draft-ietf-rtcweb-rtp-usage-07.txt   draft-ietf-rtcweb-rtp-usage-08.txt 
RTCWEB Working Group C. S. Perkins RTCWEB Working Group C. S. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund Intended status: Standards Track M. Westerlund
Expires: January 16, 2014 Ericsson Expires: March 05, 2014 Ericsson
J. Ott J. Ott
Aalto University Aalto University
July 15, 2013 September 01, 2013
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-07 draft-ietf-rtcweb-rtp-usage-08
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.
skipping to change at page 1, line 39 skipping to change at page 1, line 39
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 16, 2014. This Internet-Draft will expire on March 05, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the Copyright (c) 2013 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
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 6 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 6
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 8
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 11
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 13 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 13
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 14 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 14
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15
5.2.4. Associating RTP Media Streams and Signalling Contexts 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16
6.1. Negative Acknowledgements and RTP Retransmission . . . . 16 6.1. Negative Acknowledgements and RTP Retransmission . . . . 16
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 17 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 17
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 17 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 17
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 18 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 18
7.2. RTCP Limitations for Congestion Control . . . . . . . . . 19 7.2. RTCP Limitations for Congestion Control . . . . . . . . . 19
7.3. Congestion Control Interoperability and Legacy Systems . 19 7.3. Congestion Control Interoperability and Legacy Systems . 19
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 21 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 21
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23
12. RTP Implementation Considerations . . . . . . . . . . . . . . 23 12. RTP Implementation Considerations . . . . . . . . . . . . . . 23
12.1. RTP Sessions and PeerConnections . . . . . . . . . . . . 24 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 24
12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . 25 12.1.1. Use of Multiple Media Flows Within an RTP Session . 24
12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . 25 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 25
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . 27 12.1.3. Differentiated Treatment of Flows . . . . . . . . . 30
12.5. Contributing Sources and the CSRC List . . . . . . . . . 28 12.2. Source, Flow, and Participant Identification . . . . . . 31
12.6. Media Synchronization . . . . . . . . . . . . . . . . . 28 12.2.1. Media Streams . . . . . . . . . . . . . . . . . . . 31
12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . 29 12.2.2. Media Streams: SSRC Collision Detection . . . . . . 32
12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . 30 12.2.3. Media Synchronisation Context . . . . . . . . . . . 33
12.9. Differentiated Treatment of Flows . . . . . . . . . . . 30 12.2.4. Correlation of Media Streams . . . . . . . . . . . . 34
13. Security Considerations . . . . . . . . . . . . . . . . . . . 32 13. Security Considerations . . . . . . . . . . . . . . . . . . . 34
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 34
15. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 33 15. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 35
16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33 16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 35
17. References . . . . . . . . . . . . . . . . . . . . . . . . . 33 17. References . . . . . . . . . . . . . . . . . . . . . . . . . 35
17.1. Normative References . . . . . . . . . . . . . . . . . . 34 17.1. Normative References . . . . . . . . . . . . . . . . . . 35
17.2. Informative References . . . . . . . . . . . . . . . . . 37 17.2. Informative References . . . . . . . . . . . . . . . . . 38
Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 38 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 40
A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 39
A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . 41
A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 44
A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . 45
A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 47
A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . 50
A.4. Translator Based . . . . . . . . . . . . . . . . . . . . 52
A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . 52
A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53
A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55
A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . 58
A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 60
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.
The Web Real-Time communication (WebRTC) framework provides the The Web Real-Time communication (WebRTC) framework provides the
protocol building blocks to support direct, interactive, real-time protocol building blocks to support direct, interactive, real-time
communication using audio, video, collaboration, games, etc., between communication using audio, video, collaboration, games, etc., between
two peers' web-browsers. This memo describes how the RTP framework two peers' web-browsers. This memo describes how the RTP framework
is to be used in the WebRTC context. It proposes a baseline set of is to be used in the WebRTC context. It proposes a baseline set of
RTP features that are to be implemented by all WebRTC-aware end- RTP features that are to be implemented by all WebRTC-aware end-
points, along with suggested extensions for enhanced functionality. points, along with suggested extensions for enhanced functionality.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete This memo specifies a protocol intended for use within the WebRTC
WebRTC framework, of which this memo is a part. framework, but is not restricted to that context. An overview of the
WebRTC framework is given in [I-D.ietf-rtcweb-overview].
The structure of this memo is as follows. Section 2 outlines our The structure of this memo is as follows. Section 2 outlines our
rationale in preparing this memo and choosing these RTP features. rationale in preparing this memo and choosing these RTP features.
Section 3 defines terminology. Requirements for core RTP protocols Section 3 defines terminology. Requirements for core RTP protocols
are described in Section 4 and suggested RTP extensions are described are described in Section 4 and suggested RTP extensions are described
in Section 5. Section 6 outlines mechanisms that can increase in Section 5. Section 6 outlines mechanisms that can increase
robustness to network problems, while Section 7 describes congestion robustness to network problems, while Section 7 describes congestion
control and rate adaptation mechanisms. The discussion of mandated control and rate adaptation mechanisms. The discussion of mandated
RTP mechanisms concludes in Section 8 with a review of performance RTP mechanisms concludes in Section 8 with a review of performance
monitoring and network management tools that can be used in the monitoring and network management tools that can be used in the
skipping to change at page 5, line 43 skipping to change at page 5, line 26
directly in RTP and RTCP packets, or as a contributing source directly in RTP and RTCP packets, or as a contributing source
(CSRC) in RTP packets from a mixer. The RTP Session scope is (CSRC) in RTP packets from a mixer. The RTP Session scope is
hence decided by the endpoints' network interconnection topology, hence decided by the endpoints' network interconnection topology,
in combination with RTP and RTCP forwarding strategies deployed by in combination with RTP and RTCP forwarding strategies deployed by
endpoints and any interconnecting middle nodes. endpoints and any interconnecting middle nodes.
WebRTC MediaStream: The MediaStream concept defined by the W3C in WebRTC MediaStream: The MediaStream concept defined by the W3C in
the API. the API.
Other terms are used according to their definitions from the RTP Other terms are used according to their definitions from the RTP
Specification [RFC3550] and WebRTC overview Specification [RFC3550].
[I-D.ietf-rtcweb-overview] documents.
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 and that need to be implemented, along with the mandated RTP profiles and
payload formats. Also described are the core extensions providing payload formats. Also described are the core extensions providing
essential features that all WebRTC implementations need to implement essential features that all WebRTC implementations need to implement
to function effectively on today's networks. to function effectively on today's networks.
4.1. RTP and RTCP 4.1. RTP and RTCP
skipping to change at page 6, line 24 skipping to change at page 6, line 12
functionality implementations of RTP, but are REQUIRED in all WebRTC functionality implementations of RTP, but are REQUIRED in all WebRTC
implementations: implementations:
o Support for use of multiple simultaneous SSRC values in a single o Support for use of multiple simultaneous SSRC values in a single
RTP session, including support for RTP end-points that send many RTP session, including support for RTP end-points that send many
SSRC values simultaneously, following [RFC3550] and SSRC values simultaneously, following [RFC3550] and
[I-D.ietf-avtcore-rtp-multi-stream]. Support for the RTCP [I-D.ietf-avtcore-rtp-multi-stream]. Support for the RTCP
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.
* (tbd: is draft-westerlund-mmusic-max-ssrc-01 needed?) * (tbd: do endpoints need to signal the maximum number of SSRCs
that they support (e.g., draft-westerlund-mmusic-max-ssrc-01)
and/or some constraint on the maximum number of simultaneous
streams of various kinds that can be decoded?)
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 Support for sending correct synchronization information in the o Support for sending correct synchronization information in the
RTCP Sender Reports, to allow a receiver to implement lip-sync, RTCP Sender Reports, to allow a receiver to implement lip-sync,
with RECOMMENDED support for the rapid RTP synchronisation with RECOMMENDED support for the rapid RTP synchronisation
skipping to change at page 8, line 7 skipping to change at page 7, line 47
Implementations MUST support DTLS-SRTP [RFC5764] for key-management. Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
Other key management schemes MAY be supported. Other key management schemes MAY be supported.
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
The set of mandatory to implement codecs and RTP payload formats for The set of mandatory to implement codecs and RTP payload formats for
WebRTC is not specified in this memo. Implementations can support WebRTC is not specified in this memo. Implementations can support
any codec for which an RTP payload format and associated signalling any codec for which an RTP payload format and associated signalling
is defined. Implementation cannot assume that the other participants is defined. Implementation cannot assume that the other participants
in an RTP session understand any RTP payload format, no matter how in an RTP session understand any RTP payload format, no matter how
common; support for all RTP payload formats MUST be negotiated before common; the mapping between RTP payload type numbers and specific
they are used. configurations of particular RTP payload formats MUST be agreed
before those payload types/formats can be used. In an SDP context,
this can be done using the "a=rtpmap:" and "a=fmtp:" attributes
associated with an "m=" line.
Endpoints can signal support for multiple RTP payload formats, or Endpoints can signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, as long as multiple configurations of a single RTP payload format, as long as
each unique RTP payload format configuration uses a different RTP each unique RTP payload format configuration uses a different RTP
payload type number. As outlined in Section 4.8, the RTP payload payload type number. As outlined in Section 4.8, the RTP payload
type number is sometimes used to associate an RTP media stream with a type number is sometimes used to associate an RTP media stream with a
signalling context. This association is possible provided unique RTP 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
media stream can be associated with an SDP "m=" line by comparing the media stream can be associated with an SDP "m=" line by comparing the
RTP payload type numbers used by the media stream with payload types RTP payload type numbers used by the media stream with payload types
skipping to change at page 9, line 24 skipping to change at page 9, line 16
this way, separating each session using different transport-layer this way, separating each session using different transport-layer
addresses (e.g., different UDP ports) for compatibility with legacy addresses (e.g., different UDP ports) for compatibility with legacy
systems. systems.
In modern day networks, however, with the widespread use of network In modern day networks, however, with the widespread use of network
address/port translators (NAT/NAPT) and firewalls, it is desirable to address/port translators (NAT/NAPT) and firewalls, it is desirable to
reduce the number of transport-layer flows used by RTP applications. reduce the number of transport-layer flows used by RTP applications.
This can be done by sending all the RTP media streams in a single RTP This can be done by sending all the RTP media streams in a single RTP
session, which will comprise a single transport-layer flow (this will session, which will comprise a single transport-layer flow (this will
prevent the use of some quality-of-service mechanisms, as discussed prevent the use of some quality-of-service mechanisms, as discussed
in Section 12.9). Implementations are REQUIRED to support transport in Section 12.1.3). Implementations are REQUIRED to support
of all RTP media streams, independent of media type, in a single RTP transport of all RTP media streams, independent of media type, in a
session according to [I-D.ietf-avtcore-multi-media-rtp-session]. If single RTP session according to
such RTP session set-up is to be used, this MUST be negotiated during [I-D.ietf-avtcore-multi-media-rtp-session]. If multiple types of
the signalling phase [I-D.ietf-mmusic-sdp-bundle-negotiation]. media are to be used in a single RTP session, all participants in
that session MUST agree to this usage. In an SDP context,
[I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal this.
It is also possible to use a shim-based approach to run multiple RTP It is also possible to use a shim-based approach to run multiple RTP
sessions on a single transport-layer flow. This gives advantages in sessions on a single transport-layer flow. This gives advantages in
some gateway scenarios, and makes it easy to distinguish groups of some gateway scenarios, and makes it easy to distinguish groups of
RTP media streams that might need distinct processing. One way of RTP media streams that might need distinct processing. One way of
doing this is described in doing this is described in
[I-D.westerlund-avtcore-transport-multiplexing]. At the time of this [I-D.westerlund-avtcore-transport-multiplexing]. At the time of this
writing, there is no consensus to use a shim-based approach in WebRTC writing, there is no consensus to use a shim-based approach in WebRTC
implementations. implementations.
skipping to change at page 12, line 23 skipping to change at page 12, line 18
5.1. Conferencing Extensions 5.1. Conferencing Extensions
RTP is inherently a group communication protocol. Groups can be RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or using IP implemented using a centralised server, multi-unicast, or using IP
multicast. While IP multicast is popular in IPTV systems, overlay- multicast. While IP multicast is popular in IPTV systems, overlay-
based topologies dominate in interactive conferencing environments. based topologies dominate in interactive conferencing environments.
Such overlay-based topologies typically use one or more central Such overlay-based topologies typically use one or more central
servers to connect end-points in a star or flat tree topology. These servers to connect end-points in a star or flat tree topology. These
central servers can be implemented in a number of ways as discussed central servers can be implemented in a number of ways as discussed
in Appendix A, and in the memo on RTP Topologies in the memo on RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update]. [I-D.ietf-avtcore-rtp-topologies-update].
Not all of the possible the overlay-based topologies are suitable for Not all of the possible the overlay-based topologies are suitable for
use in the WebRTC environment. Specifically: use in the WebRTC environment. Specifically:
o The use of video switching MCUs makes the use of RTCP for o The use of video switching MCUs makes the use of RTCP for
congestion control and quality of service reports problematic (see congestion control and quality of service reports problematic (see
Section 3.7 of [I-D.westerlund-avtcore-rtp-topologies-update]). Section 3.6.2 of [I-D.ietf-avtcore-rtp-topologies-update]).
o The use of content modifying MCUs with RTCP termination breaks RTP o The use of content modifying MCUs with RTCP termination breaks RTP
loop detection, and prevents receivers from identifying active loop detection, and prevents receivers from identifying active
senders (see section 3.8 of senders (see section 3.8 of
[I-D.westerlund-avtcore-rtp-topologies-update]). [I-D.ietf-avtcore-rtp-topologies-update]).
o RTP Transport Translators (Topo-Translator) are not of immediate
interest to WebRTC, although the main difference compared to point
to point is the possibility of seeing multiple different transport
paths in any RTCP feedback.
Accordingly, only Point to Point (Topo-Point-to-Point), Multiple Accordingly, only Point to Point (Topo-Point-to-Point), Multiple
concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer) concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer)
topologies are needed to achieve the use-cases to be supported in topologies are needed to achieve the use-cases to be supported in
WebRTC initially. These RECOMMENDED topologies are expected to be WebRTC initially. These RECOMMENDED topologies are expected to be
supported by all WebRTC end-points (these topologies require no supported by all WebRTC end-points (these topologies require no
special RTP-layer support in the end-point if the RTP features special RTP-layer support in the end-point if the RTP features
mandated in this memo are implemented). mandated in this memo are implemented).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
skipping to change at page 16, line 5 skipping to change at page 15, line 42
the client with the audio level of the different sources mixed into a the client with the audio level of the different sources mixed into a
common mix by a RTP mixer. This enables a user interface to indicate common mix by a RTP mixer. This enables a user interface to indicate
the relative activity level of each session participant, rather than the relative activity level of each session participant, rather than
just being included or not based on the CSRC field. This is a pure just being included or not based on the CSRC field. This is a pure
optimisations of non critical functions, and is hence OPTIONAL to optimisations of non critical functions, and is hence OPTIONAL to
implement. If it is implemented, it is REQUIRED that the header implement. If it is implemented, it is REQUIRED that the header
extensions are encrypted according to extensions are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions can be considered sensitive. contained in these header extensions can be considered sensitive.
5.2.4. Associating RTP Media Streams and Signalling Contexts
(tbd: it seems likely that we need a mechanism to associate RTP media
streams with signalling contexts. The mechanism by which this is
done will likely be some combination of an RTP header extension,
periodic transmission of a new RTCP SDES item, and some signalling
extension. The semantics of those items are not yet settled; see
draft-westerlund-avtext-rtcp-sdes-srcname, draft-ietf-mmusic-msid,
and draft-even-mmusic-application-token for discussion).
6. WebRTC Use of RTP: Improving Transport Robustness 6. WebRTC Use of RTP: Improving Transport Robustness
There are tools that can make RTP media streams robust against packet There are tools that can make RTP media streams robust against packet
loss and reduce the impact of loss on media quality. However, they loss and reduce the impact of loss on media quality. However, they
all add extra bits compared to a non-robust stream. The overhead of all add extra bits compared to a non-robust stream. The overhead of
these extra bits needs to be considered, and the aggregate bit-rate these extra bits needs to be considered, and the aggregate bit-rate
MUST be rate controlled to avoid causing network congestion (see MUST be rate controlled to avoid causing network congestion (see
Section 7). As a result, improving robustness might require a lower Section 7). As a result, improving robustness might require a lower
base encoding quality, but has the potential to deliver that quality base encoding quality, but has the potential to deliver that quality
with fewer errors. The mechanisms described in the following sub- with fewer errors. The mechanisms described in the following sub-
skipping to change at page 24, line 5 skipping to change at page 24, line 5
(tbd: It is an open question whether these considerations are best (tbd: It is an open question whether these considerations are best
discussed in this draft, in the W3C WebRTC API spec, or elsewhere. discussed in this draft, in the W3C WebRTC API spec, or elsewhere.
12. RTP Implementation Considerations 12. RTP Implementation Considerations
The following discussion provides some guidance on the implementation The following discussion provides some guidance on the implementation
of the RTP features described in this memo. The focus is on a WebRTC of the RTP features described in this memo. The focus is on a WebRTC
end-point implementation perspective, and while some mention is made end-point implementation perspective, and while some mention is made
of the behaviour of middleboxes, that is not the focus of this memo. of the behaviour of middleboxes, that is not the focus of this memo.
12.1. RTP Sessions and PeerConnections 12.1. Configuration and Use of RTP Sessions
An RTP session is an association among RTP nodes, which have a single
shared SSRC space. An RTP session can include a large number of end-
points and nodes, each sourcing, sinking, manipulating, or reporting
on the RTP media streams being sent within the RTP session.
A PeerConnection is a point-to-point association between an end-point
and some other peer node. That peer node can be either an end-point
or a centralized processing node of some type. Hence, an RTP session
can terminate immediately at the far end of a PeerConnection, or it
might continue as further discussed below for multiparty sessions
(Section 12.3) and sessions with multiple end points (Section 12.7).
A PeerConnection can contain one or more RTP sessions, depending on
how it is set up, and how many UDP flows it uses. A common usage has
been to have one RTP session per media type, e.g. one for audio and
one for video, each sent over a different UDP flow. However, the
default usage in WebRTC will be to use one RTP session for all media
types, with RTP and RTCP multiplexing (Section 4.5) also mandated.
This RTP session then uses only one UDP flow. However, for legacy
interworking and flow-based network prioritization (Section 12.9), a
WebRTC end-point needs to support a mode of operation where one RTP
session per media type is used. Currently, each RTP session has to
use its own UDP flow in this case, however it might be possible to
multiplex several RTP sessions over a single UDP flow, see
Section 4.4.
The multi-unicast- or mesh-based multi-party topology (Figure 1) is a
good example for this section as it concerns the relation between RTP
sessions and PeerConnections. In this topology, each participant
sends individual unicast RTP/UDP/IP flows to each of the other
participants using independent PeerConnections in a full mesh. This
topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by
requiring one copy of the RTP media streams for each participant that
are part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the
media is very low bandwidth.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast
The multi-unicast topology could be implemented as a single RTP
session, spanning multiple peer-to-peer transport layer connections,
or as several pairwise RTP sessions, one between each pair of peers.
To maintain a coherent mapping between the relation between RTP
sessions and PeerConnections we recommend that one implements this as
individual RTP sessions. The only downside is that end-point A will
not learn of the quality of any transmission happening between B and
C based on RTCP. This has not been seen as a significant downside as
no one has yet seen a clear need for why A would need to know about
the B's and C's communication. An advantage of using separate RTP
sessions is that it enables using different media bit-rates to the
different peers, thus not forcing B to endure the same quality
reductions if there are limitations in the transport from A to C as C
will.
12.2. Multiple Sources
A WebRTC end-point might have multiple cameras, microphones or audio
inputs and thus a single end-point can source multiple RTP media
streams of the same media type concurrently. Even if an end-point
does not have multiple media sources of the same media type it has to
support transmission using multiple SSRCs concurrently in the same
RTP session. This is due to the requirement on an WebRTC end-point
to support multiple media types in one RTP session. For example, one
audio and one video source can result in the end-point sending with
two different SSRCs in the same RTP session. As multi-party
conferences are supported, as discussed below in Section 12.3, a
WebRTC end-point will need to be capable of receiving, decoding and
play out multiple RTP media streams of the same type concurrently.
tbd: there needs to be a way of indicating how RTP stream relate when
there are multiple sources, possibly with simulcast or layered
coding, and different types of mixer or other middlebox. It is
possible that the various BUNDLE/Plan-X proposals will solve this,
but it might also need RTP-level stream identification. To be
resolved once the outcome of the BUNDLE and plan-X discussions is
known.
tbd: Are any mechanism needed to signal limitations in the number of
active SSRC that an end-point can handle?
12.3. Multiparty
There are numerous situations and clear use cases for WebRTC
supporting RTP sessions supporting multi-party. This can be realized
in a number of ways using a number of different implementation
strategies. In the following, the focus is on the different set of
WebRTC end-point requirements that arise from different sets of
multi-party topologies.
The multi-unicast mesh (Figure 1)-based multi-party topology
discussed above provides a non-centralized solution but can incur a
heavy tax on the end-points' outgoing paths. It can also consume
large amount of encoding resources if each outgoing stream is
specifically encoded. If an encoding is transmitted to multiple
parties, as in some implementations of the mesh case, a requirement
on the end-point becomes to be able to create RTP media streams
suitable for multiple destinations requirements. These requirements
can both be dependent on transport path and the different end-points
preferences related to play out of the media.
+---+ +------------+ +---+ A WebRTC end-point will be a simultaneous participant in one or more
| A |<---->| |<---->| B | RTP sessions. Each RTP session can convey multiple media flows, and
+---+ | | +---+ can include media data from multiple end-points. In the following,
| Mixer | we outline some ways in which WebRTC end-points can configure and use
+---+ | | +---+ RTP sessions.
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 2: RTP Mixer with Only Unicast Paths 12.1.1. Use of Multiple Media Flows Within an RTP Session
A Mixer (Figure 2) is an RTP end-point that optimizes the RTP is a group communication protocol, and in a WebRTC context every
transmission of RTP media streams from certain perspectives, either RTP session can potentially contain multiple media flows. There are
by only sending some of the received RTP media stream to any given several reasons why this might be desirable:
receiver or by providing a combined RTP media stream out of a set of
contributing streams. There are various methods of implementation as
discussed in Appendix A.3. A common aspect is that these central
nodes can use a number of tools to control the media encoding
provided by a WebRTC end-point. This includes functions like
requesting breaking the encoding chain and have the encoder produce a
so called Intra frame. Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between frame-rate and spatial quality.
A mixer gets a significant responsibility to correctly perform Multiple media types: Outside of WebRTC, it is common to use one RTP
congestion control, source identification, manage synchronization session for each type of media (e.g., one RTP session for audio
while providing the application with suitable media optimizations. and one for video, each sent on a different UDP port). However,
to reduce the number of UDP ports used, the default in WebRTC is
to send all types of media in a single RTP session, as described
in Section 4.4, using RTP and RTCP multiplexing (Section 4.5) to
further reduce the number of UDP ports needed. This RTP session
then uses only one UDP flow, but will contain multiple RTP media
streams, each containing a different type of media. A common
example might be an end-point with a camera and microphone that
sends two RTP streams, one video and one audio, into a single RTP
session.
Mixers also need to be trusted nodes when it comes to security as it Multiple Capture Devices: A WebRTC end-point might have multiple
manipulates either RTP or the media itself before sending it on cameras, microphones, or other media capture devices, and so might
towards the end-point(s), thus they need to be able to decrypt and want to generate several RTP media streams of the same media type.
then encrypt it before sending it out. Alternatively, it might want to send media from a single capture
device in several different formats or quality settings at once.
Both can result in a single end-point sending multiple RTP media
streams of the same media type into a single RTP session at the
same time.
12.4. SSRC Collision Detection Associated Repair Data: An end-point might send a media stream that
is somehow associated with another stream. For example, it might
send an RTP stream that contains FEC or retransmission data
relating to another stream. Some RTP payload formats send this
sort of associated repair data as part of the original media
stream, while others send it as a separate stream.
The RTP standard [RFC3550] requires any RTP implementation to have Layered or Multiple Description Coding: An end-point can use a
support for detecting and handling SSRC collisions, i.e., resolve the layered media codec, for example H.264 SVC, or a multiple
conflict when two different end-points use the same SSRC value. This description codec, that generates multiple media flows, each with
requirement also applies to WebRTC end-points. There are several a distinct RTP SSRC, within a single RTP session.
scenarios where SSRC collisions can occur.
In a point-to-point session where each SSRC is associated with either RTP Mixers, Translators, and Other Middleboxes: An RTP session, in
of the two end-points and where the main media carrying SSRC the WebRTC context, is a point-to-point association between an
identifier will be announced in the signalling channel, a collision end-point and some other peer device, where those devices share a
is less likely to occur due to the information about used SSRCs common SSRC space. The peer device might be another WebRTC end-
provided by Source-Specific SDP Attributes [RFC5576]. Still if both point, or it might be an RTP mixer, translator, or some other form
end-points start uses an new SSRC identifier prior to having of media processing middlebox. In the latter cases, the middlebox
signalled it to the peer and received acknowledgement on the might send mixed or relayed RTP streams from several participants,
signalling message, there can be collisions. The Source-Specific SDP that the WebRTC end-point will need to render. Thus, even though
Attributes [RFC5576] contains no mechanism to resolve SSRC collisions a WebRTC end-point might only be a member of a single RTP session,
or reject a end-points usage of an SSRC. the peer device might be extending that RTP session to incorporate
other end-points. WebRTC is a group communication environment and
end-points need to be capable of receiving, decoding, and playing
out multiple RTP media streams at once, even in a single RTP
session.
There could also appear SSRC values that are not signalled. This is (tbd: Are any mechanism needed to signal limitations in the number
more likely than it appears as certain RTP functions need extra SSRCs of active SSRC that an end-point can handle?)
to provide functionality related to another (the "main") SSRC, for
example, SSRC multiplexed RTP retransmission [RFC4588]. In those
cases, an end-point can create a new SSRC that strictly doesn't need
to be announced over the signalling channel to function correctly on
both RTP and PeerConnection level.
The more likely case for SSRC collision is that multiple end-points (tbd: need to discuss signalling for the above here, preferably by
in a multiparty conference create new sources and signals those referring to a separate document that describes SDP use for WebRTC)
towards the central server. In cases where the SSRC/CSRC are
propagated between the different end-points from the central node
collisions can occur.
Another scenario is when the central node manages to connect an end- 12.1.2. Use of Multiple RTP Sessions
point's PeerConnection to another PeerConnection the end-point
already has, thus forming a loop where the end-point will receive its
own traffic. While 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 becomes even more problematic when media mixers,
and so on, are involved, where the stream received is a different
stream but still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long In addition to sending and receiving multiple media streams within a
as the same RTP session is extended across multiple PeerConnections single RTP session, a WebRTC end-point might participate in multiple
by a RTP middlebox. To resolve the more generic case where multiple RTP sessions. There are several reasons why a WebRTC end-point might
PeerConnections are interconnected, then identification of the media choose to do this:
source(s) part of a MediaStreamTrack being propagated across multiple
interconnected PeerConnection needs to be preserved across these
interconnections.
12.5. Contributing Sources and the CSRC List To interoperate with legacy devices: The common practice in the non-
WebRTC world is to send different types of media in separate RTP
sessions, for example using one RTP session for audio and another
RTP session, on a different UDP port, for video. All WebRTC end-
points need to support the option of sending different types of
media on different RTP sessions, so they can interwork with such
legacy devices. This is discussed further in Section 4.4.
RTP allows a mixer, or other RTP-layer middlebox, to combine media To provide enhanced quality of service: Some network-based quality
flows from multiple sources to form a new media flow. The RTP data of service mechanisms operate on the granularity of UDP 5-tuples.
packets in that new flow will include a Contributing Source (CSRC) If it is desired to use these mechanisms to provide differentiated
list, indicating which original SSRCs contributed to the combined quality of service for some RTP flows, then those RTP flows need
packet. As described in Section 4.1, implementations need to support to be sent in a separate RTP session using a different UDP port
reception of RTP data packets containing a CSRC list and RTCP packets number, and with appropriate quality of service marking. This is
that relate to sources present in the CSRC list. discussed further in Section 12.1.3.
The CSRC list can change on a packet-by-packet basis, depending on To separate media with different purposes: An end-point might want
the mixing operation being performed. Knowledge of what sources to send media streams that have different purposes on different
contributed to a particular RTP packet can be important if the user RTP sessions, to make it easy for the peer device to distinguish
interface indicates which participants are active in the session. them. For example, some centralised multiparty conferencing
Changes in the CSRC list included in packets needs to be exposed to systems display the active speaker in high resolution, but show
the WebRTC application using some API, if the application is to be low resolution "thumbnails" of other participants. Such systems
able to track changes in session participation. It is desirable to might configure the end-points to send simulcast high- and low-
map CSRC values back into WebRTC MediaStream identities as they cross resolution versions of their video using separate RTP sessions, to
this API, to avoid exposing the SSRC/CSRC name space to JavaScript simplify the operation of the central mixer In the WebRTC context
applications. this appears to be most easily accomplished by establishing
multiple PeerConnection all being feed the same set of WebRTC
MediaStreams. Each PeerConnection is then configured to deliver a
particular media quality and thus media bit-rate, and will produce
an independently encoded version with the codec parameters agreed
specifically in the context of that PeerConnection. The central
mixer can always distinguish packets corresponding to the low- and
high-resolution streams by inspecting their SSRC, RTP payload
type, or some other information contained in RTP header extensions
or RTCP packets, but it can be easier to distinguish the flows if
they arrive on separate RTP sessions on separate UDP ports.
If the mixer-to-client audio level extension [RFC6465] is being used To directly connect with multiple peers: A multi-party conference
in the session (see Section 5.2.3), the information in the CSRC list does not need to use a central mixer. Rather, a multi-unicast
is augmented by audio level information for each contributing source. mesh can be created, comprising several distinct RTP sessions,
This information can usefully be exposed in the user interface. with each participant sending RTP traffic over a separate RTP
session (that is, using an independent an PeerConnection object)
to every other participant, as shown in Figure 1. This topology
has the benefit of not requiring a central mixer node that is
trusted to access and manipulate the media data. The downside is
that it increases the used bandwidth at each sender by requiring
one copy of the RTP media streams for each participant that are
part of the same session beyond the sender itself.
This memo does not require implementations to be able to add a CSRC The multi-unicast topology could also be implemented as a single
list to outgoing RTP packets. It is expected that the any CSRC list RTP session, spanning multiple peer-to-peer transport layer
will be added by a mixer or other middlebox that performs in-network connections, or as several pairwise RTP sessions, one between each
processing of RTP streams. If there is a desire to allow end-system pair of peers. To maintain a coherent mapping between the
mixing, the requirement in Section 4.1 will need to be updated to relation between RTP sessions and PeerConnection objects we
support setting the CSRC list in outgoing RTP data packets. recommend that this is implemented as several individual RTP
sessions. The only downside is that end-point A will not learn of
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,
whereas it would it all three participants were part of a single
RTP session. Experience with the Mbone tools (experimental RTP-
based multicast conferencing tools from the late 1990s) has showed
that RTCP reception quality reports for third parties can usefully
be presented to the users in a way that helps them understand
asymmetric network problems, and the approach of using separate
RTP sessions prevents this. However, an advantage of using
separate RTP sessions is that it enables using different media
bit-rates and RTP session configurations between the different
peers, thus not forcing B to endure the same quality reductions if
there are limitations in the transport from A to C as C will. It
it believed that these advantages outweigh the limitations in
debugging power.
12.6. Media Synchronization To indirectly connect with multiple peers: A common scenario in
multi-party conferencing is to create indirect connections to
multiple peers, using an RTP mixer, translator, or some other type
of RTP middlebox. Figure 2 outlines a simple topology that might
be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP media streams from
certain perspectives, either by only sending some of the received
RTP media stream to any given receiver, or by providing a combined
RTP media stream out of a set of contributing streams.
When an end-point sends media from more than one media source, it There are various methods of implementation for the middlebox. If
needs to consider if (and which of) these media sources are to be implemented as a standard RTP mixer or translator, a single RTP
synchronized. In RTP/RTCP, synchronisation is provided by having a session will extend across the middlebox and encompass all the
set of RTP media streams be indicated as coming from the same end-points in one multi-party session. Other types of middlebox
synchronisation context and logical end-point by using the same CNAME might use separate RTP sessions between each end-point and the
identifier. middlebox. A common aspect is that these central nodes can use a
number of tools to control the media encoding provided by a WebRTC
end-point. This includes functions like requesting breaking the
encoding chain and have the encoder produce a so called Intra
frame. Another is limiting the bit-rate of a given stream to
better suit the mixer view of the multiple down-streams. Others
are controlling the most suitable frame-rate, picture resolution,
the trade-off between frame-rate and spatial quality. The
middlebox gets the significant responsibility to correctly perform
congestion control, source identification, manage synchronization
while providing the application with suitable media optimizations.
The middlebox is also has to be a trusted node when it comes to
security, since it manipulates either the RTP header or the media
itself (or both) received from one end-point, before sending it on
towards the end-point(s), thus they need to be able to decrypt and
then encrypt it before sending it out.
The next provision is that the internal clocks of all media sources, RTP Mixers can create a situation where an end-point experiences a
i.e., what drives the RTP timestamp, can be correlated to a system situation in-between a session with only two end-points and
clock that is provided in RTCP Sender Reports encoded in an NTP multiple RTP sessions. Mixers are expected to not forward RTCP
format. By correlating all RTP timestamps to a common system clock reports regarding RTP media streams across themselves. This is
for all sources, the timing relation of the different RTP media due to the difference in the RTP media streams provided to the
streams, also across multiple RTP sessions can be derived at the different end-points. The original media source lacks information
receiver and, if desired, the streams can be synchronized. The about a mixer's manipulations prior to sending it the different
requirement is for the media sender to provide the correlation receivers. This scenario also results in that an end-point's
information; it is up to the receiver to use it or not. feedback or requests goes to the mixer. When the mixer can't act
on this by itself, it is forced to go to the original media source
to fulfil the receivers request. This will not necessarily be
explicitly visible any RTP and RTCP traffic, but the interactions
and the time to complete them will indicate such dependencies.
12.7. Multiple RTP End-points Providing source authentication in multi-party scenarios is a
challenge. In the mixer-based topologies, end-points source
authentication is based on, firstly, verifying that media comes
from the mixer by cryptographic verification and, secondly, trust
in the mixer to correctly identify any source towards the end-
point. In RTP sessions where multiple end-points are directly
visible to an end-point, all end-points will have knowledge about
each others' master keys, and can thus inject packets claimed to
come from another end-point in the session. Any node performing
relay can perform non-cryptographic mitigation by preventing
forwarding of packets that have SSRC fields that came from other
end-points before. For cryptographic verification of the source
SRTP would require additional security mechanisms, for example
TESLA for SRTP [RFC4383], that are not part of the base WebRTC
standards.
Some usages of RTP beyond the recommend topologies result in that an To forward media between multiple peers: It might be desirable for
WebRTC end-point sending media in an RTP session out over a single an end-point that receives an RTP media stream to be able to
PeerConnection will receive receiver reports from multiple RTP forward that media stream to a third party. The are obvious
receivers. Note that receiving multiple receiver reports is expected security and privacy implications in this, but also potential
because any RTP node that has multiple SSRCs has to report to the uses. If it is to be allowed, there are two implementation
media sender. The difference here is that they are multiple nodes, strategies: either the browser can relay the flow at the RTP
and thus will likely have different path characteristics. layer, or it transcode and forward the media at the application
layer.
RTP Mixers can create a situation where an end-point experiences a A relay approach will result in the RTP session be extended beyond
situation in-between a session with only two end-points and multiple the PeerConnection, making both the original end-point and the
end-points. Mixers are expected to not forward RTCP reports destination to which the media is forwarded part of the RTP
regarding RTP media streams across themselves. This is due to the session. These end-points can have different path
difference in the RTP media streams provided to the different end- characteristics, and hence different reception quality. Thus
points. The original media source lacks information about a mixer's sender's congestion control needs to be capable of handling this.
manipulations prior to sending it the different receivers. This The security solution can either support mechanism that the sender
scenario also results in that an end-point's feedback or requests informs both receivers of the key; alternatively the end-point
goes to the mixer. When the mixer can't act on this by itself, it is that is forwarding the media needs to decrypt and then re-encrypt
forced to go to the original media source to fulfil the receivers using a new key. The relay based approach has the advantage that
request. This will not necessarily be explicitly visible any RTP and the forwarding end-point does not need to transcode the media,
RTCP traffic, but the interactions and the time to complete them will thus maintaining the quality of the encoding and reducing the
indicate such dependencies. computational complexity requirements. If the right security
solutions are supported then the end-point that receives the
forwarded media will be able to verify the authenticity of the
media coming from the original sender. A downside is that the
original sender is forced to take both receivers into
consideration when delivering content.
The topologies in which an end-point receives receiver reports from The media transcoder approach is similar to having the forwarding
multiple other end-points are the centralized relay, multicast and an end-point act as Mixer, terminating the RTP session, combined with
end-point forwarding an RTP media stream. Having multiple RTP nodes a transcoder. The original sender will only see a single receiver
receive an RTP flow and send reports and feedback about it has of its media. The receiving end-point will responsible to produce
several impacts. As previously discussed (Section 12.3) any codec a RTP media stream suitable for onwards transmission. This might
control and rate control needs to be capable of merging the require media transcoding for congestion control purpose to
requirements and preferences to provide a single best encoding produce a suitable bit-rate. Thus loosing media quality in the
according to the situation RTP media stream. Specifically, when it transcoding and forcing the forwarding end-point to spend the
comes to congestion control it needs to be capable of identifying the resource on the transcoding. The media transcoding does result in
different end-points to form independent congestion state information a separation of the two different legs removing almost all
for each different path. dependencies, and allowing the forwarding end-point to optimize
its media transcoding operation. It also allows forwarding
without the original sender being aware of the forwarding. The
cost is greatly increased computational complexity on the
forwarding node.
Providing source authentication in multi-party scenarios is a (tbd: ought media forwarding be allowed?)
challenge. In the mixer-based topologies, end-points source
authentication is based on, firstly, verifying that media comes from
the mixer by cryptographic verification and, secondly, trust in the
mixer to correctly identify any source towards the end-point. In RTP
sessions where multiple end-points are directly visible to an end-
point, all end-points will have knowledge about each others' master
keys, and can thus inject packets claimed to come from another end-
point in the session. Any node performing relay can perform non-
cryptographic mitigation by preventing forwarding of packets that
have SSRC fields that came from other end-points before. For
cryptographic verification of the source SRTP would require
additional security mechanisms, like TESLA for SRTP [RFC4383].
12.8. Simulcast +---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
This section discusses simulcast in the meaning of providing a node, Figure 1: Multi-unicast using several RTP sessions
for example a Mixer, with multiple different encoded versions of the
same media source. In the WebRTC context, this could be accomplished
in two ways. One is to establish multiple PeerConnection all being
feed the same set of WebRTC MediaStreams. Another method is to use
multiple WebRTC MediaStreams that are differently configured when it
comes to the media parameters. This would result in that multiple
different RTP Media Streams (SSRCs) being in used with different
encoding based on the same media source (camera, microphone).
When intending to use simulcast it is important that this is made +---+ +-------------+ +---+
explicit so that the end-points don't automatically try to optimize | A |<---->| |<---->| B |
away the different encodings and provide a single common version. +---+ | RTP mixer, | +---+
Thus, some explicit indications that the intent really is to have | translator, |
different media encodings is likely needed. It is to be noted that | or other |
it might be a central node, rather than an WebRTC end-point that +---+ | middlebox | +---+
would benefit from receiving simulcast media sources. | C |<---->| |<---->| D |
+---+ +-------------+ +---+
tbd: How to perform simulcast needs to be determined and the Figure 2: RTP mixer with only unicast paths
appropriate API or signalling for its usage needs to be defined.
12.9. Differentiated Treatment of Flows 12.1.3. Differentiated Treatment of Flows
There are use cases for differentiated treatment of RTP media There are use cases for differentiated treatment of RTP media
streams. Such differentiation can happen at several places in the streams. Such differentiation can happen at several places in the
system. First of all is the prioritization within the end-point system. First of all is the prioritization within the end-point
sending the media, which controls, both which RTP media streams that sending the media, which controls, both which RTP media streams that
will be sent, and their allocation of bit-rate out of the current will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control. available aggregate as determined by the congestion control.
It is expected that the WebRTC API will allow the application to It is expected that the WebRTC API will allow the application to
indicate relative priorities for different MediaStreamTracks. These indicate relative priorities for different MediaStreamTracks. These
skipping to change at page 31, line 17 skipping to change at page 30, line 37
streams and FEC. The importance of such associated RTP traffic flows streams and FEC. The importance of such associated RTP traffic flows
is dependent on the media type and codec used, in regards to how is dependent on the media type and codec used, in regards to how
robust that codec is to packet loss. However, a default policy might robust that codec is to packet loss. However, a default policy might
to be to use the same priority for associated RTP flows as for the to be to use the same priority for associated RTP flows as for the
primary RTP flow. primary RTP flow.
Secondly, the network can prioritize packet flows, including RTP Secondly, the network can prioritize packet flows, including RTP
media streams. Typically, differential treatment includes two steps, media streams. Typically, differential treatment includes two steps,
the first being identifying whether an IP packet belongs to a class the first being identifying whether an IP packet belongs to a class
that has to be treated differently, the second the actual mechanism that has to be treated differently, the second the actual mechanism
to prioritize packets. This is done according to three methods; to prioritize packets. This is done according to three methods:
DiffServ: The end-point marks a packet with a DiffServ code point to DiffServ: The end-point marks a packet with a DiffServ code point to
indicate to the network that the packet belongs to a particular indicate to the network that the packet belongs to a particular
class. class.
Flow based: Packets that need to be given a particular treatment are Flow based: Packets that need to be given a particular treatment are
identified using a combination of IP and port address. identified using a combination of IP and port address.
Deep Packet Inspection: A network classifier (DPI) inspects the Deep Packet Inspection: A network classifier (DPI) inspects the
packet and tries to determine if the packet represents a packet and tries to determine if the packet represents a
skipping to change at page 32, line 24 skipping to change at page 31, line 45
particular RTP media flow need to be marked. RTCP compound packets particular RTP media flow need to be marked. RTCP compound packets
with Sender Reports (SR), ought to be marked with the same priority with Sender Reports (SR), ought to be marked with the same priority
as the RTP media flow itself, so the RTCP-based round-trip time (RTT) as the RTP media flow itself, so the RTCP-based round-trip time (RTT)
measurements are done using the same flow priority as the media flow measurements are done using the same flow priority as the media flow
experiences. RTCP compound packets containing RR packet ought to be experiences. RTCP compound packets containing RR packet ought to be
sent with the priority used by the majority of the RTP media flows sent with the priority used by the majority of the RTP media flows
reported on. RTCP packets containing time-critical feedback packets reported on. RTCP packets containing time-critical feedback packets
can use higher priority to improve the timeliness and likelihood of can use higher priority to improve the timeliness and likelihood of
delivery of such feedback. delivery of such feedback.
12.2. Source, Flow, and Participant Identification
12.2.1. Media Streams
Each RTP media stream is identified by a unique synchronisation
source (SSRC) identifier. The SSRC identifier is carried in the RTP
data packets comprising a media stream, and is also used to identify
that stream in the corresponding RTCP reports. The SSRC is chosen as
discussed in Section 4.8. The first stage in demultiplexing RTP and
RTCP packets received at a WebRTC end-point is to separate the media
streams based on their SSRC value; once that is done, additional
demultiplexing steps can determine how and where to render the media.
RTP allows a mixer, or other RTP-layer middlebox, to combine media
flows from multiple sources to form a new media flow. The RTP data
packets in that new flow can include a Contributing Source (CSRC)
list, indicating which original SSRCs contributed to the combined
packet. As described in Section 4.1, implementations need to support
reception of RTP data packets containing a CSRC list and RTCP packets
that relate to sources present in the CSRC list. The CSRC list can
change on a packet-by-packet basis, depending on the mixing operation
being performed. Knowledge of what sources contributed to a
particular RTP packet can be important if the user interface
indicates which participants are active in the session. Changes in
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
changes in session participation. It is desirable to map CSRC values
back into WebRTC MediaStream identities as they cross this API, to
avoid exposing the SSRC/CSRC name space to JavaScript applications.
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
is augmented by audio level information for each contributing source.
This information can usefully be exposed in the user interface.
12.2.2. Media Streams: SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have
support for detecting and handling SSRC collisions, i.e., resolve the
conflict when two different end-points use the same SSRC value. This
requirement also applies to WebRTC end-points. There are several
scenarios where SSRC collisions can occur.
In a point-to-point session where each SSRC is associated with either
of the two end-points and where the main media carrying SSRC
identifier will be announced in the signalling channel, a collision
is less likely to occur due to the information about used SSRCs
provided by Source-Specific SDP Attributes [RFC5576]. Still if both
end-points start uses an new SSRC identifier prior to having
signalled it to the peer and received acknowledgement on the
signalling message, there can be collisions. The Source-Specific SDP
Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
or reject a end-points usage of an SSRC.
There could also appear SSRC values that are not signalled. This is
more likely than it appears as certain RTP functions need extra SSRCs
to provide functionality related to another (the "main") SSRC, for
example, SSRC multiplexed RTP retransmission [RFC4588]. In those
cases, an end-point can create a new SSRC that strictly doesn't need
to be announced over the signalling channel to function correctly on
both RTP and PeerConnection level.
The more likely case for SSRC collision is that multiple end-points
in a multiparty conference create new sources and signals those
towards the central server. In cases where the SSRC/CSRC are
propagated between the different end-points from the central node
collisions can occur.
Another scenario is when the central node manages to connect an end-
point's PeerConnection to another PeerConnection the end-point
already has, thus forming a loop where the end-point will receive its
own traffic. While 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 becomes even more problematic when media mixers,
and so on, are involved, where the stream received is a different
stream but still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long
as the same RTP session is extended across multiple PeerConnections
by a RTP middlebox. To resolve the more generic case where multiple
PeerConnections are interconnected, then identification of the media
source(s) part of a MediaStreamTrack being propagated across multiple
interconnected PeerConnection needs to be preserved across these
interconnections.
12.2.3. Media Synchronisation Context
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
synchronized. In RTP/RTCP, synchronisation is provided by having a
set of RTP media streams be indicated as coming from the same
synchronisation context and logical end-point by using the same RTCP
CNAME identifier.
The next provision is that the internal clocks of all media sources,
i.e., what drives the RTP timestamp, can be correlated to a system
clock that is provided in RTCP Sender Reports encoded in an NTP
format. By correlating all RTP timestamps to a common system clock
for all sources, the timing relation of the different RTP media
streams, also across multiple RTP sessions can be derived at the
receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not.
12.2.4. Correlation of Media Streams
(tbd: this need to outline the approach to mapping media streams to
the signalling context defined in the unified plan)
(tbd: need to discuss correlation between associated RTP streams, for
example between a media stream and its associated FEC stream)
13. Security Considerations 13. Security Considerations
The overall security architecture for WebRTC is described in The overall security architecture for WebRTC is described in
[I-D.ietf-rtcweb-security-arch], and security considerations for the [I-D.ietf-rtcweb-security-arch], and security considerations for the
WebRTC framework are described in [I-D.ietf-rtcweb-security]. These WebRTC framework are described in [I-D.ietf-rtcweb-security]. These
considerations apply to this memo also. considerations apply to this memo also.
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.
skipping to change at page 33, line 26 skipping to change at page 35, line 15
15. Open Issues 15. Open Issues
This section contains a summary of the open issues or to be done This section contains a summary of the open issues or to be done
things noted in the document: things noted in the document:
1. tbd: The API mapping to RTP level concepts has to be agreed and 1. tbd: The API mapping to RTP level concepts has to be agreed and
documented in Section 11. documented in Section 11.
2. tbd: An open question if any requirements are needed to agree and 2. tbd: An open question if any requirements are needed to agree and
limit the number of simultaneously used media sources (SSRCs) limit the number of simultaneously used media sources (SSRCs)
within an RTP session. See Section 12.2 and Section 4.1. within an RTP session. See Section 4.1.
3. tbd: The method for achieving simulcast of a media source has to 3. tbd: The method for achieving simulcast of a media source has to
be decided as discussed in Section 12.8. be decided.
4. tbd: Possible documentation of what support for differentiated 4. tbd: Possible documentation of what support for differentiated
treatment that are needed on RTP level as the API and the network treatment that are needed on RTP level as the API and the network
level specification matures as discussed in Section 12.9. level specification matures as discussed in Section 12.1.3.
5. tbd: Editing of Appendix A to remove redundancy between this and
the update of RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
16. Acknowledgements 16. Acknowledgements
The authors would like to thank Harald Alvestrand, Cary Bran, Charles The authors would like to thank Harald Alvestrand, Cary Bran, Charles
Eckel, Cullen Jennings, Bernard Aboba, and the other members of the Eckel, Cullen Jennings, Bernard Aboba, and the other members of the
IETF RTCWEB working group for their valuable feedback. IETF RTCWEB working group for their valuable feedback.
17. References 17. References
17.1. Normative References 17.1. Normative References
[I-D.ietf-avtcore-6222bis] [I-D.ietf-avtcore-6222bis]
Begen, A., Perkins, C., Wing, D., and E. Rescorla, 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)", draft-ietf-avtcore-6222bis-06 Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
(work in progress), July 2013. (work in progress), July 2013.
[I-D.ietf-avtcore-avp-codecs] [I-D.ietf-avtcore-avp-codecs]
Terriberry, T., "Update to Remove DVI4 from the Terriberry, T., "Update to Remove DVI4 from the
skipping to change at page 34, line 27 skipping to change at page 36, line 6
[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-03 (work in ietf-avtcore-multi-media-rtp-session-03 (work in
progress), July 2013. progress), July 2013.
[I-D.ietf-avtcore-rtp-circuit-breakers] [I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control: Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf- Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-02 (work in progress), avtcore-rtp-circuit-breakers-03 (work in progress), July
February 2013. 2013.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ", Grouping RTCP Reception Statistics and Other Feedback ",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
in progress), July 2013. in progress), July 2013.
[I-D.ietf-avtcore-rtp-multi-stream] [I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
skipping to change at page 35, line 14 skipping to change at page 36, line 39
Petit-Huguenin, M. and G. Zorn, "Support for Multiple Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", draft-ietf-avtext- Clock Rates in an RTP Session", draft-ietf-avtext-
multiple-clock-rates-09 (work in progress), April 2013. multiple-clock-rates-09 (work in progress), April 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-04 (work in progress), June 2013. bundle-negotiation-04 (work in progress), June 2013.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work
in progress), February 2013.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-07 (work in progress), July 2013. rtcweb-security-arch-07 (work in progress), July 2013.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-05 (work in progress), July 2013. ietf-rtcweb-security-05 (work in progress), July 2013.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport", draft-westerlund-avtcore-
transport-multiplexing-05 (work in progress), February
2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, December Payload Format Specifications", BCP 36, RFC 2736, December
1999. 1999.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
skipping to change at page 37, line 21 skipping to change at page 38, line 39
[I-D.alvestrand-rtcweb-msid] [I-D.alvestrand-rtcweb-msid]
Alvestrand, H., "Cross Session Stream Identification in Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", draft-alvestrand- the Session Description Protocol", draft-alvestrand-
rtcweb-msid-02 (work in progress), May 2012. rtcweb-msid-02 (work in progress), May 2012.
[I-D.ietf-avt-srtp-ekt] [I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03 Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011. (work in progress), October 2011.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-00 (work in progress),
April 2013.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-07 (work
in progress), August 2013.
[I-D.ietf-rtcweb-qos] [I-D.ietf-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-rtcweb- other packet markings for RTCWeb QoS", draft-ietf-rtcweb-
qos-00 (work in progress), October 2012. qos-00 (work in progress), October 2012.
[I-D.ietf-rtcweb-use-cases-and-requirements] [I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft- Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-11 (work in ietf-rtcweb-use-cases-and-requirements-11 (work in
progress), June 2013. progress), June 2013.
skipping to change at page 37, line 43 skipping to change at page 39, line 22
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media", draft-jesup-rtp- Requirements For Real Time Media", draft-jesup-rtp-
congestion-reqs-00 (work in progress), March 2012. congestion-reqs-00 (work in progress), March 2012.
[I-D.westerlund-avtcore-multiplex-architecture] [I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Perkins, C., and H. Alvestrand, Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP", "Guidelines for using the Multiplexing Features of RTP",
draft-westerlund-avtcore-multiplex-architecture-03 (work draft-westerlund-avtcore-multiplex-architecture-03 (work
in progress), February 2013. in progress), February 2013.
[I-D.westerlund-avtcore-rtp-topologies-update] [I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and S. Wenger, "RTP Topologies", draft- Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
westerlund-avtcore-rtp-topologies-update-02 (work in Single Lower-Layer Transport", draft-westerlund-avtcore-
progress), February 2013. transport-multiplexing-05 (work in progress), February
2013.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003. 2003.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006. Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
skipping to change at page 38, line 41 skipping to change at page 40, line 19
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009. 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.
Appendix A. Supported RTP Topologies
RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies. Some
of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing. Properties
of some RTP topologies are discussed in
[I-D.westerlund-avtcore-rtp-topologies-update], and we further
describe those expected to be useful for WebRTC in the following. We
also goes into important RTP session aspects that the topology or
implementation variant can place on a WebRTC end-point.
This section includes RTP topologies beyond the RECOMMENDED ones.
This in an attempt to highlight the differences and the in many case
small differences in implementation to support a larger set of
possible topologies.
(tbd: This section needs reworking and clearer relation to
[I-D.westerlund-avtcore-rtp-topologies-update].)
A.1. Point to Point
The point-to-point RTP topology (Figure 3) is the simplest scenario
for WebRTC applications. This is going to be very common for user to
user calls.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 3: Point to Point
This being the basic one lets use the topology to high-light a couple
of details that are common for all RTP usage in the WebRTC context.
First is the intention to multiplex RTP and RTCP over the same UDP-
flow. Secondly is the question of using only a single RTP session or
one per media type for legacy interoperability. Thirdly is the
question of using multiple sender sources (SSRCs) per end-point.
Historically, RTP and RTCP have been run on separate UDP ports. With
the increased use of Network Address/Port Translation (NAPT) this has
become problematic, since maintaining multiple NAT bindings can be
costly. It also complicates firewall administration, since multiple
ports need to be opened to allow RTP traffic. To reduce these costs
and session set-up times, support for multiplexing RTP data packets
and RTCP control packets on a single port [RFC5761] will be
supported.
In cases where there is only one type of media (e.g., a voice-only
call) this topology will be implemented as a single RTP session, with
bidirectional flows of RTP and RTCP packets, all then multiplexed
onto a single 5-tuple. If multiple types of media are to be used
(e.g., audio and video), then each type media can be sent as a
separate RTP session using a different 5-tuple, allowing for separate
transport level treatment of each type of media. Alternatively, all
types of media can be multiplexed onto a single 5-tuple as a single
RTP session, or as several RTP sessions if using a demultiplexing
shim. Multiplexing different types of media onto a single 5-tuple
places some limitations on how RTP is used, as described in "RTP
Multiplexing Architecture"
[I-D.westerlund-avtcore-multiplex-architecture]. It is not expected
that these limitations will significantly affect the scenarios
targeted by WebRTC, but they can impact interoperability with legacy
systems.
An RTP session have good support for simultaneously transport
multiple media sources. Each media source uses an unique SSRC
identifier and each SSRC has independent RTP sequence number and
timestamp spaces. This is being utilized in WebRTC for several
cases. One is to enable multiple media sources of the same type, an
end-point that has two video cameras can potentially transmit video
from both to its peer(s). Another usage is when a single RTP session
is being used for both multiple media types, thus an end-point can
transmit both audio and video to the peer(s). Thirdly to support
multi-party cases as will be discussed below support for multiple
SSRC of the same media type is needed.
Thus we can introduce a couple of different notations in the below
two alternate figures of a single peer connection in a point to point
set-up. The first depicting a setup where the peer connection
established has two different RTP sessions, one for audio and one for
video. The second one using a single RTP session. In both cases A
has two video streams to send and one audio stream. B has only one
audio and video stream. These are used to illustrate the relation
between a peerConnection, the UDP flow(s), the RTP session(s) and the
SSRCs that will be used in the later cases also. In the below
figures RTCP flows are not included. They will flow bi-directionally
between any RTP session instances in the different nodes.
+-A-------------+ +-B-------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1----+ | | |
| | | | +-Audio-| |-Audio-+ | | | |
| | | | | AA1|---------------->| | | | | |
| | | | | |<----------------|BA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| | | | | |
| | +-UDP2------| |-UDP2------+ | |
| | | +-RTP2----| |-RTP1----+ | | |
| | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+---------------+ +---------------+
Figure 4: Point to Point: Multiple RTP sessions
As can be seen above in the Point to Point: Multiple RTP sessions
(Figure 4) the single Peer Connection contains two RTP sessions over
different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be
different, normally on source and destination ports. The first RTP
session (RTP1) carries audio, one stream in each direction AA1 and
BA1. The second RTP session contains two video streams from A (AV1
and AV2) and one from B to A (BV1).
+-A-------------+ +-B-------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1----+ | | |
| | | | +-Audio-| |-Audio-+ | | | |
| | | | | AA1|---------------->| | | | | |
| | | | | |<----------------|BA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | | | | | | | |
| | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+---------------+ +---------------+
Figure 5: Point to Point: Single RTP session.
In (Figure 5) there is only a single UDP flow and RTP session (RTP1).
This RTP session carries a total of five (5) RTP media streams
(SSRCs). From A to B there is Audio (AA1) and two video (AV1 and
AV2). From B to A there is Audio (BA1) and Video (BV1).
A.2. Multi-Unicast (Mesh)
For small multiparty calls, it is practical to set up a multi-unicast
topology (Figure 6). In this topology, each participant sends
individual unicast RTP/UDP/IP flows to each of the other participants
using independent PeerConnections in a full mesh.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 6: Multi-unicast
This topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by
requiring one copy of the RTP media streams for each participant that
are part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the
media is very low bandwidth. The multi-unicast topology could be
implemented as a single RTP session, spanning multiple peer-to-peer
transport layer connections, or as several pairwise RTP sessions, one
between each pair of peers. To maintain a coherent mapping between
the relation between RTP sessions and PeerConnections we recommend
that one implements this as individual RTP sessions. The only
downside is that end-point A will not learn of the quality of any
transmission happening between B and C based on RTCP. This has not
been seen as a significant downside as now one has yet seen a need
for why A would need to know about the B's and C's communication. An
advantage of using separate RTP sessions is that it enables using
different media bit-rates to the different peers, thus not forcing B
to endure the same quality reductions if there are limitations in the
transport from A to C as C will.
+-A------------------------+ +-B-------------+
|+---+ +-PeerC1------| |-PeerC1------+ |
||MIC| | +-UDP1------| |-UDP1------+ | |
|+---+ | | +-RTP1----| |-RTP1----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC1|--+-+-+-+--->AA1|------------->| | | | | |
| | +----+ | | | | |<-------------|BA1 | | | | |
| | | | | +-------| |-------+ | | | |
| | | | +---------| |---------+ | | |
| | | +-----------| |-----------+ | |
| | +-------------| |-------------+ |
| | | |---------------+
| | |
| | | +-C-------------+
| | +-PeerC2------| |-PeerC2------+ |
| | | +-UDP2------| |-UDP2------+ | |
| | | | +-RTP2----| |-RTP2----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC2|--+-+-+-+--->AA2|------------->| | | | | |
| +----+ | | | | |<-------------|CA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+--------------------------+ +---------------+
Figure 7: Session structure for Multi-Unicast Setup
Lets review how the RTP sessions looks from A's perspective by
considering both how the media is a handled and what PeerConnections
and RTP sessions that are set-up in Figure 7. A's microphone is
captured and the digital audio can then be feed into two different
encoder instances each beeing associated with two different
PeerConnections (PeerC1 and PeerC2) each containing independent RTP
sessions (RTP1 and RTP2). The SSRCs in each RTP session will be
completely independent and the media bit-rate produced by the encoder
can also be tuned to address any congestion control requirements
between A and B differently then for the path A to C.
For media encodings which are more resource consuming, like video,
one could expect that it will be common that end-points that are
resource constrained will use a different implementation strategy
where the encoder is shared between the different PeerConnections as
shown below Figure 8.
+-A----------------------+ +-B-------------+
|+---+ | | |
||CAM| +-PeerC1------| |-PeerC1------+ |
|+---+ | +-UDP1------| |-UDP1------+ | |
| | | | +-RTP1----| |-RTP1----+ | | |
| V | | | +-Video-| |-Video-+ | | | |
|+----+ | | | | |<----------------|BV1 | | | | |
||ENC |----+-+-+-+--->AV1|---------------->| | | | | |
|+----+ | | | +-------| |-------+ | | | |
| | | | +---------| |---------+ | | |
| | | +-----------| |-----------+ | |
| | +-------------| |-------------+ |
| | | |---------------+
| | |
| | | +-C-------------+
| | +-PeerC2------| |-PeerC2------+ |
| | | +-UDP2------| |-UDP2------+ | |
| | | | +-RTP2----| |-RTP2----+ | | |
| | | | | +-Video-| |-Video-+ | | | |
| +-------+-+-+-+--->AV2|---------------->| | | | | |
| | | | | |<----------------|CV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+------------------------+ +---------------+
Figure 8: Single Encoder Multi-Unicast Setup
This will clearly save resources consumed by encoding but does
introduce the need for the end-point A to make decisions on how it
encodes the media so it suites delivery to both B and C. This is not
limited to congestion control, also preferred resolution to receive
based on dispaly area available is another aspect requiring
consideration. The need for this type of decision logic does arise
in several different topologies and implementation.
A.3. Mixer Based
An mixer (Figure 9) is a centralised point that selects or mixes
content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer can
also reduce the bit-rate needed from the mixer down to a conference
participants as the media sent from the mixer to the end-point can be
optimised in different ways. These optimisations include methods
like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is needed instead
of 3 in the depicted scenario (Figure 9).
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 9: RTP Mixer with Only Unicast Paths
Mixers have two downsides, the first is that the mixer has to be a
trusted node as they either performs media operations or at least re-
packetize the media. Both type of operations requires when using
SRTP that the mixer verifies integrity, decrypts the content, perform
its operation and form new RTP packets, encrypts and integrity
protect them. This applies to all types of mixers described below.
The second downside is that all these operations and optimization of
the session requires processing. How much depends on the
implementation as will become evident below.
The implementation of an mixer can take several different forms and
we will discuss the main themes available that doesn't break RTP.
Please note that a Mixer could also contain translator
functionalities, like a media transcoder to adjust the media bit-rate
or codec used on a particular RTP media stream.
A.3.1. Media Mixing
This type of mixer is one which clearly can be called RTP mixer is
likely the one that most thinks of when they hear the term mixer.
Its basic patter of operation is that it will receive the different
participants RTP media stream. Select which that are to be included
in a media domain mix of the incoming RTP media streams. Then create
a single outgoing stream from this mix.
Audio mixing is straight forward and commonly possible to do for a
number of participants. Lets assume that you want to mix N number of
streams from different participants. Then the mixer need to perform
decoding N times. Then it needs to produce N or N+1 mixes, the
reasons that different mixes are needed are so that each contributing
source get a mix which don't contain themselves, as this would result
in an echo. When N is lower than the number of all participants one
can produce a Mix of all N streams for the group that are curently
not included in the mix, thus N+1 mixes. These audio streams are
then encoded again, RTP packetized and sent out.
Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed
video streams can be done. In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of
other sources as smaller is another. Also here one commonly need to
produce a number of different compositions so that the contributing
part doesn't need to see themselves. Then the mixer re-encodes the
created video stream, RTP packetize it and send it out
The problem with media mixing is that it both consume large amount of
media processing and encoding resources. The second is the quality
degradation created by decoding and re-encoding the RTP media stream.
Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA1 <----+ | | | +---+ | | |
| | | | | | |(BA1+CA1)|\| | | +---+ | | |
| | | | +-------| |---------+ +-+-+-|ENC|<-| B+C | |
| | | +---------| |-----------+ | | +---+ | | |
| | +-----------| |-------------+ | | M | |
| +-------------| |---------------+ | E | |
+---------------+ | | D | |
| | I | |
+-B-------------+ | | A | |
| +-PeerC2------| |-PeerC2--------+ | | |
| | +-UDP2------| |-UDP2--------+ | | M | |
| | | +-RTP2----| |-RTP2------+ | | | I | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | X | |
| | | | | BA1|------------>|---------+-+-+-+-|DEC|->| E | |
| | | | | |<------------|MA2 <----+ | | | +---+ | R | |
| | | | +-------| |(BA1+CA1)|\| | | +---+ | | |
| | | +---------| |---------+ +-+-+-|ENC|<-| A+C | |
| | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | | | |
+---------------+ |---------------+ | | |
| | | |
+-C-------------+ | | | |
| +-PeerC3------| |-PeerC3--------+ | | |
| | +-UDP3------| |-UDP3--------+ | | | |
| | | +-RTP3----| |-RTP3------+ | | | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | CA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA3 <----+ | | | +---+ | | |
| | | | +-------| |(BA1+CA1)|\| | | +---+ | | |
| | | +---------| |---------+ +-+-+-|ENC|<-| A+B | |
| | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | +-----+ |
+---------------+ |---------------+ |
+--------------------------------+
Figure 10: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 10. The mixer present one SSRC towards the
peer client, e.g. MA1 to Peer A, which is the media mix of the other
participants. As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure. It
has to do with the SSRC space and the identity information. When A
receives the MA1 stream which is a combination of BA1 and CA1 streams
in the other PeerConnections RTP could enable the mixer to include
CSRC information in the MA1 stream to identify the contributing
source BA1 and CA1.
The CSRC has in its turn utility in RTP extensions, like the in
Section 5.2.3 discussed Mixer to Client audio levels RTP header
extension [RFC6465]. If the SSRC from one PeerConnection are used as
CSRC in another PeerConnection then RTP1, RTP2 and RTP3 becomes one
joint session as they have a common SSRC space. At this stage one
also need to consider which RTCP information one need to expose in
the different legs. For the above situation commonly nothing more
than the Source Description (SDES) information and RTCP BYE for CSRC
need to be exposed. The main goal would be to enable the correct
binding against the application logic and other information sources.
This also enables loop detection in the RTP session.
A.3.1.1. RTP Session Termination
There exist an possible implementation choice to have the RTP
sessions being separated between the different legs in the multi-
party communication session and only generate RTP media streams in
each without carrying on RTP/RTCP level any identity information
about the contributing sources. This removes both the functionality
that CSRC can provide and the possibility to use any extensions that
build on CSRC and the loop detection. It might appear a
simplification if SSRC collision would occur between two different
end-points as they can be avoided to be resolved and instead remapped
between the independent sessions if at all exposed. However, SSRC/
CSRC remapping requires that SSRC/CSRC are never exposed to the
WebRTC JavaScript client to use as reference. This as they only have
local importance if they are used on a multi-party session scope the
result would be mis-referencing. Also SSRC collision handling will
still be needed as it can occur between the mixer and the end-point.
Session termination might appear to resolve some issues, it however
creates other issues that needs resolving, like loop detection,
identification of contributing sources and the need to handle mapped
identities and ensure that the right one is used towards the right
identities and never used directly between multiple end-points.
A.3.2. Media Switching
An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption
cycle as one rewrites RTP headers. This both reduces the amount of
computational resources needed in the mixer and increases the media
quality per transmitted bit. This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional
streams the mixer produces. These streams are created by selecting
media from one of the by the mixer received RTP media streams and
forward the media using the mixers own SSRCs. The mixer can then
switch between available sources if that is needed by the concept for
the source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field has to be set to the value of the Mixer's SSRC.
Secondly, the sequence number is set to the next in the sequence of
outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration might
have to be changed if the different PeerConnections have not arrived
on the same numbering for a given configuration. This also requires
that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still needed.
Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants. Thus the
mixer has two SSRCs sending video to each peer.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | AV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|MV1 <----+-+-+-+-BV1----| | |
| | | | | |<------------|MV2 <----+-+-+-+-EV1----| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | S | |
| +-------------| |---------------+ | W | |
+---------------+ | | I | |
| | T | |
+-B-------------+ | | C | |
| +-PeerC2------| |-PeerC2--------+ | H | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP2----| |-RTP2------+ | | | M | |
| | | | +-Video-| |-Video---+ | | | | A | |
| | | | | BV1|------------>|---------+-+-+-+------->| T | |
| | | | | |<------------|MV3 <----+-+-+-+-AV1----| R | |
| | | | | |<------------|MV4 <----+-+-+-+-EV1----| I | |
| | | | +-------| |---------+ | | | | X | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
: : : :
: : : :
+-F-------------+ | | | |
| +-PeerC6------| |-PeerC6--------+ | | |
| | +-UDP6------| |-UDP6--------+ | | | |
| | | +-RTP6----| |-RTP6------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|MV11 <---+-+-+-+-AV1----| | |
| | | | | |<------------|MV12 <---+-+-+-+-EV1----| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 11: Media Switching RTP Mixer
The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participations.
To ensure that a media receiver can correctly decode the RTP media
stream after a switch, it becomes necessary to ensure for state
saving codecs that they start from default state at the point of
switching. Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on
earlier state. This can be done using Full Intra Request RTCP codec
control message as discussed in Section 5.1.1.
Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different PeerConnection. The same
arguments and considerations as discussed in Appendix A.3.1.1 applies
here.
A.3.3. Media Projecting
Another method for handling media in the RTP mixer is to project all
potential sources (SSRCs) into a per end-point independent RTP
session. The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the
mixer in another RTP session receives media from that end-point.
This is similar to the media switching Mixer but have some important
differences in RTP details.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | AV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|BV1 <----+-+-+-+--------| | |
| | | | | |<------------|CV1 <----+-+-+-+--------| | |
| | | | | |<------------|DV1 <----+-+-+-+--------| | |
| | | | | |<------------|EV1 <----+-+-+-+--------| | |
| | | | | |<------------|FV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | S | |
| +-------------| |---------------+ | W | |
+---------------+ | | I | |
| | T | |
+-B-------------+ | | C | |
| +-PeerC2------| |-PeerC2--------+ | H | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP2----| |-RTP2------+ | | | M | |
| | | | +-Video-| |-Video---+ | | | | A | |
| | | | | BV1|------------>|---------+-+-+-+------->| T | |
| | | | | |<------------|AV1 <----+-+-+-+--------| R | |
| | | | | |<------------|CV1 <----+-+-+-+--------| I | |
| | | | | | : : : |: : : : : : : : : : :| X | |
| | | | | |<------------|FV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
: : : :
: : : :
+-F-------------+ | | | |
| +-PeerC6------| |-PeerC6--------+ | | |
| | +-UDP6------| |-UDP6--------+ | | | |
| | | +-RTP6----| |-RTP6------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|AV1 <----+-+-+-+--------| | |
| | | | | | : : : |: : : : : : : : : : :| | |
| | | | | |<------------|EV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 12: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 12)
one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1. If this mixer intend to have the same behavior as in
Appendix A.3.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
will concurrently transmit media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end-
points each RTP media stream will require some rewriting when being
projected from one session into another. The main thing is that the
sequence number will need to be consecutively incremented based on
the packet actually being transmitted in each RTP session. Thus the
RTP sequence number offset will change each time a source is turned
on in RTP session.
As the RTP sessions are independent the SSRC numbers used can be
handled independently also thus working around any SSRC collisions by
having remapping tables between the RTP sessions. However the
related WebRTC MediaStream signalling need to be correspondingly
changed to ensure consistent WebRTC MediaStream to SSRC mappings
between the different PeerConnections and the same comment that
higher functions MUST NOT use SSRC as references to RTP media streams
applies also here.
The mixer will also be responsible to act on any RTCP codec control
requests coming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need
to implement conference related codec control functionalities to
provide a good experience. Full Intra Request to request from the
media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the
source to mixer link.
This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can at any
time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to
cope with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.
A.4. Translator Based
There is also a variety of translators. The core commonality is that
they do not need to make themselves visible in the RTP level by
having an SSRC themselves. Instead they sit between one or more end-
point and perform translation at some level. It can be media
transcoding, protocol translation or covering missing functionality
for a legacy end-point or simply relay packets between transport
domains or to realize multi-party. We will go in details below.
A.4.1. Transcoder
A transcoder operates on media level and really used for two
purposes, the first is to allow two end-points that doesn't have a
common set of media codecs to communicate by translating from one
codec to another. The second is to change the bit-rate to a lower
one. For WebRTC end-points communicating with each other only the
first one is relevant. In certain legacy deployment media transcoder
will be necessary to ensure both codecs and bit-rate falls within the
envelope the legacy end-point supports.
As transcoding requires access to the media, the transcoder has to be
within the security context and access any media encryption and
integrity keys. On the RTP plane a media transcoder will in practice
fork the RTP session into two different domains that are highly
decoupled when it comes to media parameters and reporting, but not
identities. To maintain signalling bindings to SSRCs a transcoder is
likely needing to use the SSRC of one end-point to represent the
transcoded RTP media stream to the other end-point(s). The
congestion control loop can be terminated in the transcoder as the
media bit-rate being sent by the transcoder can be adjusted
independently of the incoming bit-rate. However, for optimizing
performance and resource consumption the translator needs to consider
what signals or bit-rate reductions it needs to send towards the
source end-point. For example receiving a 2.5 Mbps video stream and
then send out a 250 kbps video stream after transcoding is a waste of
resources. In most cases a 500 kbps video stream from the source in
the right resolution is likely to provide equal quality after
transcoding as the 2.5 Mbps source stream. At the same time
increasing media bit-rate further than what is needed to represent
the incoming quality accurate is also wasted resources.
+-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|----+ |
| | | | | |<------------|BA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<-+ | |
| | | +---------| |-----------+ | | +---+ | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
| | | |
+-B-------------+ | | | |
| +-PeerC2------| |-PeerC2--------+ | | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP1----| |-RTP1------+ | | | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | BA1|------------>|---------+-+-+-+-|DEC|--+ | |
| | | | | |<------------|AA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<---+ |
| | | +---------| |-----------+ | | +---+ |
| | +-----------| |-------------+ | |
| +-------------| |---------------+ |
+---------------+ +-----------------------------+
Figure 13: Media Transcoder
Figure 13 exposes some important details. First of all you can see
the SSRC identifiers used by the translator are the corresponding
end-points. Secondly, there is a relation between the RTP sessions
in the two different PeerConnections that are represented by having
both parts be identified by the same level and they need to share
certain contexts. Also certain type of RTCP messages will need to be
bridged between the two parts. Certain RTCP feedback messages are
likely needed to be sourced by the translator in response to actions
by the translator and its media encoder.
A.4.2. Gateway / Protocol Translator
Gateways are used when some protocol feature that are needed are not
supported by an end-point wants to participate in session. This RTP
translator in Figure 14 takes on the role of ensuring that from the
perspective of participant A, participant B appears as a fully
compliant WebRTC end-point (that is, it is the combination of the
Translator and participant B that looks like a WebRTC end point).
+------------+
| |
+---+ | Translator | +---+
| A |<---->| to legacy |<---->| B |
+---+ | end-point | +---+
WebRTC | | Legacy
+------------+
Figure 14: Gateway (RTP translator) towards legacy end-point
For WebRTC there are a number of requirements that could force the
need for a gateway if a WebRTC end-point is to communicate with a
legacy end-point, such as support of ICE and DTLS-SRTP for key
management. On RTP level the main functions that might be missing in
a legacy implementation that otherwise support RTP are RTCP in
general, SRTP implementation, congestion control and feedback
messages needed to make it work.
+-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1-----------------------+|
| | | | +-Audio-| |-Audio---+ ||
| | | | | AA1|------------>|---------+----------------+ ||
| | | | | |<------------|BA1 <----+--------------+ | ||
| | | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | +-------| |---------+ +---+-+ | | ||
| | | +---------| |---------------+| T | | | ||
| | +-----------| |-----------+ | || R | | | ||
| +-------------| |-------------+ || A | | | ||
+---------------+ | || N | | | ||
| || S | | | ||
+-B-(Legacy)----+ | || L | | | ||
| | | || A | | | ||
| +-UDP2------| |-UDP2------+ || T | | | ||
| | +-RTP1----| |-RTP1----------+| E | | | ||
| | | +-Audio-| |-Audio---+ +---+-+ | | ||
| | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | BA1|------------>|---------+--------------+ | ||
| | | | |<------------|AA1 <----+----------------+ ||
| | | +-------| |---------+ ||
| | +---------| |----------------------------+|
| +-----------| |-----------+ |
| | | |
+---------------+ +-----------------------------+
Figure 15: RTP/RTCP Protocol Translator
The legacy gateway can be implemented in several ways and what it
need to change is highly dependent on what functions it need to proxy
for the legacy end-point. One possibility is depicted in Figure 15
where the RTP media streams are compatible and forward without
changes. However, their RTP header values are captured to enable the
RTCP translator to create RTCP reception information related to the
leg between the end-point and the translator. This can then be
combined with the more basic RTCP reports that the legacy endpoint
(B) provides to give compatible and expected RTCP reporting to A.
Thus enabling at least full congestion control on the path between A
and the translator. If B has limited possibilities for congestion
response for the media then the translator might need the capability
to perform media transcoding to address cases where it otherwise
would need to terminate media transmission.
As the translator are generating RTP/RTCP traffic on behalf of B to A
it will need to be able to correctly protect these packets that it
translates or generates. Thus security context information are
needed in this type of translator if it operates on the RTP/RTCP
packet content or media. In fact one of the more likely scenario is
that the translator (gateway) will need to have two different
security contexts one towards A and one towards B and for each RTP/
RTCP packet do a authenticity verification, decryption followed by a
encryption and integrity protection operation to resolve mismatch in
security systems.
A.4.3. Relay
There exist a class of translators that operates on transport level
below RTP and thus do not effect RTP/RTCP packets directly. They
come in two distinct flavours, the one used to bridge between two
different transport or address domains to more function as a gateway
and the second one which is to to provide a group communication
feature as depicted below in Figure 16.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 16: RTP Translator (Relay) with Only Unicast Paths
The first kind is straight forward and is likely to exist in WebRTC
context when an legacy end-point is compatible with the exception for
ICE, and thus needs a gateway that terminates the ICE and then
forwards all the RTP/RTCP traffic and key management to the end-point
only rewriting the IP/UDP to forward the packet to the legacy node.
The second type is useful if one wants a less complex central node or
a central node that is outside of the security context and thus do
not have access to the media. This relay takes on the role of
forwarding the media (RTP and RTCP) packets to the other end-points
but doesn't perform any RTP or media processing. Such a device
simply forwards the media from each sender to all of the other
participants, and is sometimes called a transport-layer translator.
In Figure 16, participant A will only need to send a media once to
the relay, which will redistribute it by sending a copy of the stream
to participants B, C, and D. Participant A will still receive three
RTP streams with the media from B, C and D if they transmit
simultaneously. This is from an RTP perspective resulting in an RTP
session that behaves equivalent to one transporter over an IP Any
Source Multicast (ASM).
This results in one common RTP session between all participants
despite that there will be independent PeerConnections created to the
translator as depicted below Figure 17.
+-A-------------+ +-RELAY--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1-------------------------+ |
| | | | +-Video-| |-Video---+ | |
| | | | | AV1|------------>|---------------------------+ | |
| | | | | |<------------|BV1 <--------------------+ | | |
| | | | | |<------------|CV1 <------------------+ | | | |
| | | | +-------| |---------+ | | | | |
| | | +---------| |-------------------+ ^ ^ V | |
| | +-----------| |-------------+ | | | | | | |
| +-------------| |---------------+ | | | | | |
+---------------+ | | | | | | |
| | | | | | |
+-B-------------+ | | | | | | |
| +-PeerC2------| |-PeerC2--------+ | | | | | |
| | +-UDP2------| |-UDP2--------+ | | | | | | |
| | | +-RTP2----| |-RTP1--------------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | |
| | | | | BV1|------------>|-----------------------+ | | | |
| | | | | |<------------|AV1 <----------------------+ | |
| | | | | |<------------|CV1 <--------------------+ | | |
| | | | +-------| |---------+ | | | | |
| | | +---------| |-------------------+ | | | | |
| | +-----------| |-------------+ | | V ^ V | |
| +-------------| |---------------+ | | | | | |
+---------------+ | | | | | | |
: | | | | | |
: | | | | | |
+-C-------------+ | | | | | | |
| +-PeerC3------| |-PeerC3--------+ | | | | | |
| | +-UDP3------| |-UDP3--------+ | | | | | | |
| | | +-RTP3----| |-RTP1--------------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | |
| | | | | CV1|------------>|-------------------------+ | | |
| | | | | |<------------|AV1 <----------------------+ | |
| | | | | |<------------|BV1 <------------------+ | |
| | | | +-------| |---------+ | |
| | | +---------| |------------------------------+ |
| | +-----------| |-------------+ | |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 17: Transport Multi-party Relay
As the Relay RTP and RTCP packets between the UDP flows as indicated
by the arrows for the media flow a given WebRTC end-point, like A
will see the remote sources BV1 and CV1. There will be also two
different network paths between A, and B or C. This results in that
the client A has to be capable of handling that when determining
congestion state that there might exist multiple destinations on the
far side of a PeerConnection and that these paths have to be treated
differently. It also results in a requirement to combine the
different congestion states into a decision to transmit a particular
RTP media stream suitable to all participants.
It is also important to note that the relay can not perform selective
relaying of some sources and not others. The reason is that the RTCP
reporting in that case becomes inconsistent and without explicit
information about it being blocked has to be interpreted as severe
congestion.
In this usage it is also necessary that the session management has
configured a common set of RTP configuration including RTP payload
formats as when A sends a packet with pt=97 it will arrive at both B
and C carrying pt=97 and having the same packetization and encoding,
no entity will have manipulated the packet.
When it comes to security there exist some additional requirements to
ensure that the property that the relay can't read the media traffic
is enforced. First of all the key to be used has to be agreed such
so that the relay doesn't get it, e.g. no DTLS-SRTP handshake with
the relay, instead some other method needs to be used. Secondly, the
keying structure has to be capable of handling multiple end-points in
the same RTP session.
The second problem can basically be solved in two ways. Either a
common master key from which all derive their per source key for
SRTP. The second alternative which might be more practical is that
each end-point has its own key used to protects all RTP/RTCP packets
it sends. Each participants key are then distributed to the other
participants. This second method could be implemented using DTLS-
SRTP to a special key server and then use Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the
other participants in the RTP session Figure 18. The first one could
be achieved using MIKEY messages in SDP.
+---+ +---+
| | +-----------+ | |
| A |<------->| DTLS-SRTP |<------->| C |
| |<-- -->| HOST |<-- -->| |
+---+ \ / +-----------+ \ / +---+
X X
+---+ / \ +-----------+ / \ +---+
| |<-- -->| RTP |<-- -->| |
| B |<------->| RELAY |<------->| D |
| | +-----------+ | |
+---+ +---+
Figure 18: DTLS-SRTP host and RTP Relay Separated
The relay can still verify that a given SSRC isn't used or spoofed by
another participant within the multi-party session by binding SSRCs
on their first usage to a given source address and port pair.
Packets carrying that source SSRC from other addresses can be
suppressed to prevent spoofing. This is possible as long as SRTP is
used which leaves the SSRC of the packet originator in RTP and RTCP
packets in the clear. If such packet level method for enforcing
source authentication within the group, then there exist
cryptographic methods such as TESLA [RFC4383] that could be used for
true source authentication.
A.5. End-point Forwarding
An WebRTC end-point (B in Figure 19) will receive a WebRTC
MediaStream (set of SSRCs) over a PeerConnection (from A). For the
moment is not decided if the end-point is allowed or not to in its
turn send that WebRTC MediaStream over another PeerConnection to C.
This section discusses the RTP and end-point implications of allowing
such functionality, which on the API level is extremely simplistic to
perform.
+---+ +---+ +---+
| A |--->| B |--->| C |
+---+ +---+ +---+
Figure 19: MediaStream Forwarding
There exist two main approaches to how B forwards the media from A to
C. The first one is to simply relay the RTP media stream. The
second one is for B to act as a transcoder. Lets consider both
approaches.
A relay approach will result in that the WebRTC end-points will have
to have the same capabilities as being discussed in Relay
(Appendix A.4.3). Thus A will see an RTP session that is extended
beyond the PeerConnection and see two different receiving end-points
with different path characteristics (B and C). Thus A's congestion
control needs to be capable of handling this. The security solution
can either support mechanism that allows A to inform C about the key
A is using despite B and C having agreed on another set of keys.
Alternatively B will decrypt and then re-encrypt using a new key.
The relay based approach has the advantage that B does not need to
transcode the media thus both maintaining the quality of the encoding
and reducing B's complexity requirements. If the right security
solutions are supported then also C will be able to verify the
authenticity of the media coming from A. As downside A are forced to
take both B and C into consideration when delivering content.
The media transcoder approach is similar to having B act as Mixer
terminating the RTP session combined with the transcoder as discussed
in Appendix A.4.1. A will only see B as receiver of its media. B
will responsible to produce a RTP media stream suitable for the B to
C PeerConnection. This might require media transcoding for
congestion control purpose to produce a suitable bit-rate. Thus
loosing media quality in the transcoding and forcing B to spend the
resource on the transcoding. The media transcoding does result in a
separation of the two different legs removing almost all
dependencies. B could choice to implement logic to optimize its
media transcoding operation, by for example requesting media
properties that are suitable for C also, thus trying to avoid it
having to transcode the content and only forward the media payloads
between the two sides. For that optimization to be practical WebRTC
end-points have to support sufficiently good tools for codec control.
A.6. Simulcast
This section discusses simulcast in the meaning of providing a node,
for example a stream switching Mixer, with multiple different encoded
version of the same media source. In the WebRTC context that appears
to be most easily accomplished by establishing multiple
PeerConnection all being feed the same set of WebRTC MediaStreams.
Each PeerConnection is then configured to deliver a particular media
quality and thus media bit-rate. This will work well as long as the
end-point implements media encoding according to Figure 7. Then each
PeerConnection will receive an independently encoded version and the
codec parameters can be agreed specifically in the context of this
PeerConnection.
For simulcast to work one needs to prevent that the end-point deliver
content encoded as depicted in Figure 8. If a single encoder
instance is feed to multiple PeerConnections the intention of
performing simulcast will fail.
Thus it needs to be considered to explicitly signal which of the two
implementation strategies that are desired and which will be done.
At least making the application and possible the central node
interested in receiving simulcast of an end-points RTP media streams
to be aware if it will function or not.
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
 End of changes. 63 change blocks. 
1417 lines changed or deleted 447 lines changed or added

This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/