet, and unnecessarily increase the mean RTCP not needed in every packet, and unnecessarily increase the mean RTCP
packet size. This can limit the frequency at which RTCP packets can packet size. This can limit the frequency at which RTCP packets can
be sent within the RTCP bandwidth share. be sent within the RTCP bandwidth share.
To avoid this problem, [RFC5506] specifies how to reduce the mean To avoid this problem, [RFC5506] specifies how to reduce the mean
RTCP message size and allow for more frequent feedback. Frequent RTCP message size and allow for more frequent feedback. Frequent
feedback, in turn, is essential to make real-time applications feedback, in turn, is essential to make real-time applications
quickly aware of changing network conditions, and to allow them to quickly aware of changing network conditions, and to allow them to
adapt their transmission and encoding behaviour. Support for non- adapt their transmission and encoding behaviour. Support for non-
compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be
negotiated using the signalling channel before use. For backwards negotiated using the signalling channel before use. For backwards
compatibility, implementations are also REQUIRED to support the use compatibility, implementations are also REQUIRED to support the use
of compound RTCP feedback packets if the remote endpoint does not of compound RTCP feedback packets if the remote end-point does not
agree to the use of non-compound RTCP in the signalling exchange. agree to the use of non-compound RTCP in the signalling exchange.
4.7. Symmetric RTP/RTCP 4.7. Symmetric RTP/RTCP
To ease traversal of NAT and firewall devices, implementations are To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use Symmetric RTP [RFC4961]. The reasons REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason
for using symmetric RTP is primarily to avoid issues with NAT and for using symmetric RTP is primarily to avoid issues with NATs and
Firewalls by ensuring that the flow is actually bi-directional and Firewalls by ensuring that the send and receive RTP packet streams,
thus kept alive and registered as flow the intended recipient as well as RTCP, are actually bi-directional transport-layer flows.
actually wants. In addition, it saves resources, specifically ports This will keep alive the NAT and firewall pinholes, and help indicate
at the end-points, but also in the network as NAT mappings or consent that the receive direction is a transport-layer flow the
firewall state is not unnecessary bloated. Also the amount of QoS intended recipient actually wants. In addition, it saves resources,
state is reduced. specifically ports at the end-points, but also in the network as NAT
mappings or firewall state is not unnecessary bloated. The amount of
per flow QoS state kept in the network is also reduced.
4.8. Choice of RTP Synchronisation Source (SSRC) 4.8. Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP synchronisation Implementations are REQUIRED to support signalled RTP synchronisation
source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined
in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also
support the "previous-ssrc" source attribute defined in Section 6.2 support the "previous-ssrc" source attribute defined in Section 6.2
of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be
supported. supported.
Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP
session is OPTIONAL. Implementations MUST be prepared to accept RTP session is OPTIONAL. Implementations MUST be prepared to accept RTP
and RTCP packets using SSRCs that have not been explicitly signalled and RTCP packets using SSRCs that have not been explicitly signalled
ahead of time. Implementations MUST support random SSRC assignment, ahead of time. Implementations MUST support random SSRC assignment,
and MUST support SSRC collision detection and resolution, according and MUST support SSRC collision detection and resolution, according
to [RFC3550]. When using signalled SSRC values, collision detection to [RFC3550]. When using signalled SSRC values, collision detection
MUST be performed as described in Section 5 of [RFC5576]. MUST be performed as described in Section 5 of [RFC5576].
It is often desirable to associate an RTP media stream with a non-RTP It is often desirable to associate an RTP packet stream with a non-
context. For users of the WebRTC API a mapping between SSRCs and RTP context. For users of the WebRTC API a mapping between SSRCs and
MediaStreamTracks are provided per Section 11. For gateways or other MediaStreamTracks are provided per Section 11. For gateways or other
usages it is possible to associate an RTP media stream with an "m=" usages it is possible to associate an RTP packet stream with an "m="
line in a session description formatted using SDP. If SSRCs are line in a session description formatted using SDP. If SSRCs are
signalled this is straightforward (in SDP the "a=ssrc:" line will be signalled this is straightforward (in SDP the "a=ssrc:" line will be
at the media level, allowing a direct association with an "m=" line). at the media level, allowing a direct association with an "m=" line).
If SSRCs are not signalled, the RTP payload type numbers used in an If SSRCs are not signalled, the RTP payload type numbers used in an
RTP media stream are often sufficient to associate that media stream RTP packet stream are often sufficient to associate that packet
with a signalling context (e.g., if RTP payload type numbers are stream with a signalling context (e.g., if RTP payload type numbers
assigned as described in Section 4.3 of this memo, the RTP payload are assigned as described in Section 4.3 of this memo, the RTP
types used by an RTP media stream can be compared with values in SDP payload types used by an RTP packet stream can be compared with
"a=rtpmap:" lines, which are at the media level in SDP, and so map to values in SDP "a=rtpmap:" lines, which are at the media level in SDP,
an "m=" line). and so map to an "m=" line).
4.9. Generation of the RTCP Canonical Name (CNAME) 4.9. Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronisation Source identifier for an RTP end-point. While the Synchronisation Source
(SSRC) identifier for an RTP endpoint can change if a collision is (SSRC) identifier for an RTP end-point can change if a collision is
detected, or when the RTP application is restarted, its RTCP CNAME is detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged, so that RTP endpoints can be uniquely meant to stay unchanged for the duration of a RTCPeerConnection
identified and associated with their RTP media streams within a set [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely
of related RTP sessions. For proper functionality, each RTP endpoint identified and associated with their RTP packet streams within a set
needs to have at least one unique RTCP CNAME value. An endpoint MAY of related RTP sessions.
have multiple CNAMEs, as the CNAME also identifies a particular
synchronisation context, i.e. all SSRC associated with a CNAME share Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP
a common reference clock, and if an endpoint have SSRCs associated CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs
with different reference clocks it will need to use multiple CNAMEs. identify a particular synchronisation context, i.e., all SSRCs
This ought not be common, and if possible reference clocks ought to associated with a single RTCP CNAME share a common reference clock.
be mapped to each other and one chosen to be used with RTP and RTCP. If an end-point has SSRCs that are associated with several
unsynchronised reference clocks, and hence different synchronisation
contexts, it will need to use multiple RTCP CNAMEs, one for each
synchronisation context.
Taking the discussion in Section 11 into account, a WebRTC end-point
MUST NOT use more than one RTCP CNAME in the RTP sessions belonging
to single RTCPeerConnection (that is, an RTCPeerConnection forms a
synchronisation context). RTP middleboxes MAY generate RTP packet
streams associated with more than one RTCP CNAME, to allow them to
avoid having to resynchronize media from multiple different end-
points part of a multi-party RTP session.
The RTP specification [RFC3550] includes guidelines for choosing a The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME, but these are not sufficient in the presence of NAT unique RTP CNAME, but these are not sufficient in the presence of NAT
devices. In addition, long-term persistent identifiers can be devices. In addition, long-term persistent identifiers can be
problematic from a privacy viewpoint. Accordingly, support for problematic from a privacy viewpoint (Section 13). Accordingly, a
generating a short-term persistent RTCP CNAMEs following [RFC7022] is WebRTC endpoint MUST generate a new, unique, short-term persistent
RECOMMENDED. RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a
single exception; if explicitly requested at creation an
RTCPeerConnection MAY use the same CNAME as as an existing
RTCPeerConnection within their common same-origin context.
An WebRTC end-point MUST support reception of any CNAME that matches An WebRTC end-point MUST support reception of any CNAME that matches
the syntax limitations specified by the RTP specification [RFC3550] the syntax limitations specified by the RTP specification [RFC3550]
and cannot assume that any CNAME will be chosen according to the form and cannot assume that any CNAME will be chosen according to the form
suggested above. suggested above.
4.10. Handling of Leap Seconds
The guidelines regarding handling of leap seconds to limit their
impact on RTP media playout and synchronization given in [RFC7164]
SHOULD be followed.
5. WebRTC Use of RTP: Extensions 5. WebRTC Use of RTP: Extensions
There are a number of RTP extensions that are either needed to obtain There are a number of RTP extensions that are either needed to obtain
full functionality, or extremely useful to improve on the baseline full functionality, or extremely useful to improve on the baseline
performance, in the WebRTC application context. One set of these performance, in the WebRTC application context. One set of these
extensions is related to conferencing, while others are more generic extensions is related to conferencing, while others are more generic
in nature. The following subsections describe the various RTP in nature. The following subsections describe the various RTP
extensions mandated or suggested for use within the WebRTC context. extensions mandated or suggested for use within the WebRTC context.
5.1. Conferencing Extensions 5.1. Conferencing Extensions and Topologies
RTP is inherently a group communication protocol. Groups can be RTP is a protocol that inherently supports group communication.
implemented using a centralised server, multi-unicast, or using IP Groups can be implemented by having each endpoint send its RTP packet
multicast. While IP multicast is popular in IPTV systems, overlay- streams to an RTP middlebox that redistributes the traffic, by using
based topologies dominate in interactive conferencing environments. a mesh of unicast RTP packet streams between endpoints, or by using
Such overlay-based topologies typically use one or more central an IP multicast group to distribute the RTP packet streams. These
servers to connect end-points in a star or flat tree topology. These topologies can be implemented in a number of ways as discussed in
central servers can be implemented in a number of ways as discussed
in the memo on RTP Topologies
[I-D.ietf-avtcore-rtp-topologies-update]. [I-D.ietf-avtcore-rtp-topologies-update].
Not all of the possible the overlay-based topologies are suitable for While the use of IP multicast groups is popular in IPTV systems, the
use in the WebRTC environment. Specifically: topologies based on RTP middleboxes are dominant in interactive video
conferencing environments. Topologies based on a mesh of unicast
transport-layer flows to create a common RTP session have not seen
widespread deployment to date. Accordingly, WebRTC implementations
are not expected to support topologies based on IP multicast groups
or to support mesh-based topologies, such as a point-to-multipoint
mesh configured as a single RTP session (Topo-Mesh in the terminology
of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to-
multipoint mesh constructed using several RTP sessions, implemented
in the WebRTC context using independent RTCPeerConnections, can be
expected to be utilised by WebRTC applications and needs to be
supported.
o The use of video switching MCUs makes the use of RTCP for WebRTC implementations of RTP endpoints implemented according to this
congestion control and quality of service reports problematic (see memo are expected to support all the topologies described in
Section 3.6.2 of [I-D.ietf-avtcore-rtp-topologies-update]). [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
and receive unicast RTP packet streams to and from some peer device,
provided that peer can participate in performing congestion control
on the RTP packet streams. The peer device could be another RTP
endpoint, or it could be an RTP middlebox that redistributes the RTP
packet streams to other RTP endpoints. This limitation means that
some of the RTP middlebox-based topologies are not suitable for use
in the WebRTC environment. Specifically:
o The use of content modifying MCUs with RTCP termination breaks RTP o Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
loop detection, and prevents receivers from identifying active since they make the use of RTCP for congestion control and quality
senders (see section 3.8 of of service reports problematic (see Section 3.8 of
[I-D.ietf-avtcore-rtp-topologies-update]). [I-D.ietf-avtcore-rtp-topologies-update]).
Accordingly, only Point to Point (Topo-Point-to-Point), Multiple o The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer) SHOULD NOT be used because its safe use requires a congestion
topologies are needed to achieve the use-cases to be supported in control algorithm or RTP circuit breaker that handles point to
WebRTC initially. These RECOMMENDED topologies are expected to be multipoint, which has not yet been standardised.
supported by all WebRTC end-points (these topologies require no
special RTP-layer support in the end-point if the RTP features The following topology can be used, however it has some issues worth
mandated in this memo are implemented). noting:
o Content modifying MCUs with RTCP termination (Topo-RTCP-
terminating-MCU) MAY be used. Note that in this RTP Topology, RTP
loop detection and identification of active senders is the
responsibility of the WebRTC application; since the clients are
isolated from each other at the RTP layer, RTP cannot assist with
these functions (see section 3.9 of
[I-D.ietf-avtcore-rtp-topologies-update]).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
designed to be used with centralised conferencing, where an RTP designed to be used with centralised conferencing, where an RTP
middlebox (e.g., a conference bridge) receives a participant's RTP middlebox (e.g., a conference bridge) receives a participant's RTP
media streams and distributes them to the other participants. These packet streams and distributes them to the other participants. These
extensions are not necessary for interoperability; an RTP endpoint extensions are not necessary for interoperability; an RTP end-point
that does not implement these extensions will work correctly, but that does not implement these extensions will work correctly, but
might offer poor performance. Support for the listed extensions will might offer poor performance. Support for the listed extensions will
greatly improve the quality of experience and, to provide a greatly improve the quality of experience and, to provide a
reasonable baseline quality, some of these extensions are mandatory reasonable baseline quality, some of these extensions are mandatory
to be supported by WebRTC end-points. to be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile The RTCP conferencing extensions are defined in Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio- AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully AVPF [RFC5104]; they are fully usable by the Secure variant of this
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124]. profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request (FIR) 5.1.1. Full Intra Request (FIR)
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1
Codec Control Messages [RFC5104]. This message is used to make the of the Codec Control Messages [RFC5104]. It is used to make the
mixer request a new Intra picture from a participant in the session. mixer request a new Intra picture from a participant in the session.
This is used when switching between sources to ensure that the This is used when switching between sources to ensure that the
receivers can decode the video or other predictive media encoding receivers can decode the video or other predictive media encoding
with long prediction chains. WebRTC senders MUST understand and with long prediction chains. WebRTC senders MUST understand and
react to the FIR feedback message since it greatly improves the user react to FIR feedback messages they receiver, since this greatly
experience when using centralised mixer-based conferencing; support improves the user experience when using centralised mixer-based
for sending the FIR message is OPTIONAL. conferencing. Support for sending FIR messages is OPTIONAL.
5.1.2. Picture Loss Indication (PLI) 5.1.2. Picture Loss Indication (PLI)
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/ The Picture Loss Indication message is defined in Section 6.3.1 of
AVPF profile [RFC4585]. It is used by a receiver to tell the sending the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
encoder that it lost the decoder context and would like to have it sending encoder that it lost the decoder context and would like to
repaired somehow. This is semantically different from the Full Intra have it repaired somehow. This is semantically different from the
Request above as there could be multiple ways to fulfil the request. Full Intra Request above as there could be multiple ways to fulfil
WebRTC senders MUST understand and react to this feedback message as the request. WebRTC senders MUST understand and react to PLI
a loss tolerance mechanism; receivers MAY send PLI messages. feedback messages as a loss tolerance mechanism. Receivers MAY send
PLI messages.
5.1.3. Slice Loss Indication (SLI) 5.1.3. Slice Loss Indication (SLI)
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF The Slice Loss Indication message is defined in Section 6.3.2 of the
profile [RFC4585]. It is used by a receiver to tell the encoder that RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
it has detected the loss or corruption of one or more consecutive encoder that it has detected the loss or corruption of one or more
macro blocks, and would like to have these repaired somehow. It is consecutive macro blocks, and would like to have these repaired
RECOMMENDED that receivers generate SLI feedback messages if slices somehow. It is RECOMMENDED that receivers generate SLI feedback
are lost when using a codec that supports the concept of macro messages if slices are lost when using a codec that supports the
blocks. A sender that receives an SLI feedback message SHOULD concept of macro blocks. A sender that receives an SLI feedback
attempt to repair the lost slice(s). message SHOULD attempt to repair the lost slice(s).
5.1.4. Reference Picture Selection Indication (RPSI) 5.1.4. Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) messages are defined in Reference Picture Selection Indication (RPSI) messages are defined in
Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding
standards allow the use of older reference pictures than the most standards allow the use of older reference pictures than the most
recent one for predictive coding. If such a codec is in use, and if recent one for predictive coding. If such a codec is in use, and if
the encoder has learnt that encoder-decoder synchronisation has been the encoder has learnt that encoder-decoder synchronisation has been
lost, then a known as correct reference picture can be used as a base lost, then a known as correct reference picture can be used as a base
for future coding. The RPSI message allows this to be signalled. for future coding. The RPSI message allows this to be signalled.
Receivers that detect that encoder-decoder synchronisation has been Receivers that detect that encoder-decoder synchronisation has been
lost SHOULD generate an RPSI feedback message if codec being used lost SHOULD generate an RPSI feedback message if codec being used
supports reference picture selection. A RTP media stream sender that supports reference picture selection. A RTP packet stream sender
receives such an RPSI message SHOULD act on that messages to change that receives such an RPSI message SHOULD act on that messages to
the reference picture, if it is possible to do so within the change the reference picture, if it is possible to do so within the
available bandwidth constraints. available bandwidth constraints, and with the codec being used.
5.1.5. Temporal-Spatial Trade-off Request (TSTR) 5.1.5. Temporal-Spatial Trade-off Request (TSTR)
The temporal-spatial trade-off request and notification are defined The temporal-spatial trade-off request and notification are defined
in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used
to ask the video encoder to change the trade-off it makes between to ask the video encoder to change the trade-off it makes between
temporal and spatial resolution, for example to prefer high spatial temporal and spatial resolution, for example to prefer high spatial
image quality but low frame rate. Support for TSTR requests and image quality but low frame rate. Support for TSTR requests and
notifications is OPTIONAL. notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
This feedback message is defined in Sections 3.5.4 and 4.2.1 of the The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of
Codec Control Messages [RFC5104]. This message and its notification the Codec Control Messages [RFC5104]. This request and its
message are used by a media receiver to inform the sending party that notification message are used by a media receiver to inform the
there is a current limitation on the amount of bandwidth available to sending party that there is a current limitation on the amount of
this receiver. This can be various reasons for this: for example, an bandwidth available to this receiver. This can be various reasons
RTP mixer can use this message to limit the media rate of the sender for this: for example, an RTP mixer can use this message to limit the
being forwarded by the mixer (without doing media transcoding) to fit media rate of the sender being forwarded by the mixer (without doing
the bottlenecks existing towards the other session participants. media transcoding) to fit the bottlenecks existing towards the other
WebRTC senders are REQUIRED to implement support for TMMBR messages, session participants. WebRTC senders are REQUIRED to implement
and MUST follow bandwidth limitations set by a TMMBR message received support for TMMBR messages, and MUST follow bandwidth limitations set
for their SSRC. The sending of TMMBR requests is OPTIONAL. by a TMMBR message received for their SSRC. The sending of TMMBR
requests is OPTIONAL.
5.2. Header Extensions 5.2. Header Extensions
The RTP specification [RFC3550] provides the capability to include The RTP specification [RFC3550] provides the capability to include
RTP header extensions containing in-band data, but the format and RTP header extensions containing in-band data, but the format and
semantics of the extensions are poorly specified. The use of header semantics of the extensions are poorly specified. The use of header
extensions is OPTIONAL in the WebRTC context, but if they are used, extensions is OPTIONAL in the WebRTC context, but if they are used,
they MUST be formatted and signalled following the general mechanism they MUST be formatted and signalled following the general mechanism
for RTP header extensions defined in [RFC5285], since this gives for RTP header extensions defined in [RFC5285], since this gives
well-defined semantics to RTP header extensions. well-defined semantics to RTP header extensions.
As noted in [RFC5285], the requirement from the RTP specification As noted in [RFC5285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions MUST be ignored" [RFC3550] stands. To be specific, header extensions MUST
only be used for data that can safely be ignored by the recipient only be used for data that can safely be ignored by the recipient
without affecting interoperability, and MUST NOT be used when the without affecting interoperability, and MUST NOT be used when the
presence of the extension has changed the form or nature of the rest presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream of the packet in a way that is not compatible with the way the stream
is signalled (e.g., as defined by the payload type). Valid examples is signalled (e.g., as defined by the payload type). Valid examples
might include metadata that is additional to the usual RTP of RTP header extensions might include metadata that is additional to
information. the usual RTP information, but that can safely be ignored without
compromising interoperability.
5.2.1. Rapid Synchronisation 5.2.1. Rapid Synchronisation
Many RTP sessions require synchronisation between audio, video, and Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however, specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented in addition to RTCP SR-based described in [RFC6051] be implemented in addition to RTCP SR-based
synchronisation. The rapid synchronisation extensions use the synchronisation. The rapid synchronisation extensions use the
general RTP header extension mechanism [RFC5285], which requires general RTP header extension mechanism [RFC5285], which requires
signalling, but are otherwise backwards compatible. signalling, but are otherwise backwards compatible.
5.2.2. Client-to-Mixer Audio Level 5.2.2. Client-to-Mixer Audio Level
The Client to Mixer Audio Level extension [RFC6464] is an RTP header The Client to Mixer Audio Level extension [RFC6464] is an RTP header
extension used by a client to inform a mixer about the level of audio extension used by an endpoint to inform a mixer about the level of
activity in the packet to which the header is attached. This enables audio activity in the packet to which the header is attached. This
a central node to make mixing or selection decisions without decoding enables an RTP middlebox to make mixing or selection decisions
or detailed inspection of the payload, reducing the complexity in without decoding or detailed inspection of the payload, reducing the
some types of central RTP nodes. It can also save decoding resources complexity in some types of mixer. It can also save decoding
in receivers, which can choose to decode only the most relevant RTP resources in receivers, which can choose to decode only the most
media streams based on audio activity levels. relevant RTP packet streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to The Client-to-Mixer Audio Level [RFC6464] header extension is
be implemented. If it is implemented, it is REQUIRED that the header RECOMMENDED to be implemented. If this header extension is
extensions are encrypted according to [RFC6904] since the information implemented, it is REQUIRED that implementations are capable of
contained in these header extensions can be considered sensitive. encrypting the header extension according to [RFC6904] since the
information contained in these header extensions can be considered
sensitive. It is further RECOMMENDED that this encryption is used,
unless the encryption has been explicitly disabled through API or
signalling.
5.2.3. Mixer-to-Client Audio Level 5.2.3. Mixer-to-Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides The Mixer to Client Audio Level header extension [RFC6465] provides
the client with the audio level of the different sources mixed into a an endpoint with the audio level of the different sources mixed into
common mix by a RTP mixer. This enables a user interface to indicate a common mix by a RTP mixer. This enables a user interface to
the relative activity level of each session participant, rather than indicate the relative activity level of each session participant,
just being included or not based on the CSRC field. This is a pure rather than just being included or not based on the CSRC field. This
optimisations of non critical functions, and is hence OPTIONAL to is a pure optimisations of non critical functions, and is hence
implement. If it is implemented, it is REQUIRED that the header OPTIONAL to implement. If this header extension is implemented, it
extensions are encrypted according to [RFC6904] since the information is REQUIRED that implementations are capable of encrypting the header
contained in these header extensions can be considered sensitive. extension according to [RFC6904] since the information contained in
these header extensions can be considered sensitive. It is further
RECOMMENDED that this encryption is used, unless the encryption has
been explicitly disabled through API or signalling.
6. WebRTC Use of RTP: Improving Transport Robustness 6. WebRTC Use of RTP: Improving Transport Robustness
There are tools that can make RTP media streams robust against packet There are tools that can make RTP packet streams robust against
loss and reduce the impact of loss on media quality. However, they packet loss and reduce the impact of loss on media quality. However,
all add extra bits compared to a non-robust stream. The overhead of they all add overhead compared to a non-robust stream. The overhead
these extra bits needs to be considered, and the aggregate bit-rate needs to be considered, and the aggregate bit-rate MUST be rate
MUST be rate controlled to avoid causing network congestion (see controlled to avoid causing network congestion (see Section 7). As a
Section 7). As a result, improving robustness might require a lower result, improving robustness might require a lower base encoding
base encoding quality, but has the potential to deliver that quality quality, but has the potential to deliver that quality with fewer
with fewer errors. The mechanisms described in the following sub- errors. The mechanisms described in the following sub-sections can
sections can be used to improve tolerance to packet loss. be used to improve tolerance to packet loss.
6.1. Negative Acknowledgements and RTP Retransmission 6.1. Negative Acknowledgements and RTP Retransmission
As a consequence of supporting the RTP/SAVPF profile, implementations As a consequence of supporting the RTP/SAVPF profile, implementations
can support negative acknowledgements (NACKs) for RTP data packets can send negative acknowledgements (NACKs) for RTP data packets
[RFC4585]. This feedback can be used to inform a sender of the loss [RFC4585]. This feedback can be used to inform a sender of the loss
of particular RTP packets, subject to the capacity limitations of the of particular RTP packets, subject to the capacity limitations of the
RTCP feedback channel. A sender can use this information to optimise RTCP feedback channel. A sender can use this information to optimise
the user experience by adapting the media encoding to compensate for the user experience by adapting the media encoding to compensate for
known lost packets, for example. known lost packets.
RTP Media Stream Senders are REQUIRED to understand the Generic NACK RTP packet stream Senders are REQUIRED to understand the Generic NACK
message defined in Section 6.2.1 of [RFC4585], but MAY choose to message defined in Section 6.2.1 of [RFC4585], but MAY choose to
ignore this feedback (following Section 4.2 of [RFC4585]). Receivers ignore some or all of this feedback (following Section 4.2 of
MAY send NACKs for missing RTP packets; [RFC4585] provides some [RFC4585]). Receivers MAY send NACKs for missing RTP packets.
guidelines on when to send NACKs. It is not expected that a receiver Guidelines on when to send NACKs are provided in [RFC4585]. It is
will send a NACK for every lost RTP packet, rather it needs to not expected that a receiver will send a NACK for every lost RTP
consider the cost of sending NACK feedback, and the importance of the packet, rather it needs to consider the cost of sending NACK
lost packet, to make an informed decision on whether it is worth feedback, and the importance of the lost packet, to make an informed
telling the sender about a packet loss event. decision on whether it is worth telling the sender about a packet
loss event.
The RTP Retransmission Payload Format [RFC4588] offers the ability to The RTP Retransmission Payload Format [RFC4588] offers the ability to
retransmit lost packets based on NACK feedback. Retransmission needs retransmit lost packets based on NACK feedback. Retransmission needs
to be used with care in interactive real-time applications to ensure to be used with care in interactive real-time applications to ensure
that the retransmitted packet arrives in time to be useful, but can that the retransmitted packet arrives in time to be useful, but can
be effective in environments with relatively low network RTT (an RTP be effective in environments with relatively low network RTT (an RTP
sender can estimate the RTT to the receivers using the information in sender can estimate the RTT to the receivers using the information in
RTCP SR and RR packets, as described at the end of Section 6.4.1 of RTCP SR and RR packets, as described at the end of Section 6.4.1 of
[RFC3550]). The use of retransmissions can also increase the forward [RFC3550]). The use of retransmissions can also increase the forward
RTP bandwidth, and can potentially worsen the problem if the packet RTP bandwidth, and can potentially caused increased packet loss if
loss was caused by network congestion. We note, however, that the original packet loss was caused by network congestion. We note,
retransmission of an important lost packet to repair decoder state however, that retransmission of an important lost packet to repair
can have lower cost than sending a full intra frame. It is not decoder state can have lower cost than sending a full intra frame.
appropriate to blindly retransmit RTP packets in response to a NACK. It is not appropriate to blindly retransmit RTP packets in response
to a NACK. The importance of lost packets and the likelihood of them
The importance of lost packets and the likelihood of them arriving in arriving in time to be useful needs to be considered before RTP
time to be useful needs to be considered before RTP retransmission is retransmission is used.
used.
Receivers are REQUIRED to implement support for RTP retransmission Receivers are REQUIRED to implement support for RTP retransmission
packets [RFC4588]. Senders MAY send RTP retransmission packets in packets [RFC4588]. Senders MAY send RTP retransmission packets in
response to NACKs if the RTP retransmission payload format has been response to NACKs if the RTP retransmission payload format has been
negotiated for the session, and if the sender believes it is useful negotiated for the session, and if the sender believes it is useful
to send a retransmission of the packet(s) referenced in the NACK. An to send a retransmission of the packet(s) referenced in the NACK. An
RTP sender does not need to retransmit every NACKed packet. RTP sender does not need to retransmit every NACKed packet.
6.2. Forward Error Correction (FEC) 6.2. Forward Error Correction (FEC)
The use of Forward Error Correction (FEC) can provide an effective The use of Forward Error Correction (FEC) can provide an effective
protection against some degree of packet loss, at the cost of steady protection against some degree of packet loss, at the cost of steady
bandwidth overhead. There are several FEC schemes that are defined bandwidth overhead. There are several FEC schemes that are defined
for use with RTP. Some of these schemes are specific to a particular for use with RTP. Some of these schemes are specific to a particular
RTP payload format, others operate across RTP packets and can be used RTP payload format, others operate across RTP packets and can be used
with any payload format. It needs to be noted that using redundant with any payload format. It needs to be noted that using redundant
encoding or FEC will lead to increased play out delay, which needs to encoding or FEC will lead to increased play out delay, which needs to
be considered when choosing the redundancy or FEC formats and their be considered when choosing the redundancy or FEC formats and their
respective parameters. respective parameters.
If an RTP payload format negotiated for use in a WebRTC session If an RTP payload format negotiated for use in a RTCPeerConnection
supports redundant transmission or FEC as a standard feature of that supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the WebRTC session, payload format, then that support MAY be used in the
subject to any appropriate signalling. RTCPeerConnection, subject to any appropriate signalling.
There are several block-based FEC schemes that are designed for use There are several block-based FEC schemes that are designed for use
with RTP independent of the chosen RTP payload format. At the time with RTP independent of the chosen RTP payload format. At the time
of this writing there is no consensus on which, if any, of these FEC of this writing there is no consensus on which, if any, of these FEC
schemes is appropriate for use in the WebRTC context. Accordingly, schemes is appropriate for use in the WebRTC context. Accordingly,
this memo makes no recommendation on the choice of block-based FEC this memo makes no recommendation on the choice of block-based FEC
for WebRTC use. for WebRTC use.
7. WebRTC Use of RTP: Rate Control and Media Adaptation 7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in heterogeneous network environments using a WebRTC will be used in heterogeneous network environments using a
variety set of link technologies, including both wired and wireless variety set of link technologies, including both wired and wireless
links, to interconnect potentially large groups of users around the links, to interconnect potentially large groups of users around the
world. As a result, the network paths between users can have widely world. As a result, the network paths between users can have widely
varying one-way delays, available bit-rates, load levels, and traffic varying one-way delays, available bit-rates, load levels, and traffic
mixtures. Individual end-points can send one or more RTP media mixtures. Individual end-points can send one or more RTP packet
streams to each participant in a WebRTC conference, and there can be streams to each participant in a WebRTC conference, and there can be
several participants. Each of these RTP media streams can contain several participants. Each of these RTP packet streams can contain
different types of media, and the type of media, bit rate, and number different types of media, and the type of media, bit rate, and number
of flows can be highly asymmetric. Non-RTP traffic can share the of RTP packet streams as well as transport-layer flows can be highly
network paths with RTP flows. Since the network environment is not asymmetric. Non-RTP traffic can share the network paths with RTP
predictable or stable, WebRTC endpoints MUST ensure that the RTP transport-layer flows. Since the network environment is not
predictable or stable, WebRTC end-points MUST ensure that the RTP
traffic they generate can adapt to match changes in the available traffic they generate can adapt to match changes in the available
network capacity. network capacity.
The quality of experience for users of WebRTC implementation is very The quality of experience for users of WebRTC implementation is very
dependent on effective adaptation of the media to the limitations of dependent on effective adaptation of the media to the limitations of
the network. End-points have to be designed so they do not transmit the network. End-points have to be designed so they do not transmit
significantly more data than the network path can support, except for significantly more data than the network path can support, except for
very short time periods, otherwise high levels of network packet loss very short time periods, otherwise high levels of network packet loss
or delay spikes will occur, causing media quality degradation. The or delay spikes will occur, causing media quality degradation. The
limiting factor on the capacity of the network path might be the link limiting factor on the capacity of the network path might be the link
bandwidth, or it might be competition with other traffic on the link bandwidth, or it might be competition with other traffic on the link
(this can be non-WebRTC traffic, traffic due to other WebRTC flows, (this can be non-WebRTC traffic, traffic due to other WebRTC flows,
or even competition with other WebRTC flows in the same session). or even competition with other WebRTC flows in the same session).
An effective media congestion control algorithm is therefore an An effective media congestion control algorithm is therefore an
essential part of the WebRTC framework. However, at the time of this essential part of the WebRTC framework. However, at the time of this
writing, there is no standard congestion control algorithm that can writing, there is no standard congestion control algorithm that can
be used for interactive media applications such as WebRTC flows. be used for interactive media applications such as WebRTC's flows.
Some requirements for congestion control algorithms for WebRTC Some requirements for congestion control algorithms for
sessions are discussed in [I-D.ietf-rmcat-cc-requirements], and it is RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements].
expected that a future version of this memo will mandate the use of a It is expected that a future version of this memo will mandate the
congestion control algorithm that satisfies these requirements. use of a congestion control algorithm that satisfies these
requirements.
7.1. Boundary Conditions and Circuit Breakers 7.1. Boundary Conditions and Circuit Breakers
In the absence of a concrete congestion control algorithm, all WebRTC In the absence of a concrete congestion control algorithm, all WebRTC
implementations MUST implement the RTP circuit breaker algorithm that implementations MUST implement the RTP circuit breaker algorithm that
is in described [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP
circuit breaker is designed to enable applications to recognise and circuit breaker is designed to enable applications to recognise and
react to situations of extreme network congestion. However, since react to situations of extreme network congestion. However, since
the RTP circuit breaker might not be triggered until congestion the RTP circuit breaker might not be triggered until congestion
becomes extreme, it cannot be considered a substitute for congestion becomes extreme, it cannot be considered a substitute for congestion
control, and applications MUST also implement congestion control to control, and applications MUST also implement congestion control to
allow them to adapt to changes in network capacity. Any future RTP allow them to adapt to changes in network capacity. Any future RTP
congestion control algorithms are expected to operate within the congestion control algorithms are expected to operate within the
envelope allowed by the circuit breaker. envelope allowed by the circuit breaker.
The session establishment signalling will also necessarily establish The session establishment signalling will also necessarily establish
skipping to change at page 19, line 21 skipping to change at page 20, line 35
"b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
The combination of media codec choice and signalled bandwidth limits The combination of media codec choice and signalled bandwidth limits
SHOULD be used to limit traffic based on known bandwidth limitations, SHOULD be used to limit traffic based on known bandwidth limitations,
for example the capacity of the edge links, to the extent possible. for example the capacity of the edge links, to the extent possible.
7.2. RTCP Limitations for Congestion Control 7.2. RTCP Limitations for Congestion Control
Experience with the congestion control algorithms of TCP [RFC5681], Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent roughly once per that feedback on packet arrivals needs to be sent frequently (roughly
round trip time. We note that the real-time media traffic might not once per round trip time is common). We note that the real-time
have to adapt to changing path conditions as rapidly as needed for media traffic might not be able to adapt to changing path conditions
the elastic applications TCP was designed for, but frequent feedback as rapidly as elastic applications using TCP, but frequent feedback,
is still needed to allow the congestion control algorithm to track perhaps on the order of once per video frame, is still needed to
the path dynamics. allow the congestion control algorithm to track the path dynamics.
The total RTCP bandwidth is normally limited in its transmission rate As an example of the type of RTCP congestion control feedback that is
to a fraction of the nominal RTP traffic (by default 5%). RTCP possible, consider one of the simplest scenarios for WebRTC: a point
packets are larger than, e.g., TCP ACKs (even when non-compound RTCP to point video call between two end systems. There will be four RTP
packets are used). The RTP media stream bit rate thus limits the flows in this scenario, two audio and two video, with all four flows
maximum feedback rate as a function of the mean RTCP packet size. 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 doesn't cause transmissions to
stop). Assume all four flows are sent in a single RTP session, each
using a separate SSRC. Further, assume each SSRC sends RTCP reports
for all other SSRCs in the session (i.e., the optimisations in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving
the worst case for the RTCP overhead). When all members are senders
like this, the RTCP timing rules in Sections 6.2 and 6.3 of [RFC3550]
and [RFC4585] reduce to:
Interactive communication might not be able to afford waiting for rtcp_interval = avg_rtcp_size * n / rtcp_bw
where avg_rtcp_size is measured in octets, and the rtcp_bw is the
bandwidth available for RTCP. The average RTCP size will depend on
the amount of feedback that is sent in each RTCP packet, on the
number of members in the session, and on the size of source
description (RTCP SDES) information sent. As a baseline, each RTCP
packet will be a compound RTCP packet that contains an RTCP SR and an
RTCP SDES packet. In the scenario above, each RTCP SR packet will
contain three report blocks, once for each of the other RTP SSRCs
sending data, for a total of 100 octets (this is 8 octets header, 20
octets sender info, and 3 * 24 octets report blocks). The RTCP SDES
packet will comprise a header (4 octets), an originating SSRC (4
octets), a CNAME chunk, and padding. If the CNAME follows [RFC7022]
and it will be 19 octets in size, and require 1 octet of padding.
The resulting compound RTCP packet will be 128 octets in size. If
sent in UDP/IPv4 with no IP options and using Secure RTP, which adds
20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit Authentication tag), the
avg_rtcp_size will therefore be 170 octets, including the header
overhead. The value n is this scenario is 4, and the rtcp_bw is
assumed to be 5% of the session bandwidth.
If it is desired to send RTCP feedback packets on average 30 times
per second, to correspond to one RTCP report every frame for 30fps
video, we can invert the above rtcp_interval calculation to get an
rtcp_bw that gives an interval of 1/30th of a second or lower. This
corresponds to an rtcp_bw of 20400 octets per second (since 1/30 =
170 * 4 / 20400). This is 163200 bits per second, which if 5% of the
session bandwidth, gives a session bandwidth of approximately 3.3Mbps
(i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a
total data rate of approximately 3.4Mbps). That is, RTCP can report
on every frame of video provided the session bandwidth is 3.3Mbps or
larger, when every SSRC sends a report for every video frame. Please
note that the actual RTCP transmission intervals will be within the
interval [0.0135, 0.0406]s, but maintaining an average RTCP
transmission interval of 0.033s.
Note: To achieve the RTCP transmission intervals above the RTP/
SAVPF profile with T_rr_interval=0 is used, since even when using
the reduced minimal transmission interval, the RTP/SAVP profile
would only allow sending RTCP at most every 0.11s (every third
frame of video). Using RTP/SAVPF with T_rr_interval=0 however is
capable of fully utilizing the configured 5% RTCP bandwidth
fraction.
If additional feedback beyond the standard report block is needed,
the session bandwidth needed will increase. For example, with an
additional 20 octets data being reported in each RTCP packet, the
session bandwidth needed increases to 3.5Mbps for every SSRC to be
able to report on every frame. However, the above baseline might not
be the most appropriate usage of the RTCP bandwidth. Depending on
needs, a less frequent usage of regular RTCP compound packets,
controlled by T_rr_interval combined with using the reduced size RTCP
packets, can achieve more frequent and useful reporting. Also the
reporting requirements defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] will reduced the
amount of bandwidth consumed for reporting when each endpoint has
multiple SSRCs.
Calculations such as these show that RTCP cannot be used to send per-
packet congestion feedback. RTCP can, however, be used to send
congestion feedback on each frame of video sent in an interactive
video conferencing scenario, provided the RTCP parameters are
correctly configured and the overall session bandwidth exceeds a
couple of megabits per second (the exact rate depending on the number
of session participants, the RTCP bandwidth fraction, and whether
audio and video are sent in one or two RTP sessions). Using similar
calculations, it can be shown that RTCP can likely also be used to
send feedback on a per-RTT basis, provided the RTT is not too low.
Interactive communication might not be able to afford to wait for
packet losses to occur to indicate congestion, because an increase in packet losses to occur to indicate congestion, because an increase in
play out delay due to queuing (most prominent in wireless networks) play out delay due to queuing (most prominent in wireless networks)
can easily lead to packets being dropped due to late arrival at the can easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues might need to be receiver. Therefore, more sophisticated cues might need to be
reported -- to be defined in a suitable congestion control framework reported -- to be defined in a suitable congestion control framework
as noted above -- which, in turn, increase the report size again. as noted above -- which, in turn, increase the report size again.
For example, different RTCP XR report blocks (jointly) provide the For example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control necessary details to implement a variety of congestion control
algorithms, but the (compound) report size grows quickly. algorithms, but the (compound) report size grows quickly.
In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the
reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session. An
endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video.
7.3. Congestion Control Interoperability and Legacy Systems 7.3. Congestion Control Interoperability and Legacy Systems
There are legacy implementations that do not implement RTCP, and There are legacy RTP implementations that do not implement RTCP, and
hence do not provide any congestion feedback. Congestion control hence do not provide any congestion feedback. Congestion control
cannot be performed with these end-points. WebRTC implementations cannot be performed with these end-points. WebRTC implementations
that need to interwork with such end-points MUST limit their that need to interwork with such end-points MUST limit their
transmission to a low rate, equivalent to a VoIP call using a low transmission to a low rate, equivalent to a VoIP call using a low
bandwidth codec, that is unlikely to cause any significant bandwidth codec, that is unlikely to cause any significant
congestion. congestion.
When interworking with legacy implementations that support RTCP using When interworking with legacy implementations that support RTCP using
the RTP/AVP profile [RFC3551], congestion feedback is provided in the RTP/AVP profile [RFC3551], congestion feedback is provided in
RTCP RR packets every few seconds. Implementations that have to RTCP RR packets every few seconds. Implementations that have to
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both directions. If the IETF were to standardise both sender- and both directions. If the IETF were to standardise both sender- and
receiver-based congestion control algorithms for WebRTC traffic in receiver-based congestion control algorithms for WebRTC traffic in
the future, the issues of interoperability, control, and ensuring the future, the issues of interoperability, control, and ensuring
that both directions of media flow are congestion controlled would that both directions of media flow are congestion controlled would
also need to be considered. also need to be considered.
8. WebRTC Use of RTP: Performance Monitoring 8. WebRTC Use of RTP: Performance Monitoring
As described in Section 4.1, implementations are REQUIRED to generate As described in Section 4.1, implementations are REQUIRED to generate
RTCP Sender Report (SR) and Reception Report (RR) packets relating to RTCP Sender Report (SR) and Reception Report (RR) packets relating to
the RTP media streams they send and receive. These RTCP reports can the RTP packet streams they send and receive. These RTCP reports can
be used for performance monitoring purposes, since they include basic be used for performance monitoring purposes, since they include basic
packet loss and jitter statistics. packet loss and jitter statistics.
A large number of additional performance metrics are supported by the A large number of additional performance metrics are supported by the
RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. It is not RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time
yet clear what extended metrics are appropriate for use in the WebRTC of this writing, it is not clear what extended metrics are suitable
context, so there is no requirement that implementations generate for use in the WebRTC context, so there is no requirement that
RTCP XR packets. However, implementations that can use detailed implementations generate RTCP XR packets. However, implementations
performance monitoring data MAY generate RTCP XR packets as that can use detailed performance monitoring data MAY generate RTCP
appropriate; the use of such packets SHOULD be signalled in advance. XR packets as appropriate; the use of such packets SHOULD be
signalled in advance.
All WebRTC implementations MUST be prepared to receive RTP XR report All WebRTC implementations MUST be prepared to receive RTP XR report
packets, whether or not they were signalled. There is no requirement packets, whether or not they were signalled. There is no requirement
that the data contained in such reports be used, or exposed to the that the data contained in such reports be used, or exposed to the
Javascript application, however. Javascript application, however.
9. WebRTC Use of RTP: Future Extensions 9. WebRTC Use of RTP: Future Extensions
It is possible that the core set of RTP protocols and RTP extensions It is possible that the core set of RTP protocols and RTP extensions
specified in this memo will prove insufficient for the future needs specified in this memo will prove insufficient for the future needs
skipping to change at page 22, line 11 skipping to change at page 24, line 48
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711] on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the not directly interoperate with the non-secure variants, due to the
presence of additional header fields for authentication in SRTP presence of additional header fields for authentication in SRTP
packets and cryptographic transformation of the payload. WebRTC packets and cryptographic transformation of the payload. WebRTC
requires the use of the RTP/SAVPF profile, and this MUST be requires the use of the RTP/SAVPF profile, and this MUST be
signalled if SDP is used. Interworking functions might transform signalled if SDP is used. Interworking functions might transform
this into the RTP/SAVP profile for a legacy use case, by this into the RTP/SAVP profile for a legacy use case, by
indicating to the WebRTC end-point that the RTP/SAVPF is used, and indicating to the WebRTC end-point that the RTP/SAVPF is used, and
limiting the usage of the "a=rtcp:" attribute to indicate a trr- limiting the usage of the "a=rtcp-fb:" attribute to indicate a
int value of 4 seconds. trr-int value of 4 seconds.
Transport Information: Source and destination IP address(s) and Transport Information: Source and destination IP address(s) and
ports for RTP and RTCP MUST be signalled for each RTP session. In ports for RTP and RTCP MUST be signalled for each RTP session. In
WebRTC these transport addresses will be provided by ICE that WebRTC these transport addresses will be provided by ICE that
signals candidates and arrives at nominated candidate address signals candidates and arrives at nominated candidate address
pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such
that a single port is used for RTP and RTCP flows, this MUST be that a single port, i.e. transport-layer flow, is used for RTP and
signalled (see Section 4.5). RTCP flows, this MUST be signalled (see Section 4.5).
RTP Payload Types, media formats, and format parameters: The mapping RTP Payload Types, media formats, and format parameters: The mapping
between media type names (and hence the RTP payload formats to be between media type names (and hence the RTP payload formats to be
used), and the RTP payload type numbers MUST be signalled. Each used), and the RTP payload type numbers MUST be signalled. Each
media type MAY also have a number of media type parameters that media type MAY also have a number of media type parameters that
MUST also be signalled to configure the codec and RTP payload MUST also be signalled to configure the codec and RTP payload
format (the "a=fmtp:" line from SDP). Section 4.3 of this memo format (the "a=fmtp:" line from SDP). Section 4.3 of this memo
discusses requirements for uniqueness of payload types. discusses requirements for uniqueness of payload types.
RTP Extensions: The RTP extensions to be used SHOULD be agreed upon, RTP Extensions: The RTP extensions to be used SHOULD be agreed upon,
skipping to change at page 23, line 22 skipping to change at page 25, line 51
11. WebRTC API Considerations 11. WebRTC API Considerations
The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and
Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses
the concept of a MediaStream that consists of zero or more the concept of a MediaStream that consists of zero or more
MediaStreamTracks. A MediaStreamTrack is an individual stream of MediaStreamTracks. A MediaStreamTrack is an individual stream of
media from any type of media source like a microphone or a camera, media from any type of media source like a microphone or a camera,
but also conceptual sources, like a audio mix or a video composition, but also conceptual sources, like a audio mix or a video composition,
are possible. The MediaStreamTracks within a MediaStream need to be are possible. The MediaStreamTracks within a MediaStream need to be
possible to play out synchronised. The below text uses the possible to play out synchronised.
terminology from [I-D.ietf-avtext-rtp-grouping-taxonomy].
A MediaStreamTrack's realisation in RTP in the context of an A MediaStreamTrack's realisation in RTP in the context of an
RTCPeerConnection consists of a source packet stream identified with RTCPeerConnection consists of a source packet stream identified with
an SSRC within an RTP session part of the RTCPeerConnection. The an SSRC within an RTP session part of the RTCPeerConnection. The
MediaStreamTrack can also result in additional packet streams, and MediaStreamTrack can also result in additional packet streams, and
thus SSRCs, in the same RTP session. These can be dependent packet thus SSRCs, in the same RTP session. These can be dependent packet
streams from scalable encoding of the source stream associated with streams from scalable encoding of the source stream associated with
the MediaStreamTrack, if such a media encoder is used. They can also the MediaStreamTrack, if such a media encoder is used. They can also
be redundancy packet streams, these are created when applying Forward be redundancy packet streams, these are created when applying Forward
Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to
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for multiple source packet streams to share encoded streams (but not for multiple source packet streams to share encoded streams (but not
packet streams), but this is an implementation choice to try to packet streams), but this is an implementation choice to try to
utilise such optimisations. Note that such optimizations would need utilise such optimisations. Note that such optimizations would need
to take into account that the constraints for one of the to take into account that the constraints for one of the
MediaStreamTracks can at any moment change, meaning that the encoding MediaStreamTracks can at any moment change, meaning that the encoding
configurations might no longer be identical. configurations might no longer be identical.
The same MediaStreamTrack can also be included in multiple The same MediaStreamTrack can also be included in multiple
MediaStreams, thus multiple sets of MediaStreams can implicitly need MediaStreams, thus multiple sets of MediaStreams can implicitly need
to use the same synchronisation base. To ensure that this works in to use the same synchronisation base. To ensure that this works in
all cases, and don't forces a endpoint to change synchronisation base all cases, and don't forces a end-point to change synchronisation
and CNAME in the middle of a ongoing delivery of any packet streams, base and CNAME in the middle of a ongoing delivery of any packet
which would cause media disruption; all MediaStreamTracks and their streams, which would cause media disruption; all MediaStreamTracks
associated SSRCs originating from the same endpoint MUST be sent and their associated SSRCs originating from the same end-point needs
using the same CNAME within one RTCPeerConnection as well as across to be sent using the same CNAME within one RTCPeerConnection. This
all RTCPeerConnections part of the same communication session is motivating the strong recommendation in Section 4.9 to only use a
context, which for a browser are a single origin. single CNAME.
Note: It is important that the same CNAME is not used in different The requirement on using the same CNAME for all SSRCs that
communication session contexts or origins, as that could enable originates from the same end-point, does not require middleboxes
tracking of a user and its device usage of different services. that forwards traffic from multiple end-points to only use a
See Section 4.4.1 of Security Considerations for WebRTC single CNAME.
[I-D.ietf-rtcweb-security] for further discussion.
The reasons to require the same CNAME across multiple Different CNAMEs normally need to be used for different
RTCPeerConnections is to enable synchronisation of different RTCPeerConnection instances, as specified in Section 4.9. Having two
MediaStreamTracks originating from one endpoint despite them being communication sessions with the same CNAME could enable tracking of a
transported over different RTCPeerConnections. user or device across different services (see Section 4.4.1 of
[I-D.ietf-rtcweb-security] for details). A web application can
request that the CNAMEs used in different RTCPeerConnection within a
same-orign context to be the same, this allow for synchronization of
the endpoint's RTP packet streams across the different
RTCPeerConnections.
The above will currently force a WebRTC endpoint that receives an Note: this doesn't result in a tracking issue, since the creation
of matching CNAMEs depends on existing tracking.
The above will currently force a WebRTC end-point that receives an
MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
on any RTCPeerConnection to perform resynchronisation of the stream. on any RTCPeerConnection to perform resynchronisation of the stream.
This, as the sending party needs to change the CNAME, which implies This, as the sending party needs to change the CNAME, which implies
that it has to use a locally available system clock as timebase for that it has to use a locally available system clock as timebase for
the synchronisation. Thus, the relative relation between the the synchronisation. Thus, the relative relation between the
timebase of the incoming stream and the system sending out needs to timebase of the incoming stream and the system sending out needs to
defined. This relation also needs monitoring for clock drift and defined. This relation also needs monitoring for clock drift and
likely adjustments of the synchronisation. The sending entity is likely adjustments of the synchronisation. The sending entity is
also responsible for congestion control for its the sent streams. In also responsible for congestion control for its the sent streams. In
cases of packet loss the loss of incoming data also needs to be cases of packet loss the loss of incoming data also needs to be
handled. This leads to the observation that the method that is least handled. This leads to the observation that the method that is least
likely to cause issues or interruptions in the outgoing source packet likely to cause issues or interruptions in the outgoing source packet
stream is a model of full decoding, including repair etc followed by stream is a model of full decoding, including repair etc followed by
encoding of the media again into the outgoing packet stream. encoding of the media again into the outgoing packet stream.
Optimisations of this method is clearly possible and implementation Optimisations of this method is clearly possible and implementation
specific. specific.
A WebRTC endpoint MUST support receiving multiple MediaStreamTracks, A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
where each of different MediaStreamTracks (and their sets of where each of different MediaStreamTracks (and their sets of
associated packet streams) uses different CNAMEs. However, associated packet streams) uses different CNAMEs. However,
MediaStreamTracks that are received with different CNAMEs have no MediaStreamTracks that are received with different CNAMEs have no
defined synchronisation. defined synchronisation.
Note: The motivation for supporting reception of multiple CNAMEs Note: The motivation for supporting reception of multiple CNAMEs
are to allow for forward compatibility with any future changes are to allow for forward compatibility with any future changes
that enables more efficient stream handling when endpoints relay/ that enables more efficient stream handling when end-points relay/
forward streams. It also ensures that endpoints can interoperate forward streams. It also ensures that end-points can interoperate
with certain types of multi-stream middleboxes or endpoints that with certain types of multi-stream middleboxes or end-points that
are not WebRTC. are not WebRTC.
The binding between the WebRTC MediaStreams, MediaStreamTracks and The binding between the WebRTC MediaStreams, MediaStreamTracks and
the SSRC is done as specified in "Cross Session Stream Identification the SSRC is done as specified in "Cross Session Stream Identification
in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This
document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to
map unknown source packet stream SSRCs to MediaStreamTracks and map unknown source packet stream SSRCs to MediaStreamTracks and
MediaStreams. Commonly the RTP Payload Type of any incoming packets MediaStreams. Commonly the RTP Payload Type of any incoming packets
will reveal if the packet stream is a source stream or a redundancy will reveal if the packet stream is a source stream or a redundancy
or dependent packet stream. The association to the correct source or dependent packet stream. The association to the correct source
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12. RTP Implementation Considerations 12. RTP Implementation Considerations
The following discussion provides some guidance on the implementation The following discussion provides some guidance on the implementation
of the RTP features described in this memo. The focus is on a WebRTC of the RTP features described in this memo. The focus is on a WebRTC
end-point implementation perspective, and while some mention is made end-point implementation perspective, and while some mention is made
of the behaviour of middleboxes, that is not the focus of this memo. of the behaviour of middleboxes, that is not the focus of this memo.
12.1. Configuration and Use of RTP Sessions 12.1. Configuration and Use of RTP Sessions
A WebRTC end-point will be a simultaneous participant in one or more A WebRTC end-point will be a simultaneous participant in one or more
RTP sessions. Each RTP session can convey multiple media flows, and RTP sessions. Each RTP session can convey multiple media sources,
can include media data from multiple end-points. In the following, and can include media data from multiple end-points. In the
we outline some ways in which WebRTC end-points can configure and use following, we outline some ways in which WebRTC end-points can
RTP sessions. configure and use RTP sessions.
12.1.1. Use of Multiple Media Flows Within an RTP Session 12.1.1. Use of Multiple Media Sources Within an RTP Session
RTP is a group communication protocol, and in a WebRTC context every RTP is a group communication protocol, and every RTP session can
RTP session can potentially contain multiple media flows. There are potentially contain multiple RTP packet streams. There are several
several reasons why this might be desirable: reasons why this might be desirable:
Multiple media types: Outside of WebRTC, it is common to use one RTP Multiple media types: Outside of WebRTC, it is common to use one RTP
session for each type of media (e.g., one RTP session for audio session for each type of media sources (e.g., one RTP session for
and one for video, each sent on a different UDP port). However, audio sources and one for video sources, each sent over different
to reduce the number of UDP ports used, the default in WebRTC is transport layer flows). However, to reduce the number of UDP
to send all types of media in a single RTP session, as described ports used, the default in WebRTC is to send all types of media in
in Section 4.4, using RTP and RTCP multiplexing (Section 4.5) to a single RTP session, as described in Section 4.4, using RTP and
further reduce the number of UDP ports needed. This RTP session RTCP multiplexing (Section 4.5) to further reduce the number of
then uses only one UDP flow, but will contain multiple RTP media UDP ports needed. This RTP session then uses only one bi-
streams, each containing a different type of media. A common directional transport-layer flow, but will contain multiple RTP
example might be an end-point with a camera and microphone that packet streams, each containing a different type of media. A
sends two RTP streams, one video and one audio, into a single RTP common example might be an end-point with a camera and microphone
session. that sends two RTP packet streams, one video and one audio, into a
single RTP session.
Multiple Capture Devices: A WebRTC end-point might have multiple Multiple Capture Devices: A WebRTC end-point might have multiple
cameras, microphones, or other media capture devices, and so might cameras, microphones, or other media capture devices, and so might
want to generate several RTP media streams of the same media type. want to generate several RTP packet streams of the same media
Alternatively, it might want to send media from a single capture type. Alternatively, it might want to send media from a single
device in several different formats or quality settings at once. capture device in several different formats or quality settings at
Both can result in a single end-point sending multiple RTP media once. Both can result in a single end-point sending multiple RTP
streams of the same media type into a single RTP session at the packet streams of the same media type into a single RTP session at
same time. the same time.
Associated Repair Data: An end-point might send a media stream that Associated Repair Data: An end-point might send a RTP packet stream
is somehow associated with another stream. For example, it might that is somehow associated with another stream. For example, it
send an RTP stream that contains FEC or retransmission data might send an RTP packet stream that contains FEC or
relating to another stream. Some RTP payload formats send this retransmission data relating to another stream. Some RTP payload
sort of associated repair data as part of the original media formats send this sort of associated repair data as part of the
stream, while others send it as a separate stream. source packet stream, while others send it as a separate packet
stream.
Layered or Multiple Description Coding: An end-point can use a Layered or Multiple Description Coding: An end-point can use a
layered media codec, for example H.264 SVC, or a multiple layered media codec, for example H.264 SVC, or a multiple
description codec, that generates multiple media flows, each with description codec, that generates multiple RTP packet streams,
a distinct RTP SSRC, within a single RTP session. each with a distinct RTP SSRC, within a single RTP session.
RTP Mixers, Translators, and Other Middleboxes: An RTP session, in RTP Mixers, Translators, and Other Middleboxes: An RTP session, in
the WebRTC context, is a point-to-point association between an the WebRTC context, is a point-to-point association between an
end-point and some other peer device, where those devices share a end-point and some other peer device, where those devices share a
common SSRC space. The peer device might be another WebRTC end- common SSRC space. The peer device might be another WebRTC end-
point, or it might be an RTP mixer, translator, or some other form point, or it might be an RTP mixer, translator, or some other form
of media processing middlebox. In the latter cases, the middlebox of media processing middlebox. In the latter cases, the middlebox
might send mixed or relayed RTP streams from several participants, might send mixed or relayed RTP streams from several participants,
that the WebRTC end-point will need to render. Thus, even though that the WebRTC end-point will need to render. Thus, even though
a WebRTC end-point might only be a member of a single RTP session, a WebRTC end-point might only be a member of a single RTP session,
the peer device might be extending that RTP session to incorporate the peer device might be extending that RTP session to incorporate
other end-points. WebRTC is a group communication environment and other end-points. WebRTC is a group communication environment and
end-points need to be capable of receiving, decoding, and playing end-points need to be capable of receiving, decoding, and playing
out multiple RTP media streams at once, even in a single RTP out multiple RTP packet streams at once, even in a single RTP
session. session.
12.1.2. Use of Multiple RTP Sessions 12.1.2. Use of Multiple RTP Sessions
In addition to sending and receiving multiple media streams within a In addition to sending and receiving multiple RTP packet streams
single RTP session, a WebRTC end-point might participate in multiple within a single RTP session, a WebRTC end-point might participate in
RTP sessions. There are several reasons why a WebRTC end-point might multiple RTP sessions. There are several reasons why a WebRTC end-
choose to do this: point might choose to do this:
To interoperate with legacy devices: The common practice in the non- To interoperate with legacy devices: The common practice in the non-
WebRTC world is to send different types of media in separate RTP WebRTC world is to send different types of media in separate RTP
sessions, for example using one RTP session for audio and another sessions, for example using one RTP session for audio and another
RTP session, on a different UDP port, for video. All WebRTC end- RTP session, on a separate transport layer flow, for video. All
points need to support the option of sending different types of WebRTC end-points need to support the option of sending different
media on different RTP sessions, so they can interwork with such types of media on different RTP sessions, so they can interwork
legacy devices. This is discussed further in Section 4.4. with such legacy devices. This is discussed further in
Section 4.4.
To provide enhanced quality of service: Some network-based quality To provide enhanced quality of service: Some network-based quality
of service mechanisms operate on the granularity of UDP 5-tuples. of service mechanisms operate on the granularity of transport
If it is desired to use these mechanisms to provide differentiated layer flows. If it is desired to use these mechanisms to provide
quality of service for some RTP flows, then those RTP flows need differentiated quality of service for some RTP packet streams,
to be sent in a separate RTP session using a different UDP port then those RTP packet streams need to be sent in a separate RTP
number, and with appropriate quality of service marking. This is session using a different transport-layer flow, and with
discussed further in Section 12.1.3. appropriate quality of service marking. This is discussed further
in Section 12.1.3.
To separate media with different purposes: An end-point might want To separate media with different purposes: An end-point might want
to send media streams that have different purposes on different to send RTP packet streams that have different purposes on
RTP sessions, to make it easy for the peer device to distinguish different RTP sessions, to make it easy for the peer device to
them. For example, some centralised multiparty conferencing distinguish them. For example, some centralised multiparty
systems display the active speaker in high resolution, but show conferencing systems display the active speaker in high
low resolution "thumbnails" of other participants. Such systems resolution, but show low resolution "thumbnails" of other
might configure the end-points to send simulcast high- and low- participants. Such systems might configure the end-points to send
resolution versions of their video using separate RTP sessions, to simulcast high- and low-resolution versions of their video using
simplify the operation of the central mixer. In the WebRTC separate RTP sessions, to simplify the operation of the RTP
context this is currently possible to accomplished by establishing middlebox. In the WebRTC context this is currently possible to
multiple WebRTC MediaStreamTracks that have the same media source accomplished by establishing multiple WebRTC MediaStreamTracks
in one (or more) RTCPeerConnection. Each MediaStreamTrack is then that have the same media source in one (or more)
configured to deliver a particular media quality and thus media RTCPeerConnection. Each MediaStreamTrack is then configured to
bit-rate, and will produce an independently encoded version with deliver a particular media quality and thus media bit-rate, and
the codec parameters agreed specifically in the context of that will produce an independently encoded version with the codec
RTCPeerConnection. The central mixer can distinguish packets parameters agreed specifically in the context of that
RTCPeerConnection. The RTP middlebox can distinguish packets
corresponding to the low- and high-resolution streams by corresponding to the low- and high-resolution streams by
inspecting their SSRC, RTP payload type, or some other information inspecting their SSRC, RTP payload type, or some other information
contained in RTP payload, RTP header extension or RTCP packets, contained in RTP payload, RTP header extension or RTCP packets,
but it can be easier to distinguish the flows if they arrive on but it can be easier to distinguish the RTP packet streams if they
separate RTP sessions on separate UDP ports. arrive on separate RTP sessions on separate transport-layer flows.
To directly connect with multiple peers: A multi-party conference To directly connect with multiple peers: A multi-party conference
does not need to use a central mixer. Rather, a multi-unicast does not need to use an RTP middlebox. Rather, a multi-unicast
mesh can be created, comprising several distinct RTP sessions, mesh can be created, comprising several distinct RTP sessions,
with each participant sending RTP traffic over a separate RTP with each participant sending RTP traffic over a separate RTP
session (that is, using an independent RTCPeerConnection object) session (that is, using an independent RTCPeerConnection object)
to every other participant, as shown in Figure 1. This topology to every other participant, as shown in Figure 1. This topology
has the benefit of not requiring a central mixer node that is has the benefit of not requiring an RTP middlebox node that is
trusted to access and manipulate the media data. The downside is trusted to access and manipulate the media data. The downside is
that it increases the used bandwidth at each sender by requiring that it increases the used bandwidth at each sender by requiring
one copy of the RTP media streams for each participant that are one copy of the RTP packet streams for each participant that are
part of the same session beyond the sender itself. part of the same session beyond the sender itself.
+---+ +---+ +---+ +---+
| A |<--->| B | | A |<--->| B |
+---+ +---+ +---+ +---+
^ ^ ^ ^
\ / \ /
\ / \ /
v v v v
+---+ +---+
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peers, thus not forcing B to endure the same quality reductions if peers, thus not forcing B to endure the same quality reductions if
there are limitations in the transport from A to C as C will. It there are limitations in the transport from A to C as C will. It
it believed that these advantages outweigh the limitations in it believed that these advantages outweigh the limitations in
debugging power. debugging power.
To indirectly connect with multiple peers: A common scenario in To indirectly connect with multiple peers: A common scenario in
multi-party conferencing is to create indirect connections to multi-party conferencing is to create indirect connections to
multiple peers, using an RTP mixer, translator, or some other type multiple peers, using an RTP mixer, translator, or some other type
of RTP middlebox. Figure 2 outlines a simple topology that might of RTP middlebox. Figure 2 outlines a simple topology that might
be used in a four-person centralised conference. The middlebox be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP media streams from acts to optimise the transmission of RTP packet streams from
certain perspectives, either by only sending some of the received certain perspectives, either by only sending some of the received
RTP media stream to any given receiver, or by providing a combined RTP packet stream to any given receiver, or by providing a
RTP media stream out of a set of contributing streams. combined RTP packet stream out of a set of contributing streams.
+---+ +-------------+ +---+ +---+ +-------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+ +---+ | RTP mixer, | +---+
| translator, | | translator, |
| or other | | or other |
+---+ | middlebox | +---+ +---+ | middlebox | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +-------------+ +---+ +---+ +-------------+ +---+
Figure 2: RTP mixer with only unicast paths Figure 2: RTP mixer with only unicast paths
There are various methods of implementation for the middlebox. If There are various methods of implementation for the middlebox. If
implemented as a standard RTP mixer or translator, a single RTP implemented as a standard RTP mixer or translator, a single RTP
session will extend across the middlebox and encompass all the session will extend across the middlebox and encompass all the
end-points in one multi-party session. Other types of middlebox end-points in one multi-party session. Other types of middlebox
might use separate RTP sessions between each end-point and the might use separate RTP sessions between each end-point and the
middlebox. A common aspect is that these central nodes can use a middlebox. A common aspect is that these RTP middleboxes can use
number of tools to control the media encoding provided by a WebRTC a number of tools to control the media encoding provided by a
end-point. This includes functions like requesting breaking the WebRTC end-point. This includes functions like requesting
encoding chain and have the encoder produce a so called Intra breaking the encoding chain and have the encoder produce a so
frame. Another is limiting the bit-rate of a given stream to called Intra frame. Another is limiting the bit-rate of a given
better suit the mixer view of the multiple down-streams. Others stream to better suit the mixer view of the multiple down-streams.
are controlling the most suitable frame-rate, picture resolution, Others are controlling the most suitable frame-rate, picture
the trade-off between frame-rate and spatial quality. The resolution, the trade-off between frame-rate and spatial quality.
middlebox gets the significant responsibility to correctly perform The middlebox gets the significant responsibility to correctly
congestion control, source identification, manage synchronisation perform congestion control, source identification, manage
while providing the application with suitable media optimizations. synchronisation while providing the application with suitable
The middlebox is also has to be a trusted node when it comes to media optimizations. The middlebox is also has to be a trusted
security, since it manipulates either the RTP header or the media node when it comes to security, since it manipulates either the
itself (or both) received from one end-point, before sending it on RTP header or the media itself (or both) received from one end-
towards the end-point(s), thus they need to be able to decrypt and point, before sending it on towards the end-point(s), thus they
then encrypt it before sending it out. need to be able to decrypt and then encrypt it before sending it
out.
RTP Mixers can create a situation where an end-point experiences a RTP Mixers can create a situation where an end-point experiences a
situation in-between a session with only two end-points and situation in-between a session with only two end-points and
multiple RTP sessions. Mixers are expected to not forward RTCP multiple RTP sessions. Mixers are expected to not forward RTCP
reports regarding RTP media streams across themselves. This is reports regarding RTP packet streams across themselves. This is
due to the difference in the RTP media streams provided to the due to the difference in the RTP packet streams provided to the
different end-points. The original media source lacks information different end-points. The original media source lacks information
about a mixer's manipulations prior to sending it the different about a mixer's manipulations prior to sending it the different
receivers. This scenario also results in that an end-point's receivers. This scenario also results in that an end-point's
feedback or requests goes to the mixer. When the mixer can't act feedback or requests goes to the mixer. When the mixer can't act
on this by itself, it is forced to go to the original media source on this by itself, it is forced to go to the original media source
to fulfil the receivers request. This will not necessarily be to fulfil the receivers request. This will not necessarily be
explicitly visible any RTP and RTCP traffic, but the interactions explicitly visible any RTP and RTCP traffic, but the interactions
and the time to complete them will indicate such dependencies. and the time to complete them will indicate such dependencies.
Providing source authentication in multi-party scenarios is a Providing source authentication in multi-party scenarios is a
skipping to change at page 30, line 26 skipping to change at page 33, line 22
each others' master keys, and can thus inject packets claimed to each others' master keys, and can thus inject packets claimed to
come from another end-point in the session. Any node performing come from another end-point in the session. Any node performing
relay can perform non-cryptographic mitigation by preventing relay can perform non-cryptographic mitigation by preventing
forwarding of packets that have SSRC fields that came from other forwarding of packets that have SSRC fields that came from other
end-points before. For cryptographic verification of the source end-points before. For cryptographic verification of the source
SRTP would require additional security mechanisms, for example SRTP would require additional security mechanisms, for example
TESLA for SRTP [RFC4383], that are not part of the base WebRTC TESLA for SRTP [RFC4383], that are not part of the base WebRTC
standards. standards.
To forward media between multiple peers: It is sometimes desirable To forward media between multiple peers: It is sometimes desirable
for an end-point that receives an RTP media stream to be able to for an end-point that receives an RTP packet stream to be able to
forward that media stream to a third party. The are some obvious forward that RTP packet stream to a third party. The are some
security and privacy implications in supporting this, but also obvious security and privacy implications in supporting this, but
potential uses. This is supported in the W3C API by taking the also potential uses. This is supported in the W3C API by taking
received and decoded media and using it as media source that is the received and decoded media and using it as media source that
re-encoding and transmitted as a new stream. is re-encoding and transmitted as a new stream.
At the RTP layer, media forwarding acts as a back-to-back RTP At the RTP layer, media forwarding acts as a back-to-back RTP
receiver and RTP sender. The receiving side terminates the RTP receiver and RTP sender. The receiving side terminates the RTP
session and decodes the media, while the sender side re-encodes session and decodes the media, while the sender side re-encodes
and transmits the media using an entirely separate RTP session. and transmits the media using an entirely separate RTP session.
The original sender will only see a single receiver of the media, The original sender will only see a single receiver of the media,
and will not be able to tell that forwarding is happening based on and will not be able to tell that forwarding is happening based on
RTP-layer information since the RTP session that is used to send RTP-layer information since the RTP session that is used to send
the forwarded media is not connected to the RTP session on which the forwarded media is not connected to the RTP session on which
the media was received by the node doing the forwarding. the media was received by the node doing the forwarding.
The end-point that is performing the forwarding is responsible for The end-point that is performing the forwarding is responsible for
producing an RTP media stream suitable for onwards transmission. producing an RTP packet stream suitable for onwards transmission.
The outgoing RTP session that is used to send the forwarded media The outgoing RTP session that is used to send the forwarded media
is entirely separate to the RTP session on which the media was is entirely separate to the RTP session on which the media was
received. This will require media transcoding for congestion received. This will require media transcoding for congestion
control purpose to produce a suitable bit-rate for the outgoing control purpose to produce a suitable bit-rate for the outgoing
RTP session, reducing media quality and forcing the forwarding RTP session, reducing media quality and forcing the forwarding
end-point to spend the resource on the transcoding. The media end-point to spend the resource on the transcoding. The media
transcoding does result in a separation of the two different legs transcoding does result in a separation of the two different legs
removing almost all dependencies, and allowing the forwarding end- removing almost all dependencies, and allowing the forwarding end-
point to optimize its media transcoding operation. The cost is point to optimize its media transcoding operation. The cost is
greatly increased computational complexity on the forwarding node. greatly increased computational complexity on the forwarding node.
Receivers of the forwarded stream will see the forwarding device Receivers of the forwarded stream will see the forwarding device
as the sender of the stream, and will not be able to tell from the as the sender of the stream, and will not be able to tell from the
RTP layer that they are receiving a forwarded stream rather than RTP layer that they are receiving a forwarded stream rather than
an entirely new media stream generated by the forwarding device. an entirely new RTP packet stream generated by the forwarding
device.
12.1.3. Differentiated Treatment of Flows 12.1.3. Differentiated Treatment of RTP Packet Streams
There are use cases for differentiated treatment of RTP media There are use cases for differentiated treatment of RTP packet
streams. Such differentiation can happen at several places in the streams. Such differentiation can happen at several places in the
system. First of all is the prioritization within the end-point system. First of all is the prioritization within the end-point
sending the media, which controls, both which RTP media streams that sending the media, which controls, both which RTP packet streams that
will be sent, and their allocation of bit-rate out of the current will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control. available aggregate as determined by the congestion control.
It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will
allow the application to indicate relative priorities for different allow the application to indicate relative priorities for different
MediaStreamTracks. These priorities can then be used to influence MediaStreamTracks. These priorities can then be used to influence
the local RTP processing, especially when it comes to congestion the local RTP processing, especially when it comes to congestion
control response in how to divide the available bandwidth between the control response in how to divide the available bandwidth between the
RTP flows. Any changes in relative priority will also need to be RTP packet streams. Any changes in relative priority will also need
considered for RTP flows that are associated with the main RTP flows, to be considered for RTP packet streams that are associated with the
such as RTP retransmission streams and FEC. The importance of such main RTP packet streams, such as redundant streams for RTP
associated RTP traffic flows is dependent on the media type and codec retransmission and FEC. The importance of such redundant RTP packet
used, in regards to how robust that codec is to packet loss. streams is dependent on the media type and codec used, in regards to
However, a default policy might to be to use the same priority for how robust that codec is to packet loss. However, a default policy
associated RTP flows as for the primary RTP flow. might to be to use the same priority for redundant RTP packet stream
as for the source RTP packet stream.
Secondly, the network can prioritize packet flows, including RTP Secondly, the network can prioritize transport-layer flows and sub-
media streams. Typically, differential treatment includes two steps, flows, including RTP packet streams. Typically, differential
the first being identifying whether an IP packet belongs to a class treatment includes two steps, the first being identifying whether an
that has to be treated differently, the second the actual mechanism IP packet belongs to a class that has to be treated differently, the
to prioritize packets. This is done according to three methods: second the actual mechanism to prioritize packets. This is done
according to three methods:
DiffServ: The end-point marks a packet with a DiffServ code point to DiffServ: The end-point marks a packet with a DiffServ code point to
indicate to the network that the packet belongs to a particular indicate to the network that the packet belongs to a particular
class. class.
Flow based: Packets that need to be given a particular treatment are Flow based: Packets that need to be given a particular treatment are
identified using a combination of IP and port address. identified using a combination of IP and port address.
Deep Packet Inspection: A network classifier (DPI) inspects the Deep Packet Inspection: A network classifier (DPI) inspects the
packet and tries to determine if the packet represents a packet and tries to determine if the packet represents a
particular application and type that is to be prioritized. particular application and type that is to be prioritized.
Flow-based differentiation will provide the same treatment to all Flow-based differentiation will provide the same treatment to all
packets within a flow, i.e., relative prioritization is not possible. packets within a transport-layer flow, i.e., relative prioritization
Moreover, if the resources are limited it might not be possible to is not possible. Moreover, if the resources are limited it might not
provide differential treatment compared to best-effort for all the be possible to provide differential treatment compared to best-effort
flows in a WebRTC application. When flow-based differentiation is for all the RTP packet streams in a WebRTC application. When flow-
available the WebRTC application needs to know about it so that it based differentiation is available the WebRTC application needs to
can provide the separation of the RTP media streams onto different know about it so that it can provide the separation of the RTP packet
UDP flows to enable a more granular usage of flow based streams onto different UDP flows to enable a more granular usage of
differentiation. That way at least providing different flow based differentiation. That way at least providing different
prioritization of audio and video if desired by application. prioritization of audio and video if desired by application.
DiffServ assumes that either the end-point or a classifier can mark DiffServ assumes that either the end-point or a classifier can mark
the packets with an appropriate DSCP so that the packets are treated the packets with an appropriate DSCP so that the packets are treated
according to that marking. If the end-point is to mark the traffic according to that marking. If the end-point is to mark the traffic
two requirements arise in the WebRTC context: 1) The WebRTC two requirements arise in the WebRTC context: 1) The WebRTC
application or browser has to know which DSCP to use and that it can application or browser has to know which DSCP to use and that it can
use them on some set of RTP media streams. 2) The information needs use them on some set of RTP packet streams. 2) The information needs
to be propagated to the operating system when transmitting the to be propagated to the operating system when transmitting the
packet. Details of this process are outside the scope of this memo packet. Details of this process are outside the scope of this memo
and are further discussed in "DSCP and other packet markings for and are further discussed in "DSCP and other packet markings for
RTCWeb QoS" [I-D.dhesikan-tsvwg-rtcweb-qos]. RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos].
For packet based marking schemes it might be possible to mark For packet based marking schemes it might be possible to mark
individual RTP packets differently based on the relative priority of individual RTP packets differently based on the relative priority of
the RTP payload. For example video codecs that have I, P, and B the RTP payload. For example video codecs that have I, P, and B
pictures could prioritise any payloads carrying only B frames less, pictures could prioritise any payloads carrying only B frames less,
as these are less damaging to loose. However, depending on the QoS as these are less damaging to loose. However, depending on the QoS
mechanism and what markings that are applied, this can result in not mechanism and what markings that are applied, this can result in not
only different packet drop probabilities but also packet reordering, only different packet drop probabilities but also packet reordering,
see [I-D.dhesikan-tsvwg-rtcweb-qos] for further discussion. As see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As default
default policy all RTP packets related to a media stream ought to be policy all RTP packets related to a RTP packet stream ought to be
provided with the same prioritization; per-packet prioritization is provided with the same prioritization; per-packet prioritization is
outside the scope of this memo, but might be specified elsewhere in outside the scope of this memo, but might be specified elsewhere in
future. future.
It is also important to consider how RTCP packets associated with a It is also important to consider how RTCP packets associated with a
particular RTP media flow need to be marked. RTCP compound packets particular RTP packet stream need to be marked. RTCP compound
with Sender Reports (SR), ought to be marked with the same priority packets with Sender Reports (SR), ought to be marked with the same
as the RTP media flow itself, so the RTCP-based round-trip time (RTT) priority as the RTP packet stream itself, so the RTCP-based round-
measurements are done using the same flow priority as the media flow trip time (RTT) measurements are done using the same transport-layer
experiences. RTCP compound packets containing RR packet ought to be flow priority as the RTP packet stream experiences. RTCP compound
sent with the priority used by the majority of the RTP media flows packets containing RR packet ought to be sent with the priority used
reported on. RTCP packets containing time-critical feedback packets by the majority of the RTP packet streams reported on. RTCP packets
can use higher priority to improve the timeliness and likelihood of containing time-critical feedback packets can use higher priority to
delivery of such feedback. improve the timeliness and likelihood of delivery of such feedback.
12.2. Source, Flow, and Participant Identification 12.2. Media Source, RTP Packet Streams, and Participant Identification
12.2.1. Media Streams 12.2.1. Media Source
Each RTP media stream is identified by a unique synchronisation Each RTP packet stream is identified by a unique synchronisation
source (SSRC) identifier. The SSRC identifier is carried in the RTP source (SSRC) identifier. The SSRC identifier is carried in each of
data packets comprising a media stream, and is also used to identify the RTP packets comprising a RTP packet stream, and is also used to
that stream in the corresponding RTCP reports. The SSRC is chosen as identify that stream in the corresponding RTCP reports. The SSRC is
discussed in Section 4.8. The first stage in demultiplexing RTP and chosen as discussed in Section 4.8. The first stage in
RTCP packets received at a WebRTC end-point is to separate the media demultiplexing RTP and RTCP packets received on a single transport
layer flow at a WebRTC end-point is to separate the RTP packet
streams based on their SSRC value; once that is done, additional streams based on their SSRC value; once that is done, additional
demultiplexing steps can determine how and where to render the media. demultiplexing steps can determine how and where to render the media.
RTP allows a mixer, or other RTP-layer middlebox, to combine media RTP allows a mixer, or other RTP-layer middlebox, to combine encoded
flows from multiple sources to form a new media flow. The RTP data streams from multiple media sources to form a new encoded stream from
packets in that new flow can include a Contributing Source (CSRC) a new media source (the mixer). The RTP packets in that new RTP
list, indicating which original SSRCs contributed to the combined packet stream can include a Contributing Source (CSRC) list,
packet. As described in Section 4.1, implementations need to support indicating which original SSRCs contributed to the combined source
stream. As described in Section 4.1, implementations need to support
reception of RTP data packets containing a CSRC list and RTCP packets reception of RTP data packets containing a CSRC list and RTCP packets
that relate to sources present in the CSRC list. The CSRC list can that relate to sources present in the CSRC list. The CSRC list can
change on a packet-by-packet basis, depending on the mixing operation change on a packet-by-packet basis, depending on the mixing operation
being performed. Knowledge of what sources contributed to a being performed. Knowledge of what media sources contributed to a
particular RTP packet can be important if the user interface particular RTP packet can be important if the user interface
indicates which participants are active in the session. Changes in indicates which participants are active in the session. Changes in
the CSRC list included in packets needs to be exposed to the WebRTC the CSRC list included in packets needs to be exposed to the WebRTC
application using some API, if the application is to be able to track application using some API, if the application is to be able to track
changes in session participation. It is desirable to map CSRC values changes in session participation. It is desirable to map CSRC values
back into WebRTC MediaStream identities as they cross this API, to back into WebRTC MediaStream identities as they cross this API, to
avoid exposing the SSRC/CSRC name space to JavaScript applications. avoid exposing the SSRC/CSRC name space to JavaScript applications.
If the mixer-to-client audio level extension [RFC6465] is being used If the mixer-to-client audio level extension [RFC6465] is being used
in the session (see Section 5.2.3), the information in the CSRC list in the session (see Section 5.2.3), the information in the CSRC list
is augmented by audio level information for each contributing source. is augmented by audio level information for each contributing source.
This information can usefully be exposed in the user interface. This information can usefully be exposed in the user interface.
12.2.2. Media Streams: SSRC Collision Detection 12.2.2. SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have The RTP standard [RFC3550] requires any RTP implementation to have
support for detecting and handling SSRC collisions, i.e., resolve the support for detecting and handling SSRC collisions, i.e., resolve the
conflict when two different end-points use the same SSRC value. This conflict when two different end-points use the same SSRC value. This
requirement also applies to WebRTC end-points. There are several requirement also applies to WebRTC end-points. There are several
scenarios where SSRC collisions can occur. scenarios where SSRC collisions can occur:
In a point-to-point session where each SSRC is associated with either o In a point-to-point session where each SSRC is associated with
of the two end-points and where the main media carrying SSRC either of the two end-points and where the main media carrying
identifier will be announced in the signalling channel, a collision SSRC identifier will be announced in the signalling channel, a
is less likely to occur due to the information about used SSRCs collision is less likely to occur due to the information about
provided by Source-Specific SDP Attributes [RFC5576]. Still if both used SSRCs provided by Source-Specific SDP Attributes [RFC5576].
end-points start uses an new SSRC identifier prior to having
signalled it to the peer and received acknowledgement on the
signalling message, there can be collisions. The Source-Specific SDP
Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
or reject a end-points usage of an SSRC.
There could also appear SSRC values that are not signalled. This is Still, collisions can occur if both end-points start uses an new
more likely than it appears as certain RTP functions need extra SSRCs SSRC identifier prior to having signalled it to the peer and
to provide functionality related to another (the "main") SSRC, for received acknowledgement on the signalling message. The Source-
example, SSRC multiplexed RTP retransmission [RFC4588]. In those Specific SDP Attributes [RFC5576] contains no mechanism to resolve
cases, an end-point can create a new SSRC that strictly doesn't need SSRC collisions or reject a end-points usage of an SSRC.
to be announced over the signalling channel to function correctly on
o SSRC values that have not been signalled could also appear in an
RTP session. This is more likely than it appears, since some RTP
functions use extra SSRCs to provide their functionality. For
example, retransmission data might be transmitted using a separate
RTP packet stream that requires its own SSRC, separate to the SSRC
of the source RTP packet stream [RFC4588]. In those cases, an
end-point can create a new SSRC that strictly doesn't need to be
announced over the signalling channel to function correctly on
both RTP and RTCPeerConnection level. both RTP and RTCPeerConnection level.
The more likely case for SSRC collision is that multiple end-points o Multiple end-points in a multiparty conference can create new
in a multiparty conference create new sources and signals those sources and signal those towards the RTP middlebox. In cases
towards the central server. In cases where the SSRC/CSRC are where the SSRC/CSRC are propagated between the different end-
propagated between the different end-points from the central node points from the RTP middlebox collisions can occur.
collisions can occur.
Another scenario is when the central node manages to connect an end- o An RTP middlebox could connect an end-point's RTCPeerConnection to
point's RTCPeerConnection to another RTCPeerConnection the end-point another RTCPeerConnection from the same end-point, thus forming a
already has, thus forming a loop where the end-point will receive its loop where the end-point will receive its own traffic. While is
own traffic. While is is clearly considered a bug, it is important is clearly considered a bug, it is important that the end-point is
that the end-point is able to recognise and handle the case when it able to recognise and handle the case when it occurs. This case
occurs. This case becomes even more problematic when media mixers, becomes even more problematic when media mixers, and so on, are
and so on, are involved, where the stream received is a different involved, where the stream received is a different stream but
stream but still contains this client's input. still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long These SSRC/CSRC collisions can only be handled on RTP level as long
as the same RTP session is extended across multiple as the same RTP session is extended across multiple
RTCPeerConnections by a RTP middlebox. To resolve the more generic RTCPeerConnections by a RTP middlebox. To resolve the more generic
case where multiple RTCPeerConnections are interconnected, then case where multiple RTCPeerConnections are interconnected, then
identification of the media source(s) part of a MediaStreamTrack identification of the media source(s) part of a MediaStreamTrack
being propagated across multiple interconnected RTCPeerConnection being propagated across multiple interconnected RTCPeerConnection
needs to be preserved across these interconnections. needs to be preserved across these interconnections.
12.2.3. Media Synchronisation Context 12.2.3. Media Synchronisation Context
When an end-point sends media from more than one media source, it When an end-point sends media from more than one media source, it
needs to consider if (and which of) these media sources are to be needs to consider if (and which of) these media sources are to be
synchronized. In RTP/RTCP, synchronisation is provided by having a synchronized. In RTP/RTCP, synchronisation is provided by having a
set of RTP media streams be indicated as coming from the same set of RTP packet streams be indicated as coming from the same
synchronisation context and logical end-point by using the same RTCP synchronisation context and logical end-point by using the same RTCP
CNAME identifier. CNAME identifier.
The next provision is that the internal clocks of all media sources, The next provision is that the internal clocks of all media sources,
i.e., what drives the RTP timestamp, can be correlated to a system i.e., what drives the RTP timestamp, can be correlated to a system
clock that is provided in RTCP Sender Reports encoded in an NTP clock that is provided in RTCP Sender Reports encoded in an NTP
format. By correlating all RTP timestamps to a common system clock format. By correlating all RTP timestamps to a common system clock
for all sources, the timing relation of the different RTP media for all sources, the timing relation of the different RTP packet
streams, also across multiple RTP sessions can be derived at the streams, also across multiple RTP sessions can be derived at the
receiver and, if desired, the streams can be synchronized. The receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not. information; it is up to the receiver to use it or not.
13. Security Considerations 13. Security Considerations
The overall security architecture for WebRTC is described in The overall security architecture for WebRTC is described in
[I-D.ietf-rtcweb-security-arch], and security considerations for the [I-D.ietf-rtcweb-security-arch], and security considerations for the
WebRTC framework are described in [I-D.ietf-rtcweb-security]. These WebRTC framework are described in [I-D.ietf-rtcweb-security]. These
considerations apply to this memo also. considerations also apply to this memo.
The security considerations of the RTP specification, the RTP/SAVPF The security considerations of the RTP specification, the RTP/SAVPF
profile, and the various RTP/RTCP extensions and RTP payload formats profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply. that form the complete protocol suite described in this memo apply.
We do not believe there are any new security considerations resulting We do not believe there are any new security considerations resulting
from the combination of these various protocol extensions. from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control The Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
handling of fundamental issues by offering confidentiality, integrity handling of fundamental issues by offering confidentiality, integrity
and partial source authentication. A mandatory to implement media and partial source authentication. A mandatory to implement media
security solution is created by combing this secured RTP profile and security solution is created by combing this secured RTP profile and
DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of
[I-D.ietf-rtcweb-security-arch]. [I-D.ietf-rtcweb-security-arch].
RTCP packets convey a Canonical Name (CNAME) identifier that is used RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate media flows that need to be synchronised across related to associate RTP packet streams that need to be synchronised across
RTP sessions. Inappropriate choice of CNAME values can be a privacy related RTP sessions. Inappropriate choice of CNAME values can be a
concern, since long-term persistent CNAME identifiers can be used to privacy concern, since long-term persistent CNAME identifiers can be
track users across multiple WebRTC calls. Section 4.9 of this memo used to track users across multiple WebRTC calls. Section 4.9 of
provides guidelines for generation of untraceable CNAME values that this memo provides guidelines for generation of untraceable CNAME
alleviate this risk. values that alleviate this risk.
The guidelines in [RFC6562] apply when using variable bit rate (VBR) The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs such as Opus (see Section 4.3 for discussion of mandated audio codecs such as Opus (see Section 4.3 for discussion of mandated
audio codecs). These guidelines in [RFC6562] also apply, but are of audio codecs). The guidelines in [RFC6562] also apply, but are of
lesser importance, when using the client-to-mixer audio level header lesser importance, when using the client-to-mixer audio level header
extensions (Section 5.2.2) or the mixer-to-client audio level header extensions (Section 5.2.2) or the mixer-to-client audio level header
extensions (Section 5.2.3). extensions (Section 5.2.3). The use of the encryption of the header
extensions are RECOMMENDED, unless there are known reasons, like RTP
middleboxes or third party monitoring that will greatly benefit from
the information, and this has been expressed using API or signalling.
If further evidence are produced to show that information leakage is
significant from audio level indications, then use of encryption
needs to be mandated at that time.
14. IANA Considerations 14. IANA Considerations
This memo makes no request of IANA. This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as Note to RFC Editor: this section is to be removed on publication as
an RFC. an RFC.
15. Acknowledgements 15. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Charles Eckel, Cullen Jennings, Dan Romascanu, and the Cary Bran, Charles Eckel, Christian Groves, Cullen Jennings, Dan
other members of the IETF RTCWEB working group for their valuable Romascanu, Martin Thomson, and the other members of the IETF RTCWEB
feedback. working group for their valuable feedback.
16. References 16. References
16.1. Normative References 16.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session] [I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft- Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-04 (work in ietf-avtcore-multi-media-rtp-session-05 (work in
progress), January 2014. progress), February 2014.
[I-D.ietf-avtcore-rtp-circuit-breakers] [I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control: Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf- Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-04 (work in progress), avtcore-rtp-circuit-breakers-05 (work in progress),
January 2014. February 2014.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ", Grouping RTCP Reception Statistics and Other Feedback ",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work
in progress), July 2013. in progress), February 2014.
[I-D.ietf-avtcore-rtp-multi-stream] [I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session", "Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-02 (work in progress), draft-ietf-avtcore-rtp-multi-stream-03 (work in progress),
January 2014. February 2014.
[I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", draft-ietf-avtext-
multiple-clock-rates-11 (work in progress), November 2013.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-08 (work in progress), January 2014. rtcweb-security-arch-09 (work in progress), February 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014. ietf-rtcweb-security-06 (work in progress), January 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, December Payload Format Specifications", BCP 36, RFC 2736, December
skipping to change at page 37, line 28 skipping to change at page 40, line 36
July 2003. July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 2003. 3556, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, March 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
skipping to change at page 38, line 44 skipping to change at page 42, line 9
[RFC7007] Terriberry, T., "Update to Remove DVI4 from the [RFC7007] Terriberry, T., "Update to Remove DVI4 from the
Recommended Codecs for the RTP Profile for Audio and Video Recommended Codecs for the RTP Profile for Audio and Video
Conferences with Minimal Control (RTP/AVP)", RFC 7007, Conferences with Minimal Control (RTP/AVP)", RFC 7007,
August 2013. August 2013.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP) "Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013. Canonical Names (CNAMEs)", RFC 7022, September 2013.
[RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC
7164, March 2014.
[W3C.WD-mediacapture-streams-20130903] [W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A. Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/WD-mediacapture- 2013, <http://www.w3.org/TR/2013/
streams-20130903>. WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910] [W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A. Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD- Browsers", World Wide Web Consortium WD WD-
webrtc-20130910, September 2013, webrtc-20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>. <http://www.w3.org/TR/2013/WD-webrtc-20130910>.
16.2. Informative References 16.2. Informative References
[I-D.dhesikan-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-dhesikan-
tsvwg-rtcweb-qos-04 (work in progress), January 2014.
[I-D.ietf-avtcore-multiplex-guidelines] [I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Perkins, C., and H. Alvestrand, Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP to "Guidelines for using the Multiplexing Features of RTP to
Support Multiple Media Streams", draft-ietf-avtcore- Support Multiple Media Streams", draft-ietf-avtcore-
multiplex-guidelines-02 (work in progress), January 2014. multiplex-guidelines-02 (work in progress), January 2014.
[I-D.ietf-avtcore-rtp-topologies-update] [I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft- Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-01 (work in progress), ietf-avtcore-rtp-topologies-update-01 (work in progress),
October 2013. October 2013.
[I-D.ietf-avtext-rtp-grouping-taxonomy] [I-D.ietf-avtext-rtp-grouping-taxonomy]
Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
"A Taxonomy of Grouping Semantics and Mechanisms for Real- "A Taxonomy of Grouping Semantics and Mechanisms for Real-
Time Transport Protocol (RTP) Sources", draft-ietf-avtext- Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
rtp-grouping-taxonomy-00 (work in progress), November rtp-grouping-taxonomy-01 (work in progress), February
2013. 2014.
[I-D.ietf-mmusic-msid] [I-D.ietf-mmusic-msid]
Alvestrand, H., "WebRTC MediaStream Identification in the Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", draft-ietf-mmusic-msid-04 Session Description Protocol", draft-ietf-mmusic-msid-05
(work in progress), February 2014. (work in progress), March 2014.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description "Negotiating Media Multiplexing Using the Session
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
bundle-negotiation-05 (work in progress), October 2013. negotiation-07 (work in progress), April 2014.
[I-D.ietf-payload-rtp-howto] [I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format", Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-13 (work in progress), draft-ietf-payload-rtp-howto-13 (work in progress),
January 2014. January 2014.
[I-D.ietf-rmcat-cc-requirements] [I-D.ietf-rmcat-cc-requirements]
Jesup, R., "Congestion Control Requirements For RMCAT", Jesup, R., "Congestion Control Requirements For RMCAT",
draft-ietf-rmcat-cc-requirements-02 (work in progress), draft-ietf-rmcat-cc-requirements-04 (work in progress),
February 2014. April 2014.
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-05 (work in Requirements", draft-ietf-rtcweb-audio-05 (work in
progress), February 2014. progress), February 2014.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-08 (work based Applications", draft-ietf-rtcweb-overview-09 (work
in progress), September 2013. in progress), February 2014.
[I-D.ietf-rtcweb-use-cases-and-requirements] [I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft- Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-14 (work in ietf-rtcweb-use-cases-and-requirements-14 (work in
progress), February 2014. progress), February 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-tsvwg-
rtcweb-qos-00 (work in progress), April 2014.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003. 2003.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006. Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion Datagram Congestion Control Protocol (DCCP) Congestion
 End of changes. 161 change blocks. 
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