draft-ietf-rtcweb-rtp-usage-13.txt   draft-ietf-rtcweb-rtp-usage-14.txt 
RTCWEB Working Group C. 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: October 25, 2014 Ericsson Expires: November 17, 2014 Ericsson
J. Ott J. Ott
Aalto University Aalto University
April 23, 2014 May 16, 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-13 draft-ietf-rtcweb-rtp-usage-14
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-
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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 October 25, 2014. This Internet-Draft will expire on November 17, 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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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 . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 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) . . . . . . 12
4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 12 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13
5.1. Conferencing Extensions and Topologies . . . . . . . . . 12 5.1. Conferencing Extensions and Topologies . . . . . . . . . 13
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 14 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 14 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 14 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 15
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 15 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 15 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 16 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 16 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 17 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 17
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 17 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 18
6.1. Negative Acknowledgements and RTP Retransmission . . . . 17 6.1. Negative Acknowledgements and RTP Retransmission . . . . 18
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 18 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 19
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20
7.2. RTCP Limitations for Congestion Control . . . . . . . . . 20 7.2. Congestion Control Interoperability and Legacy Systems . 21
7.3. Congestion Control Interoperability and Legacy Systems . 22 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 23 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 22
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 24 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 22
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 24 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 24
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25 12. RTP Implementation Considerations . . . . . . . . . . . . . . 26
12. RTP Implementation Considerations . . . . . . . . . . . . . . 28 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 26
12.1. Configuration and Use of RTP Sessions . . . . . . . . . 28 12.1.1. Use of Multiple Media Sources Within an RTP Session 26
12.1.1. Use of Multiple Media Sources Within an RTP Session 28 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 27
12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 29 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 32
12.1.3. Differentiated Treatment of RTP Packet Streams . . . 34
12.2. Media Source, RTP Packet Streams, and Participant 12.2. Media Source, RTP Packet Streams, and Participant
Identification . . . . . . . . . . . . . . . . . . . . . 35 Identification . . . . . . . . . . . . . . . . . . . . . 34
12.2.1. Media Source . . . . . . . . . . . . . . . . . . . . 36 12.2.1. Media Source Identification . . . . . . . . . . . . 34
12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 34
12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 12.2.3. Media Synchronisation Context . . . . . . . . . . . 36
13. Security Considerations . . . . . . . . . . . . . . . . . . . 38 13. Security Considerations . . . . . . . . . . . . . . . . . . . 36
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 37
16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 37
16.1. Normative References . . . . . . . . . . . . . . . . . . 39 16.1. Normative References . . . . . . . . . . . . . . . . . . 37
16.2. Informative References . . . . . . . . . . . . . . . . . 42 16.2. Informative References . . . . . . . . . . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42
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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Bi-directional Transport-layer Flow: A bi-directional transport- Bi-directional Transport-layer Flow: A bi-directional transport-
layer flow is a transport-layer flow that is symmetric. That is, layer flow is a transport-layer flow that is symmetric. That is,
the transport-layer flow in the reverse direction has a 5-tuple the transport-layer flow in the reverse direction has a 5-tuple
where the source and destination address and ports are swapped where the source and destination address and ports are swapped
compared to the forward path transport-layer flow, and the compared to the forward path transport-layer flow, and the
transport protocol is the same. transport protocol is the same.
This document uses the terminology from This document uses the terminology from
[I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used [I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used
according to their definitions from the RTP Specification [RFC3550]. according to their definitions from the RTP Specification [RFC3550].
We especially note the following frequently used terms: RTP Packet Especially note the following frequently used terms: RTP Packet
Stream, RTP Session, and End-point. 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.
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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-synchronisation; Reports, to allow receivers to implement lip-synchronisation; see
support for the rapid RTP synchronisation extensions (see Section 5.2.1 regarding support for the rapid RTP synchronisation
Section 5.2.1) is RECOMMENDED. extensions.
o Support for multiple synchronisation contexts. Participants that o Support for multiple synchronisation contexts. Participants that
send multiple simultaneous RTP packet streams SHOULD do so as part send multiple simultaneous RTP packet streams SHOULD do so as part
of 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. For compatibility with potential future synchronised manner. For compatibility with potential future
versions of this specification, or for interoperability with non- versions of this specification, or for interoperability with non-
WebRTC devices through a gateway, receivers MUST support multiple WebRTC devices through a gateway, receivers MUST support multiple
synchronisation contexts, indicated by the use of multiple RTCP synchronisation contexts, indicated by the use of multiple RTCP
CNAMEs in an RTP session. This specification requires the usage CNAMEs in an RTP session. This specification requires the usage
of a single CNAME when sending RTP Packet Streams in some of a single CNAME when sending RTP Packet Streams in some
circumstances, see Section 4.9. 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. Note that
implementations MUST ignore unknown RTCP packet types. Note that
additional RTCP Packet types are used 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 (Section 6.3.6 of
[RFC3550]) and reverse reconsideration (Section 6.3.4 of
[RFC3550]).
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 [RFC4566][RFC3556].
reporting interval described in Section 6.2 of [RFC3550] is
RECOMMENDED. o Support for the reduced minimum RTCP reporting interval described
in Section 6.2 of [RFC3550] is REQUIRED. When using the reduced
minimum RTCP reporting interval, the fixed (non-reduced) minimum
interval MUST be used when calculating the participant timeout
interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay
before sending the initial compound RTCP packet can be set to zero
(see Section 6.2 of [RFC3550] as updated by
[I-D.ietf-avtcore-rtp-multi-stream]).
o Ignore unknown RTCP packet types and RTP header extensions. This
to ensure robust handling of future extensions, middlebox
behaviours, etc., that can result in not signalled RTCP packet
types or RTP header extensions being received. If a compound RTCP
packet is received that contains a mixture of known and unknown
RTCP packet types, the known packets types need to be processed as
usual, with only the unknown packet types being discarded.
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
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End-points 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 packet stream with type number is sometimes used to associate an RTP packet stream with
a signalling context. This association is possible provided unique a signalling context. This association is possible provided unique
RTP payload type numbers are used in each context. For example, an RTP payload type numbers are used in each context. For example, an
RTP packet stream can be associated with an SDP "m=" line by RTP packet stream can be associated with an SDP "m=" line by
comparing the RTP payload type numbers used by the RTP packet stream comparing the RTP payload type numbers used by the RTP packet stream
with payload types signalled in the "a=rtpmap:" lines in the media with payload types signalled in the "a=rtpmap:" lines in the media
sections of the SDP. If RTP packet streams are being associated with sections of the SDP. This leads to the following considerations:
signalling contexts based on the RTP payload type, then the
assignment of RTP payload type numbers MUST be unique across If RTP packet streams are being associated with signalling
signalling contexts; if the same RTP payload format configuration is contexts based on the RTP payload type, then the assignment of RTP
used in multiple contexts, then a different RTP payload type number payload type numbers MUST be unique across signalling contexts.
has to be assigned in each context to ensure uniqueness. If the RTP
payload type number is not being used to associate RTP packet streams If the same RTP payload format configuration is used in multiple
with a signalling context, then the same RTP payload type number can contexts, then a different RTP payload type number has to be
be used to indicate the exact same RTP payload format configuration assigned in each context to ensure uniqueness.
in multiple contexts. A single RTP payload type number MUST NOT be
assigned to different RTP payload formats, or different If the RTP payload type number is not being used to associate RTP
configurations of the same RTP payload format, within a single RTP packet streams with a signalling context, then the same RTP
session (note that the different "m=" lines in an SDP bundle group payload type number can be used to indicate the exact same RTP
[I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session). 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 "m=" lines
in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form
a single RTP session).
An end-point that has signalled support for multiple RTP payload An end-point that has signalled support for multiple RTP payload
formats SHOULD be able to accept data in any of those payload formats formats MUST be able to accept data in any of those payload formats
at any time, unless it has previously signalled limitations on its at any time, unless it has previously signalled limitations on its
decoding capability. This requirement is constrained if several decoding capability. This requirement is constrained if several
types of media (e.g., audio and video) are sent in the same RTP types of media (e.g., audio and video) are sent in the same RTP
session. In such a case, a source (SSRC) is restricted to switching session. In such a case, a source (SSRC) is restricted to switching
only between the RTP payload formats signalled for the type of media only between the RTP payload formats signalled for the type of media
that is being sent by that source; see Section 4.4. To support rapid that is being sent by that source; see Section 4.4. To support rapid
rate adaptation by changing codec, RTP does not require advance rate adaptation by changing codec, RTP does not require advance
signalling for changes between RTP payload formats used by a single signalling for changes between RTP payload formats used by a single
SSRC that were signalled during session set-up. SSRC that were signalled during session set-up.
An RTP sender that changes between two RTP payload types that use If performing changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in different RTP clock rates, an RTP sender MUST follow the
Section 4.1 of [RFC7160]. RTP receivers MUST follow the recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST
recommendations in Section 4.3 of [RFC7160] in order to support follow the recommendations in Section 4.3 of [RFC7160] in order to
sources that switch between clock rates in an RTP session (these support sources that switch between clock rates in an RTP session
recommendations for receivers are backwards compatible with the case (these recommendations for receivers are backwards compatible with
where senders use only a single clock rate). the case 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 end-points communicating using RTP is An association amongst a set of end-points communicating using RTP is
known as an RTP session [RFC3550]. An end-point can be involved in known as an RTP session [RFC3550]. An end-point can be involved in
several RTP sessions at the same time. In a multimedia session, each several RTP sessions at the same time. In a multimedia session, each
type of media has typically been carried in a separate RTP session type of media has typically been carried in a separate RTP session
(e.g., using one RTP session for the audio, and a separate RTP (e.g., using one RTP session for the audio, and a separate RTP
session using a different transport-layer flow for the video). session using a different transport-layer flow for the video).
WebRTC implementations of RTP are REQUIRED to implement support for WebRTC implementations of RTP are REQUIRED to implement support for
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REQUIRED to support transport of all RTP packet streams, independent REQUIRED to support transport of all RTP packet streams, independent
of media type, in a single RTP session using a single transport layer of media type, in a single RTP session using a single transport layer
flow, according to [I-D.ietf-avtcore-multi-media-rtp-session]. If flow, according to [I-D.ietf-avtcore-multi-media-rtp-session]. If
multiple types of media are to be used in a single RTP session, all multiple types of media are to be used in a single RTP session, all
participants in that RTP session MUST agree to this usage. In an SDP participants in that RTP session MUST agree to this usage. In an SDP
context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to
signal such a bundle of RTP packet streams forming a single RTP signal such a bundle of RTP packet streams forming a single RTP
session. session.
Further discussion about the suitability of different RTP session Further discussion about the suitability of different RTP session
structures and multiplexing methods to different scenarios are structures and multiplexing methods to different scenarios can be
suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines]. 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
flows (e.g., two UDP ports for each RTP session, one port for RTP and flows (e.g., two UDP ports for each RTP session, one port for RTP and
one port for RTCP). With the increased use of Network Address/Port one port for RTCP). With the increased use of Network Address/Port
Translation (NAT/NAPT) this has become problematic, since maintaining Translation (NAT/NAPT) this has become problematic, since maintaining
multiple NAT bindings can be costly. It also complicates firewall multiple NAT bindings can be costly. It also complicates firewall
administration, since multiple ports need to be opened to allow RTP administration, since multiple ports need to be opened to allow RTP
traffic. To reduce these costs and session set-up times, support for traffic. To reduce these costs and session set-up times,
multiplexing RTP data packets and RTCP control packets on a single implementations are REQUIRED to support multiplexing RTP data packets
transport-layer flow for each RTP session is REQUIRED, provided it is and RTCP control packets on a single transport-layer flow [RFC5761].
negotiated in the signalling channel before use as specified in Such RTP and RTCP multiplexing MUST be negotiated in the signalling
[RFC5761]. For backwards compatibility, implementations are also channel before it is used. If SDP is used for signalling, this
REQUIRED to support RTP and RTCP sent on separate transport-layer negotiation MUST use the attributes defined in [RFC5761]. For
flows. backwards compatibility, implementations are also REQUIRED to support
RTP and RTCP sent on separate transport-layer flows.
Note that the use of RTP and RTCP multiplexed onto a single Note that the use of RTP and RTCP multiplexed onto a single
transport-layer flow ensures that there is occasional traffic sent on transport-layer flow ensures that there is occasional traffic sent on
that port, even if there is no active media traffic. This can be that port, even if there is no active media traffic. This can be
useful to keep NAT bindings alive, and is the recommend method for useful to keep NAT bindings alive [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. Implementations
compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be MUST support sending and receiving non-compound RTCP feedback packets
negotiated using the signalling channel before use. For backwards [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using
compatibility, implementations are also REQUIRED to support the use the signalling channel. If SDP is used for signalling, this
of compound RTCP feedback packets if the remote end-point does not negotiation MUST use the attributes defined in [RFC5506]. For
agree to the use of non-compound RTCP in the signalling exchange. backwards compatibility, implementations are also REQUIRED to support
the use of compound RTCP feedback packets if the remote end-point
does not 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 reason REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason
for using symmetric RTP is primarily to avoid issues with NATs and for using symmetric RTP is primarily to avoid issues with NATs and
Firewalls by ensuring that the send and receive RTP packet streams, Firewalls by ensuring that the send and receive RTP packet streams,
as well as RTCP, are actually bi-directional transport-layer flows. as well as RTCP, are actually bi-directional transport-layer flows.
This will keep alive the NAT and firewall pinholes, and help indicate This will keep alive the NAT and firewall pinholes, and help indicate
consent that the receive direction is a transport-layer flow the consent that the receive direction is a transport-layer flow the
intended recipient actually wants. In addition, it saves resources, intended recipient actually wants. In addition, it saves resources,
specifically ports at the end-points, but also in the network as NAT specifically ports at the end-points, but also in the network as NAT
mappings or firewall state is not unnecessary bloated. The amount of mappings or firewall state is not unnecessary bloated. The amount of
per flow QoS state kept in the network is also reduced. 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. If SDP is used, this MUST be done using
in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of
support the "previous-ssrc" source attribute defined in Section 6.2 [RFC5576] and the "previous-ssrc" source attribute defined in
of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be Section 6.2 of [RFC5576]; other per-SSRC attributes defined in
supported. [RFC5576] MAY be supported.
Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP While support for signalled SSRC identifiers is mandated, their use
session is OPTIONAL. Implementations MUST be prepared to accept RTP in an RTP session is OPTIONAL. Implementations MUST be prepared to
and RTCP packets using SSRCs that have not been explicitly signalled accept RTP and RTCP packets using SSRCs that have not been explicitly
ahead of time. Implementations MUST support random SSRC assignment, signalled ahead of time. Implementations MUST support random SSRC
and MUST support SSRC collision detection and resolution, according assignment, and MUST support SSRC collision detection and resolution,
to [RFC3550]. When using signalled SSRC values, collision detection according to [RFC3550]. When using signalled SSRC values, collision
MUST be performed as described in Section 5 of [RFC5576]. detection MUST be performed as described in Section 5 of [RFC5576].
It is often desirable to associate an RTP packet stream with a non- It is often desirable to associate an RTP packet stream with a non-
RTP 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 packet 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 packet stream are often sufficient to associate that packet RTP packet stream are often sufficient to associate that packet
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RTCPeerConnection within their common same-origin context. 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 4.10. Handling of Leap Seconds
The guidelines regarding handling of leap seconds to limit their The guidelines regarding handling of leap seconds to limit their
impact on RTP media playout and synchronization given in [RFC7164] impact on RTP media play-out and synchronization given in [RFC7164]
SHOULD be followed. 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.
skipping to change at page 13, line 18 skipping to change at page 13, line 45
While the use of IP multicast groups is popular in IPTV systems, the While the use of IP multicast groups is popular in IPTV systems, the
topologies based on RTP middleboxes are dominant in interactive video topologies based on RTP middleboxes are dominant in interactive video
conferencing environments. Topologies based on a mesh of unicast conferencing environments. Topologies based on a mesh of unicast
transport-layer flows to create a common RTP session have not seen transport-layer flows to create a common RTP session have not seen
widespread deployment to date. Accordingly, WebRTC implementations widespread deployment to date. Accordingly, WebRTC implementations
are not expected to support topologies based on IP multicast groups are not expected to support topologies based on IP multicast groups
or to support mesh-based topologies, such as a point-to-multipoint or to support mesh-based topologies, such as a point-to-multipoint
mesh configured as a single RTP session (Topo-Mesh in the terminology mesh configured as a single RTP session (Topo-Mesh in the terminology
of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to-
multipoint mesh constructed using several RTP sessions, implemented multipoint mesh constructed using several RTP sessions, implemented
in the WebRTC context using independent RTCPeerConnections, can be in the WebRTC context using independent RTCPeerConnections
expected to be utilised by WebRTC applications and needs to be [W3C.WD-webrtc-20130910], can be expected to be utilised by WebRTC
supported. applications and needs to be supported.
WebRTC implementations of RTP endpoints implemented according to this WebRTC implementations of RTP endpoints implemented according to this
memo are expected to support all the topologies described in memo are expected to support all the topologies described in
[I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
and receive unicast RTP packet streams to and from some peer device, and receive unicast RTP packet streams to and from some peer device,
provided that peer can participate in performing congestion control provided that peer can participate in performing congestion control
on the RTP packet streams. The peer device could be another RTP on the RTP packet streams. The peer device could be another RTP
endpoint, or it could be an RTP middlebox that redistributes the RTP endpoint, or it could be an RTP middlebox that redistributes the RTP
packet streams to other RTP endpoints. This limitation means that packet streams to other RTP endpoints. This limitation means that
some of the RTP middlebox-based topologies are not suitable for use some of the RTP middlebox-based topologies are not suitable for use
skipping to change at page 14, line 32 skipping to change at page 15, line 13
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 message is defined in Sections 3.5.1 and 4.3.1 The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1
of the Codec Control Messages [RFC5104]. It 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 FIR feedback messages they receiver, since this greatly react to FIR feedback messages they receive, since this greatly
improves the user experience when using centralised mixer-based improves the user experience when using centralised mixer-based
conferencing. Support for sending FIR messages 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 message is defined in Section 6.3.1 of The Picture Loss Indication message is defined in Section 6.3.1 of
the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
sending encoder that it lost the decoder context and would like to sending encoder that it lost the decoder context and would like to
have it repaired somehow. This is semantically different from the have it repaired somehow. This is semantically different from the
Full Intra Request above as there could be multiple ways to fulfil Full Intra Request above as there could be multiple ways to fulfil
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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 an endpoint to inform a mixer about the level of extension used by an endpoint to inform a mixer about the level of
audio activity in the packet to which the header is attached. This audio activity in the packet to which the header is attached. This
enables an RTP middlebox to make mixing or selection decisions enables an RTP middlebox to make mixing or selection decisions
without decoding or detailed inspection of the payload, reducing the without decoding or detailed inspection of the payload, reducing the
complexity in some types of mixer. It can also save decoding complexity in some types of mixers. It can also save decoding
resources in receivers, which can choose to decode only the most resources in receivers, which can choose to decode only the most
relevant RTP packet streams based on audio activity levels. relevant RTP packet streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] header extension is The Client-to-Mixer Audio Level [RFC6464] header extension is
RECOMMENDED to be implemented. If this header extension is RECOMMENDED to be implemented. If this header extension is
implemented, it is REQUIRED that implementations are capable of implemented, it is REQUIRED that implementations are capable of
encrypting the header extension according to [RFC6904] since the encrypting the header extension according to [RFC6904] since the
information contained in these header extensions can be considered information contained in these header extensions can be considered
sensitive. It is further RECOMMENDED that this encryption is used, sensitive. The use of this encryption is RECOMMENDED, however usage
unless the encryption has been explicitly disabled through API or of the encryption can be explicitly disabled through API or
signalling. 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
an endpoint with the audio level of the different sources mixed into an endpoint with the audio level of the different sources mixed into
a common mix by a RTP mixer. This enables a user interface to a common source stream by a RTP mixer. This enables a user interface
indicate the relative activity level of each session participant, to indicate the relative activity level of each session participant,
rather than just being included or not based on the CSRC field. This rather than just being included or not based on the CSRC field. This
is a pure optimisations of non critical functions, and is hence is a pure optimisation of non critical functions, and is hence
OPTIONAL to implement. If this header extension is implemented, it OPTIONAL to implement. If this header extension is implemented, it
is REQUIRED that implementations are capable of encrypting the header is REQUIRED that implementations are capable of encrypting the header
extension according to [RFC6904] since the information contained in extension according to [RFC6904] since the information contained in
these header extensions can be considered sensitive. It is further these header extensions can be considered sensitive. It is further
RECOMMENDED that this encryption is used, unless the encryption has RECOMMENDED that this encryption is used, unless the encryption has
been explicitly disabled through API or signalling. 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 packet streams robust against There are tools that can make RTP packet streams robust against
packet loss and reduce the impact of loss on media quality. However, packet loss and reduce the impact of loss on media quality. However,
they all add overhead compared to a non-robust stream. The overhead they generally some add overhead compared to a non-robust stream.
needs to be considered, and the aggregate bit-rate MUST be rate The overhead needs to be considered, and the aggregate bit-rate MUST
controlled to avoid causing network congestion (see Section 7). As a be rate controlled to avoid causing network congestion (see
result, improving robustness might require a lower base encoding Section 7). As a result, improving robustness might require a lower
quality, but has the potential to deliver that quality with fewer base encoding quality, but has the potential to deliver that quality
errors. The mechanisms described in the following sub-sections can with fewer errors. The mechanisms described in the following sub-
be used to improve tolerance to packet loss. sections can 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 send 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. known lost packets.
RTP packet 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 some or all of this feedback (following Section 4.2 of ignore some or all of this feedback (following Section 4.2 of
[RFC4585]). Receivers MAY send NACKs for missing RTP packets. [RFC4585]). Receivers MAY send NACKs for missing RTP packets.
Guidelines on when to send NACKs are provided in [RFC4585]. It is Guidelines on when to send NACKs are provided in [RFC4585]. It is
not expected that a receiver will send a NACK for every lost RTP not expected that a receiver will send a NACK for every lost RTP
packet, rather it needs to consider the cost of sending NACK packet, rather it needs to consider the cost of sending NACK
feedback, and the importance of the lost packet, to make an informed feedback, and the importance of the lost packet, to make an informed
decision on whether it is worth telling the sender about a packet decision on whether it is worth telling the sender about a packet
loss event. 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 caused increased packet loss if RTP bandwidth, and can potentially caused increased packet loss if
the original packet loss was caused by network congestion. We note, the original packet loss was caused by network congestion. Note,
however, that retransmission of an important lost packet to repair however, that retransmission of an important lost packet to repair
decoder state can have lower cost than sending a full intra frame. decoder state can have lower cost than sending a full intra frame.
It is not appropriate to blindly retransmit RTP packets in response It is not appropriate to blindly retransmit RTP packets in response
to a NACK. The importance of lost packets and the likelihood of them to a NACK. The importance of lost packets and the likelihood of them
arriving in time to be useful needs to be considered before RTP arriving in time to be useful needs to be considered before RTP
retransmission is used. retransmission is 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
skipping to change at page 19, line 47 skipping to change at page 20, line 25
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's flows. be used for interactive media applications such as WebRTC's flows.
Some requirements for congestion control algorithms for Some requirements for congestion control algorithms for
RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements].
It is expected that a future version of this memo will mandate the A future version of this memo will mandate the use of a congestion
use of a congestion control algorithm that satisfies these control algorithm that satisfies these requirements.
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 WebRTC implementations MUST implement the RTP circuit breaker
implementations MUST implement the RTP circuit breaker algorithm that algorithm that is described in
is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP circuit breaker is
circuit breaker is designed to enable applications to recognise and designed to enable applications to recognise and react to situations
react to situations of extreme network congestion. However, since of extreme network congestion. However, since the RTP circuit
the RTP circuit breaker might not be triggered until congestion breaker might not be triggered until congestion becomes extreme, it
becomes extreme, it cannot be considered a substitute for congestion cannot be considered a substitute for congestion control, and
control, and applications MUST also implement congestion control to applications MUST also implement congestion control to allow them to
allow them to adapt to changes in network capacity. Any future RTP adapt to changes in network capacity. Any future RTP congestion
congestion control algorithms are expected to operate within the control algorithms are expected to operate within the envelope
envelope allowed by the circuit breaker. allowed by the circuit breaker.
The session establishment signalling will also necessarily establish The session establishment signalling will also necessarily establish
boundaries to which the media bit-rate will conform. The choice of boundaries to which the media bit-rate will conform. The choice of
media codecs provides upper- and lower-bounds on the supported bit- media codecs provides upper- and lower-bounds on the supported bit-
rates that the application can utilise to provide useful quality, and rates that the application can utilise to provide useful quality, and
the packetization choices that exist. In addition, the signalling the packetisation choices that exist. In addition, the signalling
channel can establish maximum media bit-rate boundaries using the SDP channel can establish maximum media bit-rate boundaries using, for
"b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary
Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of
The combination of media codec choice and signalled bandwidth limits this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or
SHOULD be used to limit traffic based on known bandwidth limitations, "b=CT:" lines received from the peer, MUST be followed when sending
for example the capacity of the edge links, to the extent possible. RTP packet streams. A WebRTC endpoint receiving media SHOULD signal
its bandwidth limitations, these limitations have to be based on
7.2. RTCP Limitations for Congestion Control known bandwidth limitations, for example the capacity of the edge
links.
Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent frequently (roughly
once per round trip time is common). We note that the real-time
media traffic might not be able to adapt to changing path conditions
as rapidly as elastic applications using TCP, but frequent feedback,
perhaps on the order of once per video frame, is still needed to
allow the congestion control algorithm to track the path dynamics.
As an example of the type of RTCP congestion control feedback that is
possible, consider one of the simplest scenarios for WebRTC: a point
to point video call between two end systems. There will be four RTP
flows in this scenario, two audio and two video, with all four flows
being active for essentially all the time (the audio flows will
likely use voice activity detection and comfort noise to reduce the
packet rate during silent periods, 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:
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
play out delay due to queuing (most prominent in wireless networks)
can easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues might need to be
reported -- to be defined in a suitable congestion control framework
as noted above -- which, in turn, increase the report size again.
For example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control
algorithms, but the (compound) report size grows quickly.
7.3. Congestion Control Interoperability and Legacy Systems 7.2. Congestion Control Interoperability and Legacy Systems
There are legacy RTP 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
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With proprietary congestion control algorithms issues can arise when With proprietary congestion control algorithms issues can arise when
different algorithms and implementations interact in a communication different algorithms and implementations interact in a communication
session. If the different implementations have made different session. If the different implementations have made different
choices in regards to the type of adaptation, for example one sender choices in regards to the type of adaptation, for example one sender
based, and one receiver based, then one could end up in situation based, and one receiver based, then one could end up in situation
where one direction is dual controlled, when the other direction is where one direction is dual controlled, when the other direction is
not controlled. This memo cannot mandate behaviour for proprietary not controlled. This memo cannot mandate behaviour for proprietary
congestion control algorithms, but implementations that use such congestion control algorithms, but implementations that use such
algorithms ought to be aware of this issue, and try to ensure that algorithms ought to be aware of this issue, and try to ensure that
both effective congestion control is negotiated for media flowing in effective congestion control is negotiated for media flowing in both
both directions. If the IETF were to standardise both sender- and 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 packet streams they send and receive. These RTCP reports can the RTP packet streams they send and receive. These RTCP reports can
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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]. At the time RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time
of this writing, it is not clear what extended metrics are suitable of this writing, it is not clear what extended metrics are suitable
for use in the WebRTC context, so there is no requirement that for use in the WebRTC context, so there is no requirement that
implementations generate RTCP XR packets. However, implementations implementations generate RTCP XR packets. However, implementations
that can use detailed performance monitoring data MAY generate RTCP that can use detailed performance monitoring data MAY generate RTCP
XR packets as appropriate; the use of such packets SHOULD be XR packets as appropriate; the use of such packets SHOULD be
signalled in advance. signalled in advance.
All WebRTC implementations MUST be prepared to receive RTP XR report
packets, whether or not they were signalled. There is no requirement
that the data contained in such reports be used, or exposed to the
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
of WebRTC applications. In this case, future updates to this memo of WebRTC applications. In this case, future updates to this memo
MUST be made following the Guidelines for Writers of RTP Payload MUST be made following the Guidelines for Writers of RTP Payload
Format Specifications [RFC2736], How to Write an RTP Payload Format Format Specifications [RFC2736], How to Write an RTP Payload Format
[I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP
Control Protocol [RFC5968], and SHOULD take into account any future Control Protocol [RFC5968], and SHOULD take into account any future
guidelines for extending RTP and related protocols that have been guidelines for extending RTP and related protocols that have been
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parameters: parameters:
RTP Profile: The name of the RTP profile to be used in session. The RTP Profile: The name of the RTP profile to be used in session. The
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. Interworking functions might transform this into the
this into the RTP/SAVP profile for a legacy use case, by RTP/SAVP profile for a legacy use case, by indicating to the
indicating to the WebRTC end-point that the RTP/SAVPF is used, and WebRTC end-point that the RTP/SAVPF is used and configuring 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 [RFC5245]
signals candidates and arrives at nominated candidate address that 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, i.e. transport-layer flow, is used for RTP and that a single port, i.e. transport-layer flow, is used for RTP and
RTCP flows, this MUST be 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 use of any additional RTP header extensions and
including any parameters for each respective extension. At the RTCP packet types, including any necessary parameters, SHOULD be
very least, this will help avoiding using bandwidth for features signalled. For robustness, and for compatibility with non-WebRTC
that the other end-point will ignore. But for certain mechanisms systems that might be connected to a WebRTC session via a gateway,
there is requirement for this to happen as interoperability implementations are required to ignore unknown RTCP packets and
failure otherwise happens. RTP header extensions (See Section 4.1).
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary. This SHALL be done as described in end-points will be necessary. This SHALL be done as described in
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556], or something Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or
semantically equivalent. This also ensures that the end-points something semantically equivalent. This also ensures that the
have a common view of the RTCP bandwidth, this is important as too end-points have a common view of the RTCP bandwidth. A common
different view of the bandwidths can lead to failure to RTCP bandwidth is important as a too different view of the
interoperate. bandwidths can lead to failure to interoperate.
These parameters are often expressed in SDP messages conveyed within These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the offer an offer/answer exchange. RTP does not depend on SDP or on the offer
/answer model, but does require all the necessary parameters to be /answer model, but does require all the necessary parameters to be
agreed upon, and provided to the RTP implementation. We note that in agreed upon, and provided to the RTP implementation. Note that in
the WebRTC context it will depend on the signalling model and API how the WebRTC context it will depend on the signalling model and API how
these parameters need to be configured but they will be need to these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers. either be set in the API or explicitly signalled between the peers.
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
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the source packet stream. the source packet stream.
It is important to note that the same media source can be feeding It is important to note that the same media source can be feeding
multiple MediaStreamTracks. As different sets of constraints or multiple MediaStreamTracks. As different sets of constraints or
other parameters can be applied to the MediaStreamTrack, each other parameters can be applied to the MediaStreamTrack, each
MediaStreamTrack instance added to a RTCPeerConnection SHALL result MediaStreamTrack instance added to a RTCPeerConnection SHALL result
in an independent source packet stream, with its own set of in an independent source packet stream, with its own set of
associated packet streams, and thus different SSRC(s). It will associated packet streams, and thus different SSRC(s). It will
depend on applied constraints and parameters if the source stream and depend on applied constraints and parameters if the source stream and
the encoding configuration will be identical between different the encoding configuration will be identical between different
MediaStreamTracks sharing the same media source. Thus it is possible MediaStreamTracks sharing the same media source. If the encoding
for multiple source packet streams to share encoded streams (but not parameters and constraints are the same, an implementation could
packet streams), but this is an implementation choice to try to choose to use only one encoded stream to create the different RTP
utilise such optimisations. Note that such optimizations would need packet streams. Note that such optimisations would need to take into
to take into account that the constraints for one of the account that the constraints for one of the MediaStreamTracks can at
MediaStreamTracks can at any moment change, meaning that the encoding any moment change, meaning that the encoding configurations might no
configurations might no longer be identical. longer be identical and two different encoder instances would then be
needed.
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 end-point to change synchronisation all cases, and does not force an end-point to to disrupt the media by
base and CNAME in the middle of a ongoing delivery of any packet changing synchronisation base and CNAME during delivery of any
streams, which would cause media disruption; all MediaStreamTracks ongoing packet streams, all MediaStreamTracks and their associated
and their associated SSRCs originating from the same end-point needs SSRCs originating from the same end-point need to be sent using the
to be sent using the same CNAME within one RTCPeerConnection. This same CNAME within one RTCPeerConnection. This is motivating the
is motivating the strong recommendation in Section 4.9 to only use a strong recommendation in Section 4.9 to only use a single CNAME.
single CNAME.
The requirement on using the same CNAME for all SSRCs that The requirement on using the same CNAME for all SSRCs that
originates from the same end-point, does not require middleboxes originate from the same end-point, does not require a middlebox
that forwards traffic from multiple end-points to only use a that forwards traffic from multiple end-points to only use a
single CNAME. single CNAME.
Different CNAMEs normally need to be used for different Different CNAMEs normally need to be used for different
RTCPeerConnection instances, as specified in Section 4.9. Having two RTCPeerConnection instances, as specified in Section 4.9. Having two
communication sessions with the same CNAME could enable tracking of a communication sessions with the same CNAME could enable tracking of a
user or device across different services (see Section 4.4.1 of user or device across different services (see Section 4.4.1 of
[I-D.ietf-rtcweb-security] for details). A web application can [I-D.ietf-rtcweb-security] for details). A web application can
request that the CNAMEs used in different RTCPeerConnection within a request that the CNAMEs used in different RTCPeerConnections (within
same-orign context to be the same, this allow for synchronization of a same-orign context) be the same, this allows for synchronization of
the endpoint's RTP packet streams across the different the endpoint's RTP packet streams across the different
RTCPeerConnections. RTCPeerConnections.
Note: this doesn't result in a tracking issue, since the creation Note: this doesn't result in a tracking issue, since the creation
of matching CNAMEs depends on existing tracking. of matching CNAMEs depends on existing tracking.
The above will currently force a WebRTC end-point that receives an The above will currently force a WebRTC end-point that receives a
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 to the one it
that it has to use a locally available system clock as timebase for uses, which implies that the sender has to use a local system clock
the synchronisation. Thus, the relative relation between the as timebase for the synchronisation. Thus, the relative relation
timebase of the incoming stream and the system sending out needs to between the timebase of the incoming stream and the system sending
defined. This relation also needs monitoring for clock drift and out needs to defined. This relation also needs monitoring for clock
likely adjustments of the synchronisation. The sending entity is drift and likely adjustments of the synchronisation. The sending
also responsible for congestion control for its the sent streams. In entity is also responsible for congestion control for its sent
cases of packet loss the loss of incoming data also needs to be streams. In cases of packet loss the loss of incoming data also
handled. This leads to the observation that the method that is least needs to be handled. This leads to the observation that the method
likely to cause issues or interruptions in the outgoing source packet that is least likely to cause issues or interruptions in the outgoing
stream is a model of full decoding, including repair etc followed by source packet stream is a model of full decoding, including repair
encoding of the media again into the outgoing packet stream. etc., followed by encoding of the media again into the outgoing
Optimisations of this method is clearly possible and implementation packet stream. Optimisations of this method is clearly possible and
specific. implementation specific.
A WebRTC end-point 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 is to allow for forward compatibility with any future changes that
that enables more efficient stream handling when end-points relay/ enables more efficient stream handling when end-points relay/
forward streams. It also ensures that end-points can interoperate forward streams. It also ensures that end-points can interoperate
with certain types of multi-stream middleboxes or end-points 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. This later is relevant to handle some cases of legacy
will reveal if the packet stream is a source stream or a redundancy interop. Commonly the RTP Payload Type of any incoming packets will
or dependent packet stream. The association to the correct source reveal if the packet stream is a source stream or a redundancy or
dependent packet stream. The association to the correct source
packet stream depends on the payload format in use for the packet packet stream depends on the payload format in use for the packet
stream. stream.
Finally this specification puts a requirement on the WebRTC API to Finally this specification puts a requirement on the WebRTC API to
realize a method for determining the CSRC list (Section 4.1) as well realize a method for determining the CSRC list (Section 4.1) as well
as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) as the Mixer-to-Client audio levels (Section 5.2.3) (when supported)
and the basic requirements for this is further discussed in and the basic requirements for this is further discussed in
Section 12.2.1. Section 12.2.1.
12. RTP Implementation Considerations 12. RTP Implementation Considerations
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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 sources, RTP sessions. Each RTP session can convey multiple media sources,
and can include media data from multiple end-points. In the and can include media data from multiple end-points. In the
following, we outline some ways in which WebRTC end-points can following, some ways in which WebRTC end-points can configure and use
configure and use RTP sessions. RTP sessions is outlined.
12.1.1. Use of Multiple Media Sources Within an RTP Session 12.1.1. Use of Multiple Media Sources Within an RTP Session
RTP is a group communication protocol, and every RTP session can RTP is a group communication protocol, and every RTP session can
potentially contain multiple RTP packet streams. There are several potentially contain multiple RTP packet streams. There are 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 sources (e.g., one RTP session for session for each type of media sources (e.g., one RTP session for
audio sources and one for video sources, each sent over different audio sources and one for video sources, each sent over different
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To separate media with different purposes: An end-point might want To separate media with different purposes: An end-point might want
to send RTP packet streams that have different purposes on to send RTP packet streams that have different purposes on
different RTP sessions, to make it easy for the peer device to different RTP sessions, to make it easy for the peer device to
distinguish them. For example, some centralised multiparty distinguish them. For example, some centralised multiparty
conferencing systems display the active speaker in high conferencing systems display the active speaker in high
resolution, but show low resolution "thumbnails" of other resolution, but show low resolution "thumbnails" of other
participants. Such systems might configure the end-points to send participants. Such systems might configure the end-points to send
simulcast high- and low-resolution versions of their video using simulcast high- and low-resolution versions of their video using
separate RTP sessions, to simplify the operation of the RTP separate RTP sessions, to simplify the operation of the RTP
middlebox. In the WebRTC context this is currently possible to middlebox. In the WebRTC context this is currently possible by
accomplished by establishing multiple WebRTC MediaStreamTracks establishing multiple WebRTC MediaStreamTracks that have the same
that have the same media source in one (or more) media source in one (or more) RTCPeerConnection. Each
RTCPeerConnection. Each MediaStreamTrack is then configured to MediaStreamTrack is then configured to deliver a particular media
deliver a particular media quality and thus media bit-rate, and quality and thus media bit-rate, and will produce an independently
will produce an independently encoded version with the codec encoded version with the codec parameters agreed specifically in
parameters agreed specifically in the context of that the context of that RTCPeerConnection. The RTP middlebox can
RTCPeerConnection. The RTP middlebox can distinguish packets distinguish packets corresponding to the low- and high-resolution
corresponding to the low- and high-resolution streams by streams by inspecting their SSRC, RTP payload type, or some other
inspecting their SSRC, RTP payload type, or some other information information contained in RTP payload, RTP header extension or RTCP
contained in RTP payload, RTP header extension or RTCP packets, packets, but it can be easier to distinguish the RTP packet
but it can be easier to distinguish the RTP packet streams if they streams if they arrive on separate RTP sessions on separate
arrive on separate RTP sessions on separate transport-layer flows. 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 an RTP middlebox. 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 an RTP middlebox 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
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+---+ +---+
| C | | C |
+---+ +---+
Figure 1: Multi-unicast using several RTP sessions Figure 1: Multi-unicast using several RTP sessions
The multi-unicast topology could also be implemented as a single The multi-unicast topology could also be implemented as a single
RTP session, spanning multiple peer-to-peer transport layer RTP session, spanning multiple peer-to-peer transport layer
connections, or as several pairwise RTP sessions, one between each connections, or as several pairwise RTP sessions, one between each
pair of peers. To maintain a coherent mapping between the pair of peers. To maintain a coherent mapping between the
relation between RTP sessions and RTCPeerConnection objects we relation between RTP sessions and RTCPeerConnection objects it is
recommend that this is implemented as several individual RTP recommend that this is implemented as several individual RTP
sessions. The only downside is that end-point A will not learn of sessions. The only downside is that end-point A will not learn of
the quality of any transmission happening between B and C, since the quality of any transmission happening between B and C, since
it will not see RTCP reports for the RTP session between B and C, it will not see RTCP reports for the RTP session between B and C,
whereas it would it all three participants were part of a single whereas it would it all three participants were part of a single
RTP session. Experience with the Mbone tools (experimental RTP- RTP session. Experience with the Mbone tools (experimental RTP-
based multicast conferencing tools from the late 1990s) has showed based multicast conferencing tools from the late 1990s) has showed
that RTCP reception quality reports for third parties can usefully that RTCP reception quality reports for third parties can be
be presented to the users in a way that helps them understand presented to users in a way that helps them understand asymmetric
asymmetric network problems, and the approach of using separate network problems, and the approach of using separate RTP sessions
RTP sessions prevents this. However, an advantage of using prevents this. However, an advantage of using separate RTP
separate RTP sessions is that it enables using different media sessions is that it enables using different media bit-rates and
bit-rates and RTP session configurations between the different RTP session configurations between the different peers, thus not
peers, thus not forcing B to endure the same quality reductions if forcing B to endure the same quality reductions if there are
there are limitations in the transport from A to C as C will. It limitations in the transport from A to C as C will. It is
it believed that these advantages outweigh the limitations in 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 packet 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 packet stream to any given receiver, or by providing a RTP packet stream to any given receiver, or by providing a
skipping to change at page 32, line 23 skipping to change at page 30, line 26
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 RTP middleboxes can use middlebox. A common aspect is that these RTP middleboxes can use
a number of tools to control the media encoding provided by a a number of tools to control the media encoding provided by a
WebRTC end-point. This includes functions like requesting WebRTC end-point. This includes functions like requesting the
breaking the encoding chain and have the encoder produce a so breaking of the encoding chain and have the encoder produce a so
called Intra frame. Another is limiting the bit-rate of a given called Intra frame. Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams. stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between frame-rate and spatial quality. resolution, the trade-off between frame-rate and spatial quality.
The middlebox gets the significant responsibility to correctly The middlebox has the responsibility to correctly perform
perform congestion control, source identification, manage congestion control, source identification, manage synchronisation
synchronisation while providing the application with suitable while providing the application with suitable media optimisations.
media optimizations. The middlebox is also has to be a trusted The middlebox also has to be a trusted node when it comes to
node when it comes to security, since it manipulates either the security, since it manipulates either the RTP header or the media
RTP header or the media itself (or both) received from one end- itself (or both) received from one end-point, before sending it on
point, before sending it on towards the end-point(s), thus they towards the end-point(s), thus they need to be able to decrypt and
need to be able to decrypt and then encrypt it before sending it then re-encrypt the RTP packet stream before sending it out.
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 packet streams across themselves. This is reports regarding RTP packet streams across themselves. This is
due to the difference in the RTP packet 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
skipping to change at page 33, line 16 skipping to change at page 31, line 20
challenge. In the mixer-based topologies, end-points source challenge. In the mixer-based topologies, end-points source
authentication is based on, firstly, verifying that media comes authentication is based on, firstly, verifying that media comes
from the mixer by cryptographic verification and, secondly, trust from the mixer by cryptographic verification and, secondly, trust
in the mixer to correctly identify any source towards the end- in the mixer to correctly identify any source towards the end-
point. In RTP sessions where multiple end-points are directly point. In RTP sessions where multiple end-points are directly
visible to an end-point, all end-points will have knowledge about visible to an end-point, all end-points will have knowledge about
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 packet stream to be able to for an end-point that receives an RTP packet stream to be able to
forward that RTP packet stream to a third party. The are some forward that RTP packet stream to a third party. The are some
obvious security and privacy implications in supporting this, but obvious security and privacy implications in supporting this, but
also potential uses. This is supported in the W3C API by taking also potential uses. This is supported in the W3C API by taking
the received and decoded media and using it as media source that the received and decoded media and using it as media source that
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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 packet 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 optimise 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 RTP packet stream generated by the forwarding an entirely new RTP packet stream generated by the forwarding
device. device.
12.1.3. Differentiated Treatment of RTP Packet Streams 12.1.3. Differentiated Treatment of RTP Packet Streams
There are use cases for differentiated treatment of RTP packet There are use cases for differentiated treatment of RTP packet
skipping to change at page 34, line 35 skipping to change at page 32, line 40
retransmission and FEC. The importance of such redundant RTP packet retransmission and FEC. The importance of such redundant RTP packet
streams is dependent on the media type and codec used, in regards to streams is dependent on the media type and codec used, in regards to
how robust that codec is to packet loss. However, a default policy how robust that codec is to packet loss. However, a default policy
might to be to use the same priority for redundant RTP packet stream might to be to use the same priority for redundant RTP packet stream
as for the source RTP packet stream. as for the source RTP packet stream.
Secondly, the network can prioritize transport-layer flows and sub- Secondly, the network can prioritize transport-layer flows and sub-
flows, including RTP packet streams. Typically, differential flows, including RTP packet streams. Typically, differential
treatment includes two steps, the first being identifying whether an treatment includes two steps, the first being identifying whether an
IP packet belongs to a class that has to be treated differently, the IP packet belongs to a class that has to be treated differently, the
second the actual mechanism to prioritize packets. This is done second consisting of the actual mechanism to prioritize packets.
according to three methods: 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
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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.ietf-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.ietf-tsvwg-rtcweb-qos] for further discussion. As default see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default
policy all RTP packets related to a RTP packet 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 packet stream need to be marked. RTCP compound particular RTP packet stream need to be marked. RTCP compound
packets with Sender Reports (SR), ought to be marked with the same packets with Sender Reports (SR), ought to be marked with the same
priority as the RTP packet stream itself, so the RTCP-based round- priority as the RTP packet stream itself, so the RTCP-based round-
trip time (RTT) measurements are done using the same transport-layer trip time (RTT) measurements are done using the same transport-layer
flow priority as the RTP packet stream experiences. RTCP compound flow priority as the RTP packet stream experiences. RTCP compound
packets containing RR packet ought to be sent with the priority used packets containing RR packet ought to be sent with the priority used
by the majority of the RTP packet streams reported on. RTCP packets by the majority of the RTP packet streams reported on. RTCP packets
containing time-critical feedback packets can use higher priority to containing time-critical feedback packets can use higher priority to
improve the timeliness and likelihood of delivery of such feedback. improve the timeliness and likelihood of delivery of such feedback.
12.2. Media Source, RTP Packet Streams, and Participant Identification 12.2. Media Source, RTP Packet Streams, and Participant Identification
12.2.1. Media Source 12.2.1. Media Source Identification
Each RTP packet 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 each of source (SSRC) identifier. The SSRC identifier is carried in each of
the RTP packets comprising a RTP packet stream, and is also used to the RTP packets comprising a RTP packet stream, and is also used to
identify that stream in the corresponding RTCP reports. The SSRC is identify that stream in the corresponding RTCP reports. The SSRC is
chosen as discussed in Section 4.8. The first stage in chosen as discussed in Section 4.8. The first stage in
demultiplexing RTP and RTCP packets received on a single transport demultiplexing RTP and RTCP packets received on a single transport
layer flow at a WebRTC end-point is to separate the RTP packet 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.
skipping to change at page 36, line 38 skipping to change at page 34, line 42
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. It is desirable to expose this information to the WebRTC application
using some API, after mapping the CSRC values to WebRTC MediaStream
identities, so it can be exposed in the user interface.
12.2.2. SSRC Collision Detection 12.2.2. SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have The RTP standard requires RTP implementations to have support for
support for detecting and handling SSRC collisions, i.e., resolve the detecting and handling SSRC collisions, i.e., resolve the conflict
conflict when two different end-points use the same SSRC value. This when two different end-points use the same SSRC value (see section
requirement also applies to WebRTC end-points. There are several 8.2 of [RFC3550]). This requirement also applies to WebRTC end-
scenarios where SSRC collisions can occur: points. There are several scenarios where SSRC collisions can occur:
o In a point-to-point session where each SSRC is associated with o In a point-to-point session where each SSRC is associated with
either of the two end-points and where the main media carrying either of the two end-points and where the main media carrying
SSRC identifier will be announced in the signalling channel, a SSRC identifier will be announced in the signalling channel, a
collision is less likely to occur due to the information about collision is less likely to occur due to the information about
used SSRCs provided by Source-Specific SDP Attributes [RFC5576]. used SSRCs. If SDP is used, this information is provided by
Source-Specific SDP Attributes [RFC5576]. Still, collisions can
Still, collisions can occur if both end-points start uses an new occur if both end-points start using a new SSRC identifier prior
SSRC identifier prior to having signalled it to the peer and to having signalled it to the peer and received acknowledgement on
received acknowledgement on the signalling message. The Source- the signalling message. The Source-Specific SDP Attributes
Specific SDP Attributes [RFC5576] contains no mechanism to resolve [RFC5576] contains a mechanism to signal how the end-point
SSRC collisions or reject a end-points usage of an SSRC. resolved the SSRC collision.
o SSRC values that have not been signalled could also appear in an 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 RTP session. This is more likely than it appears, since some RTP
functions use extra SSRCs to provide their functionality. For functions use extra SSRCs to provide their functionality. For
example, retransmission data might be transmitted using a separate example, retransmission data might be transmitted using a separate
RTP packet stream that requires its own SSRC, separate to the SSRC RTP packet stream that requires its own SSRC, separate to the SSRC
of the source RTP packet stream [RFC4588]. In those cases, an 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 end-point can create a new SSRC that strictly doesn't need to be
announced over the signalling channel to function correctly on announced over the signalling channel to function correctly on
both RTP and RTCPeerConnection level. both RTP and RTCPeerConnection level.
o Multiple end-points in a multiparty conference can create new o Multiple end-points in a multiparty conference can create new
sources and signal those towards the RTP middlebox. In cases sources and signal those towards the RTP middlebox. In cases
where the SSRC/CSRC are propagated between the different end- where the SSRC/CSRC are propagated between the different end-
points from the RTP middlebox collisions can occur. points from the RTP middlebox collisions can occur.
o An RTP middlebox could connect an end-point's RTCPeerConnection to o An RTP middlebox could connect an end-point's RTCPeerConnection to
another RTCPeerConnection from the same end-point, thus forming a another RTCPeerConnection from the same end-point, thus forming a
loop where the end-point will receive its own traffic. While is loop where the end-point will receive its own traffic. While it
is clearly considered a bug, it is important that the end-point is is clearly considered a bug, it is important that the end-point is
able to recognise and handle the case when it occurs. This case able to recognise and handle the case when it occurs. This case
becomes even more problematic when media mixers, and so on, are becomes even more problematic when media mixers, and so on, are
involved, where the stream received is a different stream but involved, where the stream received is a different 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,
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 packet streams be indicated as coming from the same set of RTP packet streams be indicated as coming from the same
skipping to change at page 38, line 25 skipping to change at page 36, line 34
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 also apply to this memo. 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 It is not believed there are any new security considerations
from the combination of these various protocol extensions. resulting 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
skipping to change at page 39, line 19 skipping to change at page 37, line 28
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, Christian Groves, Cullen Jennings, Dan Cary Bran, Ben Campbell, Charles Eckel, Alex Eleftheriadis, Christian
Romascanu, Martin Thomson, and the other members of the IETF RTCWEB Groves, Cullen Jennings, Olle Johansson, Suhas Nandakumar, Dan
working group for their valuable feedback. Romascanu, Jim Spring, Martin Thomson, and the other members of the
IETF RTCWEB 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-05 (work in ietf-avtcore-multi-media-rtp-session-05 (work in
progress), February 2014. progress), February 2014.
skipping to change at page 42, line 15 skipping to change at page 40, line 24
[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 [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014. Clock Rates in an RTP Session", RFC 7160, April 2014.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC
7164, March 2014. 7164, March 2014.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-
webrtc-20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
16.2. Informative References 16.2. Informative References
[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-
skipping to change at page 43, line 46 skipping to change at page 41, line 46
[I-D.ietf-tsvwg-rtcweb-qos] [I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-tsvwg- other packet markings for RTCWeb QoS", draft-ietf-tsvwg-
rtcweb-qos-00 (work in progress), April 2014. 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
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383, February Real-time Transport Protocol (SRTP)", RFC 4383, February
2006. 2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(TFRC): The Small-Packet (SP) Variant", RFC 4828, April (ICE): A Protocol for Network Address Translator (NAT)
2007. Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, June 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010. Control Protocol (RTCP)", RFC 5968, September 2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011. Control Protocol (RTCP) Flows", RFC 6263, June 2011.
[RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the
RTP Monitoring Framework", RFC 6792, November 2012. RTP Monitoring Framework", RFC 6792, November 2012.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-
webrtc-20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
URI: http://csperkins.org/ URI: http://csperkins.org/
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