draft-ietf-avtcore-multiplex-guidelines-03.txt   draft-ietf-avtcore-multiplex-guidelines-04.txt 
Network Working Group M. Westerlund Network Working Group M. Westerlund
Internet-Draft B. Burman Internet-Draft B. Burman
Intended status: Informational Ericsson Intended status: Informational Ericsson
Expires: April 11, 2015 C. Perkins Expires: May 3, 2018 C. Perkins
University of Glasgow University of Glasgow
H. Alvestrand H. Alvestrand
Google Google
October 08, 2014 R. Even
H. Zheng
Huawei
October 30, 2017
Guidelines for using the Multiplexing Features of RTP to Support Guidelines for using the Multiplexing Features of RTP to Support
Multiple Media Streams Multiple Media Streams
draft-ietf-avtcore-multiplex-guidelines-03 draft-ietf-avtcore-multiplex-guidelines-04
Abstract Abstract
The Real-time Transport Protocol (RTP) is a flexible protocol that The Real-time Transport Protocol (RTP) is a flexible protocol that
can be used in a wide range of applications, networks, and system can be used in a wide range of applications, networks, and system
topologies. That flexibility makes for wide applicability, but can topologies. That flexibility makes for wide applicability, but can
complicate the application design process. One particular design complicate the application design process. One particular design
question that has received much attention is how to support multiple question that has received much attention is how to support multiple
media streams in RTP. This memo discusses the available options and media streams in RTP. This memo discusses the available options and
design trade-offs, and provides guidelines on how to use the design trade-offs, and provides guidelines on how to use the
multiplexing features of RTP to support multiple media streams. multiplexing features of RTP to support multiple media streams.
Status of This Memo Status of This Memo
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on April 11, 2015. This Internet-Draft will expire on May 3, 2018.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4 2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4
2.2. Subjects Out of Scope . . . . . . . . . . . . . . . . . . 6 2.2. Subjects Out of Scope . . . . . . . . . . . . . . . . . . 5
3. Reasons for Multiplexing and Grouping RTP Media Streams . . . 6 3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5
4. RTP Multiplexing Points . . . . . . . . . . . . . . . . . . . 7 3.1. Reasons for Multiplexing and Grouping RTP Media Streams . 5
4.1. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 8 3.2. RTP Multiplexing Points . . . . . . . . . . . . . . . . . 6
4.2. Synchronisation Source (SSRC) . . . . . . . . . . . . . . 9 3.2.1. RTP Session . . . . . . . . . . . . . . . . . . . . . 7
4.3. Contributing Source (CSRC) . . . . . . . . . . . . . . . 10 3.2.2. Synchronisation Source (SSRC) . . . . . . . . . . . . 8
4.4. RTP Payload Type . . . . . . . . . . . . . . . . . . . . 11 3.2.3. Contributing Source (CSRC) . . . . . . . . . . . . . 10
5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 12 3.2.4. RTP Payload Type . . . . . . . . . . . . . . . . . . 10
5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 12 3.3. Issues Related to RTP Topologies . . . . . . . . . . . . 11
5.2. Translators & Gateways . . . . . . . . . . . . . . . . . 13 3.4. Issues Related to RTP and RTCP Protocol . . . . . . . . . 13
5.3. Point to Multipoint Using Multicast . . . . . . . . . . . 13 3.4.1. The RTP Specification . . . . . . . . . . . . . . . . 13
5.4. Point to Multipoint Using an RTP Transport Translator . . 14 3.4.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 15
5.5. Point to Multipoint Using an RTP Mixer . . . . . . . . . 15 3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 15
6. RTP Multiplexing: When to Use Multiple RTP Sessions . . . . . 15 3.4.4. Forward Error Correction . . . . . . . . . . . . . . 17
6.1. RTP and RTCP Protocol Considerations . . . . . . . . . . 16 4. Particular Considerations for RTP Multiplexing . . . . . . . 17
6.1.1. The RTP Specification . . . . . . . . . . . . . . . . 16 4.1. Interworking Considerations . . . . . . . . . . . . . . . 17
6.1.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 18 4.1.1. Types of Interworking . . . . . . . . . . . . . . . . 17
6.1.3. Handling Varying Sets of Senders . . . . . . . . . . 19 4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 18
6.1.4. Cross Session RTCP Requests . . . . . . . . . . . . . 19 4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 18
6.1.5. Binding Related Sources . . . . . . . . . . . . . . . 19 4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 19
6.1.6. Forward Error Correction . . . . . . . . . . . . . . 21 4.2. Network Considerations . . . . . . . . . . . . . . . . . 20
6.1.7. Transport Translator Sessions . . . . . . . . . . . . 21 4.2.1. Quality of Service . . . . . . . . . . . . . . . . . 20
6.2. Interworking Considerations . . . . . . . . . . . . . . . 21 4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 20
6.2.1. Types of Interworking . . . . . . . . . . . . . . . . 22 4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 22
6.2.2. RTP Translator Interworking . . . . . . . . . . . . . 22 4.3. Security and Key Management Considerations . . . . . . . 23
6.2.3. Gateway Interworking . . . . . . . . . . . . . . . . 22 4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24
6.2.4. Multiple SSRC Legacy Considerations . . . . . . . . . 23 4.3.2. Key Management for Multi-party session . . . . . . . 24
6.3. Network Considerations . . . . . . . . . . . . . . . . . 24 4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25
6.3.1. Quality of Service . . . . . . . . . . . . . . . . . 24
6.3.2. NAT and Firewall Traversal . . . . . . . . . . . . . 25
6.3.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 27
6.3.4. Multiplexing multiple RTP Session on a Single
Transport . . . . . . . . . . . . . . . . . . . . . . 27
6.4. Security and Key Management Considerations . . . . . . . 28 5. Archetypes . . . . . . . . . . . . . . . . . . . . . . . . . 25
6.4.1. Security Context Scope . . . . . . . . . . . . . . . 28 5.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 25
6.4.2. Key Management for Multi-party session . . . . . . . 28 5.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 27
6.4.3. Complexity Implications . . . . . . . . . . . . . . . 29 5.3. Multiple Sessions for one Media type . . . . . . . . . . 28
7. Archetypes . . . . . . . . . . . . . . . . . . . . . . . . . 29 5.4. Multiple Media Types in one Session . . . . . . . . . . . 30
7.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 29 5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 31
7.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 31 6. Summary considerations and guidelines . . . . . . . . . . . . 31
7.3. Multiple Sessions for one Media type . . . . . . . . . . 32 6.1. Guidelines . . . . . . . . . . . . . . . . . . . . . . . 32
7.4. Multiple Media Types in one Session . . . . . . . . . . . 34 7. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 33
7.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 35 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
8. Summary considerations and guidelines . . . . . . . . . . . . 36 9. Security Considerations . . . . . . . . . . . . . . . . . . . 34
8.1. Guidelines . . . . . . . . . . . . . . . . . . . . . . . 36 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 34
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37 10.1. Normative References . . . . . . . . . . . . . . . . . . 34
10. Security Considerations . . . . . . . . . . . . . . . . . . . 37 10.2. Informative References . . . . . . . . . . . . . . . . . 34
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 37 Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 38
11.1. Normative References . . . . . . . . . . . . . . . . . . 37 Appendix B. Signalling considerations . . . . . . . . . . . . . 40
11.2. Informative References . . . . . . . . . . . . . . . . . 37 B.1. Signalling Aspects . . . . . . . . . . . . . . . . . . . 40
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 41 B.1.1. Session Oriented Properties . . . . . . . . . . . . . 40
Appendix B. Proposals for Future Work . . . . . . . . . . . . . 43 B.1.2. SDP Prevents Multiple Media Types . . . . . . . . . . 41
Appendix C. Signalling considerations . . . . . . . . . . . . . 44 B.1.3. Signalling Media Stream Usage . . . . . . . . . . . . 41
C.1. Signalling Aspects . . . . . . . . . . . . . . . . . . . 44 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42
C.1.1. Session Oriented Properties . . . . . . . . . . . . . 44
C.1.2. SDP Prevents Multiple Media Types . . . . . . . . . . 45
C.1.3. Signalling Media Stream Usage . . . . . . . . . . . . 45
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is a commonly used The Real-time Transport Protocol (RTP) [RFC3550] is a commonly used
protocol for real-time media transport. It is a protocol that protocol for real-time media transport. It is a protocol that
provides great flexibility and can support a large set of different provides great flexibility and can support a large set of different
applications. RTP has several multiplexing points designed for applications. RTP was from the beginning designed for multiple
different purposes. These enable support of multiple media streams participants in a communication session. It supports many paradigms
and switching between different encoding or packetization of the of topologies and usages, as defined in [RFC7667]. RTP has several
media. By using multiple RTP sessions, sets of media streams can be multiplexing points designed for different purposes. These enable
structured for efficient processing or identification. Thus the support of multiple media streams and switching between different
question for any RTP application designer is how to best use the RTP encoding or packetization of the media. By using multiple RTP
session, the SSRC and the payload type to meet the application's sessions, sets of media streams can be structured for efficient
needs. processing or identification. Thus the question for any RTP
application designer is how to best use the RTP session, the SSRC and
the payload type to meet the application's needs.
There have been increased interest in more advanced usage of RTP, for
example, multiple streams can occur when a single endpoint have
multiple media sources, like multiple cameras or microphones that
need to be sent simultaneously. Consequently, questions are raised
regarding the most appropriate RTP usage. The limitations in some
implementations, RTP/RTCP extensions, and signalling has also been
exposed. The authors also hope that clarification on the usefulness
of some functionalities in RTP will result in more complete
implementations in the future.
The purpose of this document is to provide clear information about The purpose of this document is to provide clear information about
the possibilities of RTP when it comes to multiplexing. The RTP the possibilities of RTP when it comes to multiplexing. The RTP
application designer needs to understand the implications that come application designer needs to understand the implications that come
from a particular usage of the RTP multiplexing points. The document from a particular usage of the RTP multiplexing points. The document
will recommend against some usages as being unsuitable, in general or will recommend against some usages as being unsuitable, in general or
for particular purposes. for particular purposes.
RTP was from the beginning designed for multiple participants in a
communication session. This is not restricted to multicast, as some
believe, but also provides functionality over unicast, using either
multiple transport flows below RTP or a network node that re-
distributes the RTP packets. The re-distributing node can for
example be a transport translator (relay) that forwards the packets
unchanged, a translator performing media or protocol translation in
addition to forwarding, or an RTP mixer that creates new sources from
the received streams. In addition, multiple streams can occur when a
single endpoint have multiple media sources, like multiple cameras or
microphones that need to be sent simultaneously.
This document has been written due to increased interest in more
advanced usage of RTP, resulting in questions regarding the most
appropriate RTP usage. The limitations in some implementations, RTP/
RTCP extensions, and signalling has also been exposed. It is
expected that some limitations will be addressed by updates or new
extensions resolving the shortcomings. The authors also hope that
clarification on the usefulness of some functionalities in RTP will
result in more complete implementations in the future.
The document starts with some definitions and then goes into the The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired existing RTP functionalities around multiplexing. Both the desired
behaviour and the implications of a particular behaviour depend on behaviour and the implications of a particular behaviour depend on
which topologies are used, which requires some consideration. This which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behaviour is followed by a discussion of some choices in multiplexing behaviour
and their impacts. Some archetypes of RTP usage are discussed. and their impacts. Some archetypes of RTP usage are discussed.
Finally, some recommendations and examples are provided. Finally, some recommendations and examples are provided.
2. Definitions 2. Definitions
2.1. Terminology 2.1. Terminology
The following terms and abbreviations are used in this document: The definitions in Section 3 of [RFC3550] are referenced normatively.
Endpoint: A single entity sending or receiving RTP packets. It can The taxonomy defined in [RFC7656] is referenced normatively.
be decomposed into several functional blocks, but as long as it
behaves a single RTP stack entity it is classified as a single The following terms and abbreviations are used in this document:
endpoint.
Multiparty: A communication situation including multiple endpoints. Multiparty: A communication situation including multiple endpoints.
In this document it will be used to refer to situations where more In this document it will be used to refer to situations where more
than two endpoints communicate. than two endpoints communicate.
Media Source: The source of a stream of data of one Media Type, It
can either be a single media capturing device such as a video
camera, a microphone, or a specific output of a media production
function, such as an audio mixer, or some video editing function.
Sending data from a Media Source can cause multiple RTP sources to
send multiple Media Streams.
Media Stream: A sequence of RTP packets using a single SSRC that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Source: The originator or source of a particular Media Stream. RTP Source: The originator or source of a particular Media Stream.
Identified using an SSRC in a particular RTP session. An RTP Identified using an SSRC in a particular RTP session. An RTP
source is the source of a single media stream, and is associated source is the source of a single media stream, and is associated
with a single endpoint and a single Media Source. An RTP Source with a single endpoint and a single Media Source. An RTP Source
is just called a Source in RFC 3550. is just called a Source in RFC 3550.
RTP Sink: A recipient of a Media Stream. The Media Sink is RTP Sink: A recipient of a Media Stream. The Media Sink is
identified using one or more SSRCs. There can be more than one identified using one or more SSRCs. There can be more than one
RTP Sink for one RTP source. RTP Sink for one RTP source.
CNAME: "Canonical name" - identifier associated with one or more RTP
sources from a single endpoint. Defined in the RTP specification
[RFC3550]. A CNAME identifies a synchronisation context. A CNAME
is associated with a single endpoint, although some RTP nodes will
use an endpoint's CNAME on that endpoints behalf. An endpoint can
use multiple CNAMEs. A CNAME is intended to be globally unique
and stable for the full duration of a communication session.
[RFC6222][I-D.ietf-avtcore-6222bis] gives updated guidelines for
choosing CNAMEs.
Media Type: Audio, video, text or data whose form and meaning are
defined by a specific real-time application.
Multiplexing: The operation of taking multiple entities as input, Multiplexing: The operation of taking multiple entities as input,
aggregating them onto some common resource while keeping the aggregating them onto some common resource while keeping the
individual entities addressable such that they can later be fully individual entities addressable such that they can later be fully
and unambiguously separated (de-multiplexed) again. and unambiguously separated (de-multiplexed) again.
RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can receive an SSRC
either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the endpoints' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by endpoints and any interconnecting middle nodes.
RTP Session Group: One or more RTP sessions that are used together RTP Session Group: One or more RTP sessions that are used together
to perform some function. Examples are multiple RTP sessions used to perform some function. Examples are multiple RTP sessions used
to carry different layers of a layered encoding. In an RTP to carry different layers of a layered encoding. In an RTP
Session Group, CNAMEs are assumed to be valid across all RTP Session Group, CNAMEs are assumed to be valid across all RTP
sessions, and designate synchronisation contexts that can cross sessions, and designate synchronisation contexts that can cross
RTP sessions. RTP sessions.
Source: Term that ought not be used alone. An RTP Source, as
identified by its SSRC, is the source of a single Media Stream; a
Media Source can be the source of mutiple Media Streams.
SSRC: A 32-bit unsigned integer used as identifier for a RTP Source.
CSRC: Contributing Source, A SSRC identifier used in a context, like
the RTP headers CSRC list, where it is clear that the Media Source
is not the source of the media stream, instead only a contributor
to the Media Stream.
Signalling: The process of configuring endpoints to participate in Signalling: The process of configuring endpoints to participate in
one or more RTP sessions. one or more RTP sessions.
2.2. Subjects Out of Scope 2.2. Subjects Out of Scope
This document is focused on issues that affect RTP. Thus, issues This document is focused on issues that affect RTP. Thus, issues
that involve signalling protocols, such as whether SIP, Jingle or that involve signalling protocols, such as whether SIP, Jingle or
some other protocol is in use for session configuration, the some other protocol is in use for session configuration, the
particular syntaxes used to define RTP session properties, or the particular syntaxes used to define RTP session properties, or the
constraints imposed by particular choices in the signalling constraints imposed by particular choices in the signalling
protocols, are mentioned only as examples in order to describe the protocols, are mentioned only as examples in order to describe the
RTP issues more precisely. RTP issues more precisely.
This document assumes the applications will use RTCP. While there This document assumes the applications will use RTCP. While there
are such applications that don't send RTCP, they do not conform to are such applications that don't send RTCP, they do not conform to
the RTP specification, and thus can be regarded as reusing the RTP the RTP specification, and thus can be regarded as reusing the RTP
packet format but not implementing the RTP protocol. packet format but not implementing the RTP protocol.
3. Reasons for Multiplexing and Grouping RTP Media Streams 3. RTP Multiplexing Overview
3.1. Reasons for Multiplexing and Grouping RTP Media Streams
The reasons why an endpoint might choose to send multiple media The reasons why an endpoint might choose to send multiple media
streams are widespread. In the below discussion, please keep in mind streams are widespread. In the below discussion, please keep in mind
that the reasons for having multiple media streams vary and include that the reasons for having multiple media streams vary and include
but are not limited to the following: but are not limited to the following:
o Multiple Media Sources o Multiple Media Sources
o Multiple Media Streams might be needed to represent one Media o Multiple Media Streams might be needed to represent one Media
Source (for instance when using layered encodings) Source (for instance when using layered encodings)
skipping to change at page 7, line 18 skipping to change at page 6, line 9
o Alternative Encodings, for instance different codecs for the same o Alternative Encodings, for instance different codecs for the same
audio stream audio stream
o Alternative formats, for instance multiple resolutions of the same o Alternative formats, for instance multiple resolutions of the same
video stream video stream
For each of these, it is necessary to decide if each additional media For each of these, it is necessary to decide if each additional media
stream gets its own SSRC multiplexed within a RTP Session, or if it stream gets its own SSRC multiplexed within a RTP Session, or if it
is necessary to use additional RTP sessions to group the media is necessary to use additional RTP sessions to group the media
streams. The choice between these made due to one reason might not streams. The choice between these made due to one reason might not
be the choice suitable for another reason. In the above list, the be the choice suitable for another reason. The clearest
different items have different levels of maturity in the discussion understanding is associated with multiple media sources of the same
on how to solve them. The clearest understanding is associated with media type. However, all warrant discussion and clarification on how
multiple media sources of the same media type. However, all warrant to deal with them. As the discussion below will show, in reality we
discussion and clarification on how to deal with them. As the cannot choose a single one of the two solutions. To utilise RTP well
discussion below will show, in reality we cannot choose a single one and as efficiently as possible, both are needed. The real issue is
of the two solutions. To utilise RTP well and as efficiently as finding the right guidance on when to create RTP sessions and when
possible, both are needed. The real issue is finding the right additional SSRCs in an RTP session is the right choice.
guidance on when to create RTP sessions and when additional SSRCs in
an RTP session is the right choice.
4. RTP Multiplexing Points 3.2. RTP Multiplexing Points
This section describes the multiplexing points present in the RTP This section describes the multiplexing points present in the RTP
protocol that can be used to distinguish media streams and groups of protocol that can be used to distinguish media streams and groups of
media streams. Figure 1 outlines the process of demultiplexing media streams. Figure 1 outlines the process of demultiplexing
incoming RTP streams: incoming RTP streams:
| |
| packets | packets
+-- v +-- v
| +------------+ | +------------+
| | Socket | | | Socket |
| +------------+ | +------------+
| || || | || ||
RTP | RTP/ || |+-----> SCTP ( ...and any other protocols) RTP | RTP/ || |+-----> SCTP ( ...and any other protocols)
Session | RTCP || +------> STUN (multiplexed using same port) Session | RTCP || +------> STUN (multiplexed using same port)
+-- || +-- ||
+-- || +-- ||
| (split by SSRC) | (split by SSRC)
| || || || | || || ||
| || || || | || || ||
Media | +--+ +--+ +--+ Media | +--+ +--+ +--+
Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, FEC, etc. Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, FEC, etc.
| +--+ +--+ +--+ | +--+ +--+ +--+
+-- | | | +-- | | |
(pick rending context based on PT)
(pick rendering context based on PT)
+-- | / | +-- | / |
| +---+ | | +---+ |
| / | | | / | |
Payload | +--+ +--+ +--+ Payload | +--+ +--+ +--+
Formats | |CR| |CR| |CR| Codecs and rendering Formats | |CR| |CR| |CR| Codecs and rendering
| +--+ +--+ +--+ | +--+ +--+ +--+
+-- +--
Figure 1: RTP Demultiplexing Process Figure 1: RTP Demultiplexing Process
4.1. RTP Session 3.2.1. RTP Session
An RTP Session is the highest semantic layer in the RTP protocol, and An RTP Session is the highest semantic layer in the RTP protocol, and
represents an association between a group of communicating endpoints. represents an association between a group of communicating endpoints.
The set of participants that form an RTP session is defined as those The set of participants that form an RTP session is defined as those
that share a single synchronisation source space [RFC3550]. That is, that share a single synchronisation source space [RFC3550]. That is,
if a group of participants are each aware of the synchronisation if a group of participants are each aware of the synchronisation
source identifiers belonging to the other participants, then those source identifiers belonging to the other participants, then those
participants are in a single RTP session. A participant can become participants are in a single RTP session. A participant can become
aware of a synchronisation source identifier by receiving an RTP aware of a synchronisation source identifier by receiving an RTP
packet containing it in the SSRC field or CSRC list, by receiving an packet containing it in the SSRC field or CSRC list, by receiving an
RTCP packet mentioning it in an SSRC field, or through signalling RTCP packet mentioning it in an SSRC field, or through signalling
(e.g., the SDP "a=ssrc:" attribute). Thus, the scope of an RTP (e.g., the SDP OCGBPa=ssrc:OCOe attribute). Thus, the scope of an
session is determined by the participants' network interconnection RTP session is determined by the participants' network
topology, in combination with RTP and RTCP forwarding strategies interconnection topology, in combination with RTP and RTCP forwarding
deployed by the endpoints and any middleboxes, and by the signalling. strategies deployed by the endpoints and any middleboxes, and by the
signalling.
RTP does not contain a session identifier. Rather, it relies on the RTP does not contain a session identifier. Rather, it relies on the
underlying transport layer to separate different sessions, and on the underlying transport layer to separate different sessions, and on the
signalling to identify sessions in a manner that is meaningful to the signalling to identify sessions in a manner that is meaningful to the
application. The signalling layer might give sessions an explicit application. The signalling layer might give sessions an explicit
identifier, or their identification might be implicit based on the identifier, or their identification might be implicit based on the
addresses and ports used. Accordingly, a single RTP Session can have addresses and ports used. Accordingly, a single RTP Session can have
multiple associated identifiers, explicit and implicit, belonging to multiple associated identifiers, explicit and implicit, belonging to
different contexts. For example, when running RTP on top of UDP/IP, different contexts. For example, when running RTP on top of UDP/IP,
an RTP endpoint can identify and delimit an RTP Session from other an RTP endpoint can identify and delimit an RTP Session from other
RTP Sessions using the UDP source and destination IP addresses and RTP Sessions using the UDP source and destination IP addresses and
UDP port numbers. Another example is when using SDP grouping UDP port numbers. Another example is when using SDP grouping
framework [RFC5888] which uses an identifier per "m="-line; if there framework [RFC5888] which uses an identifier per OCGBPm=OCOe-line; if
is a one-to-one mapping between "m="-lines and RTP sessions, that there is a one-to-one mapping between OCGBPm=OCOe-lines and RTP
grouping framework identifier will identify an RTP Session. sessions, that grouping framework identifier will identify an RTP
Session. [I-D.ietf-mmusic-sdp-bundle-negotiation] extends the
OCGBPm-OCGBP-line for bundled media, which adds complexity to
demultiplexing media stream. Section 10.2 of
[I-D.ietf-mmusic-sdp-bundle-negotiation] provides information about
how RTP/RTCP streams are associated with SDP media description.
RTP sessions are globally unique, but their identity can only be RTP sessions are globally unique, but their identity can only be
determined by the communication context at an endpoint of the determined by the communication context at an endpoint of the
session, or by a middlebox that is aware of the session context. The session, or by a middlebox that is aware of the session context. The
relationship between RTP sessions depending on the underlying relationship between RTP sessions depending on the underlying
application, transport, and signalling protocol. The RTP protocol application, transport, and signalling protocol. The RTP protocol
makes no normative statements about the relationship between makes no normative statements about the relationship between
different RTP sessions, however the applications that use more than different RTP sessions, however the applications that use more than
one RTP session will have some higher layer understanding of the one RTP session will have some higher layer understanding of the
relationship between the sessions they create. relationship between the sessions they create.
4.2. Synchronisation Source (SSRC) 3.2.2. Synchronisation Source (SSRC)
A synchronisation source (SSRC) identifies an RTP source or an RTP A synchronisation source (SSRC) identifies an RTP source or an RTP
sink. Every endpoint will have at least one synchronisation source sink. Every endpoint will have at least one synchronisation source
identifier, even if it does not send media (endpoints that are only identifier, even if it does not send media (endpoints that are only
RTP sinks still send RTCP, and use their synchronisation source RTP sinks still send RTCP, and use their synchronisation source
identifier in the RTCP packets they send). An endpoint can have identifier in the RTCP packets they send). An endpoint can have
multiple synchronisation sources identifiers if it contains multiple multiple synchronisation sources identifiers if it contains multiple
RTP sources (i.e., if it sends multiple media streams). Endpoints RTP sources (i.e., if it sends multiple media streams). Endpoints
that are both RTP sources and RTP sinks use the same synchronisation that are both RTP sources and RTP sinks use the same synchronisation
sources in both roles. At any given time, a RTP source has one and sources in both roles. At any given time, a RTP source has one and
only one SSRC - although that can change over the lifetime of the RTP only one SSRC - although that can change over the lifetime of the RTP
source or sink. source or sink.
The synchronisation Source identifier is a 32-bit unsigned integer. The synchronisation Source identifier is a 32-bit unsigned integer.
It is present in every RTP and RTCP packet header, and in the payload It is present in every RTP and RTCP packet header, and in the payload
of some RTCP packet types. It can also be present in SDP signalling. of some RTCP packet types. It can also be present in SDP signalling.
Unless pre-signalled using the SDP "a=ssrc:" attribute [RFC5576], the Unless pre-signalled using the SDP OCGBPa=ssrc:OCOe attribute
synchronisation source identifier is chosen at random. It is not [RFC5576], the synchronisation source identifier is chosen at random.
dependent on the network address of the endpoint, and is intended to It is not dependent on the network address of the endpoint, and is
be unique within an RTP session. Synchronisation source identifier intended to be unique within an RTP session. Synchronisation source
collisions can occur, and are handled as specified in [RFC3550] and identifier collisions can occur, and are handled as specified in
[RFC5576], resulting in the synchronisation source identifier of the [RFC3550] and [RFC5576], resulting in the synchronisation source
affecting RTP sources and/or sinks changing. An RTP source that identifier of the affecting RTP sources and/or sinks changing. An
changes its RTP Session identifier (e.g. source transport address) RTP source that changes its RTP Session identifier (e.g. source
during a session has to choose a new SSRC identifier to avoid being transport address) during a session has to choose a new SSRC
interpreted as looped source. identifier to avoid being interpreted as looped source.
Synchronisation source identifiers that belong to the same Synchronisation source identifiers that belong to the same
synchronisation context (i.e., that represent media streams that can synchronisation context (i.e., that represent media streams that can
be synchronised using information in RTCP SR packets) are indicated be synchronised using information in RTCP SR packets) are indicated
by use of identical CNAME chunks in corresponding RTCP SDES packets. by use of identical CNAME chunks in corresponding RTCP SDES packets.
SDP signalling can also be used to provide explicit grouping of SDP signalling can also be used to provide explicit grouping of
synchronisation sources [RFC5576]. synchronisation sources [RFC5576].
In some cases, the same SSRC Identifier value is used to relate In some cases, the same SSRC Identifier value is used to relate
streams in two different RTP Sessions, such as in Multi-Session streams in two different RTP Sessions, such as in Multi-Session
Transmission of scalable video [RFC6190]. This is NOT RECOMMENDED Transmission of scalable video [RFC6190]. This is to be avoided
since there is no guarantee of uniqueness in SSRC values across since there is no guarantee of uniqueness in SSRC values across
RTP sessions. RTP sessions.
Note that RTP sequence number and RTP timestamp are scoped by the Note that RTP sequence number and RTP timestamp are scoped by the
synchronisation source. Each RTP source will have a different synchronisation source. Each RTP source will have a different
synchronisation source, and the corresponding media stream will have synchronisation source, and the corresponding media stream will have
a separate RTP sequence number and timestamp space. a separate RTP sequence number and timestamp space.
An SSRC identifier is used by different type of sources as well as An SSRC identifier is used by different type of sources as well as
sinks: sinks:
Real Media Source: Connected to a "physical" media source, for Real Media Source: Connected to a OCGBPphysicalOCOe media source,
example a camera or microphone. for example a camera or microphone.
Processed Media Source: A source with some attributed property Processed Media Source: A source with some attributed property
generated by some network node, for example a filtering function generated by some network node, for example a filtering function
in an RTP mixer that provides the most active speaker based on in an RTP mixer that provides the most active speaker based on
some criteria, or a mix representing a set of other sources. some criteria, or a mix representing a set of other sources.
RTP Sink: A source that does not generate any RTP media stream in RTP Sink: A source that does not generate any RTP media stream in
itself (e.g. an endpoint or middlebox only receiving in an RTP itself (e.g. an endpoint or middlebox only receiving in an RTP
session). It still needs a sender SSRC for use as source in RTCP session). It still needs a sender SSRC for use as source in RTCP
reports. reports.
Note that a endpoint that generates more than one media type, e.g. a Note that an endpoint that generates more than one media type, e.g.
conference participant sending both audio and video, need not (and a conference participant sending both audio and video, need not (and
commonly does not) use the same SSRC value across RTP sessions. RTCP commonly does not) use the same SSRC value across RTP sessions. RTCP
Compound packets containing the CNAME SDES item is the designated Compound packets containing the CNAME SDES item is the designated
method to bind an SSRC to a CNAME, effectively cross-correlating method to bind an SSRC to a CNAME, effectively cross-correlating
SSRCs within and between RTP Sessions as coming from the same SSRCs within and between RTP Sessions as coming from the same
endpoint. The main property attributed to SSRCs associated with the endpoint. The main property attributed to SSRCs associated with the
same CNAME is that they are from a particular synchronisation context same CNAME is that they are from a particular synchronisation context
and can be synchronised at playback. and can be synchronised at playback.
An RTP receiver receiving a previously unseen SSRC value will An RTP receiver receiving a previously unseen SSRC value will
interpret it as a new source. It might in fact be a previously interpret it as a new source. It might in fact be a previously
existing source that had to change SSRC number due to an SSRC existing source that had to change SSRC number due to an SSRC
conflict. However, the originator of the previous SSRC ought to have conflict. However, the originator of the previous SSRC ought to have
ended the conflicting source by sending an RTCP BYE for it prior to ended the conflicting source by sending an RTCP BYE for it prior to
starting to send with the new SSRC, so the new SSRC is anyway starting to send with the new SSRC, so the new SSRC is anyway
effectively a new source. effectively a new source.
4.3. Contributing Source (CSRC) 3.2.3. Contributing Source (CSRC)
The Contributing Source (CSRC) is not a separate identifier. Rather The Contributing Source (CSRC) is not a separate identifier. Rather
a synchronisation source identifier is listed as a CSRC in the RTP a synchronisation source identifier is listed as a CSRC in the RTP
header of a packet generated by an RTP mixer if the corresponding header of a packet generated by an RTP mixer if the corresponding
SSRC was in the header of one of the packets that contributed to the SSRC was in the header of one of the packets that contributed to the
mix. mix.
It is not possible, in general, to extract media represented by an It is not possible, in general, to extract media represented by an
individual CSRC since it is typically the result of a media mixing individual CSRC since it is typically the result of a media mixing
(merge) operation by an RTP mixer on the individual media streams (merge) operation by an RTP mixer on the individual media streams
corresponding to the CSRC identifiers. The exception is the case corresponding to the CSRC identifiers. The exception is the case
when only a single CSRC is indicated as this represent forwarding of when only a single CSRC is indicated as this represent forwarding of
a media stream, possibly modified. The RTP header extension for a media stream, possibly modified. The RTP header extension for
Mixer-to-Client Audio Level Indication [RFC6465] expands on the Mixer-to-Client Audio Level Indication [RFC6465] expands on the
receivers information about a packet with a CSRC list. Due to these receivers information about a packet with a CSRC list. Due to these
restrictions, CSRC will not be considered a fully qualified restrictions, CSRC will not be considered a fully qualified
multiplexing point and will be disregarded in the rest of this multiplexing point and will be disregarded in the rest of this
document. document.
4.4. RTP Payload Type 3.2.4. RTP Payload Type
Each Media Stream utilises one or more RTP payload formats. An RTP Each Media Stream utilises one or more RTP payload formats. An RTP
payload format describes how the output of a particular media codec payload format describes how the output of a particular media codec
is framed and encoded into RTP packets. The payload format used is is framed and encoded into RTP packets. The payload format used is
identified by the payload type field in the RTP data packet header. identified by the payload type field in the RTP data packet header.
The combination therefore identifies a specific Media Stream encoding The combination therefore identifies a specific Media Stream encoding
format. The format definition can be taken from [RFC3551] for format. The format definition can be taken from [RFC3551] for
statically allocated payload types, but ought to be explicitly statically allocated payload types, but ought to be explicitly
defined in signalling, such as SDP, both for static and dynamic defined in signalling, such as SDP, both for static and dynamic
Payload Types. The term "format" here includes whatever can be Payload Types. The term OCGBPformatOCOe here includes whatever can
described by out-of-band signalling means. In SDP, the term "format" be described by out-of-band signalling means. In SDP, the term
includes media type, RTP timestamp sampling rate, codec, codec OCGBPformatOCOe includes media type, RTP timestamp sampling rate,
configuration, payload format configurations, and various robustness codec, codec configuration, payload format configurations, and
mechanisms such as redundant encodings [RFC2198]. various robustness mechanisms such as redundant encodings [RFC2198].
The payload type is scoped by sending endpoint within an RTP Session. The payload type is scoped by sending endpoint within an RTP Session.
All synchronisation sources sent from an single endpoint share the All synchronisation sources sent from a single endpoint share the
same payload types definitions. The RTP Payload Type is designed same payload types definitions. The RTP Payload Type is designed
such that only a single Payload Type is valid at any time instant in such that only a single Payload Type is valid at any time instant in
the RTP source's RTP timestamp time line, effectively time- the RTP source's RTP timestamp time line, effectively time-
multiplexing different Payload Types if any change occurs. The multiplexing different Payload Types if any change occurs. The
payload type used can change on a per-packet basis for an SSRC, for payload type used can change on a per-packet basis for an SSRC, for
example a speech codec making use of generic comfort noise [RFC3389]. example a speech codec making use of generic comfort noise [RFC3389].
If there is a true need to send multiple Payload Types for the same If there is a true need to send multiple Payload Types for the same
SSRC that are valid for the same instant, then redundant encodings SSRC that are valid for the same instant, then redundant encodings
[RFC2198] can be used. Several additional constraints than the ones [RFC2198] can be used. Several additional constraints than the ones
mentioned above need to be met to enable this use, one of which is mentioned above need to be met to enable this use, one of which is
that the combined payload sizes of the different Payload Types ought that the combined payload sizes of the different Payload Types ought
not exceed the transport MTU. not exceed the transport MTU.
Other aspects of RTP payload format use are described in RTP Payload Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto]. HowTo [RFC8088].
The payload type is not a multiplexing point at the RTP layer (see The payload type is not a multiplexing point at the RTP layer (see
Appendix A for a detailed discussion of why using the payload type as Appendix A for a detailed discussion of why using the payload type as
an RTP multiplexing point does not work). The RTP payload type is, an RTP multiplexing point does not work). The RTP payload type is,
however, used to determine how to render a media stream, and so can however, used to determine how to render a media stream, and so can
be viewed as selecting a rendering context. The rendering context be viewed as selecting a rendering context. The rendering context
can be defined by the signalling, and the RTP payload type number is can be defined by the signalling, and the RTP payload type number is
sometimes used to associate an RTP media stream with the signalling. sometimes used to associate an RTP media stream with the signalling.
This association is possible provided unique RTP payload type numbers This association is possible provided unique RTP payload type numbers
are used in each context. For example, an RTP media stream can be are used in each context. For example, an RTP media stream can be
associated with an SDP "m=" line by comparing the RTP payload type associated with an SDP OCGBPm=OCOe line by comparing the RTP payload
numbers used by the media stream with payload types signalled in the type numbers used by the media stream with payload types signalled in
"a=rtpmap:" lines in the media sections of the SDP. If RTP media the OCGBPa=rtpmap:OCOe lines in the media sections of the SDP. If
streams are being associated with signalling contexts based on the RTP media streams are being associated with signalling contexts based
RTP payload type, then the assignment of RTP payload type numbers on the RTP payload type, then the assignment of RTP payload type
MUST be unique across signalling contexts; if the same RTP payload numbers needs to be unique across signalling contexts; if the same
format configuration is used in multiple contexts, then a different RTP payload format configuration is used in multiple contexts, then a
RTP payload type number has to be assigned in each context to ensure different RTP payload type number has to be assigned in each context
uniqueness. If the RTP payload type number is not being used to to ensure uniqueness. If the RTP payload type number is not being
associated RTP media streams with a signalling context, then the same used to associated RTP media streams with a signalling context, then
RTP payload type number can be used to indicate the exact same RTP the same RTP payload type number can be used to indicate the exact
payload format configuration in multiple contexts. same RTP payload format configuration in multiple contexts. In case
of bundled media, Section 10.2 of
[I-D.ietf-mmusic-sdp-bundle-negotiation] provides more information on
SDP signalling.
5. RTP Topologies and Issues 3.3. Issues Related to RTP Topologies
The impact of how RTP multiplexing is performed will in general vary The impact of how RTP multiplexing is performed will in general vary
with how the RTP Session participants are interconnected, described with how the RTP Session participants are interconnected, described
by RTP Topology [RFC5117] and its intended successor by RTP Topology [RFC7667].
[I-D.westerlund-avtcore-rtp-topologies-update].
5.1. Point to Point
Even the most basic use case, denoted Topo-Point-to-Point in Even the most basic use case, denoted Topo-Point-to-Point in
[I-D.westerlund-avtcore-rtp-topologies-update], raises a number of [RFC7667], raises a number of considerations that are discussed in
considerations that are discussed in detail below (Section 6). They detail in following sections. They range over such aspects as:
range over such aspects as:
o Does my communication peer support RTP as defined with multiple o Does my communication peer support RTP as defined with multiple
SSRCs? SSRCs?
o Do I need network differentiation in form of QoS? o Do I need network differentiation in form of QoS?
o Can the application more easily process and handle the media o Can the application more easily process and handle the media
streams if they are in different RTP sessions? streams if they are in different RTP sessions?
o Do I need to use additional media streams for RTP retransmission o Do I need to use additional media streams for RTP retransmission
or FEC. or FEC.
o etc. o etc.
The point to point topology can contain one to many RTP sessions with For some Point to Multi-point topologies (e.g. Topo-ASM and Topo-SSM
one to many media sources per session, each having one or more RTP in [RFC7667]), multicast is used to interconnect the session
sources per media source. participants. Special considerations (documented in Section 4.2.3)
need to be made as multicast is a one to many distribution system.
5.2. Translators & Gateways
A point to point communication can end up in a situation when the Sometimes an RTP communication can end up in a situation when the
peer it is communicating with is not compatible with the other peer peer it is communicating with is not compatible with the other peer
for various reasons: for various reasons:
o No common media codec for a media type thus requiring transcoding o No common media codec for a media type thus requiring transcoding
o Different support for multiple RTP sources and RTP sessions o Different support for multiple RTP sources and RTP sessions
o Usage of different media transport protocols, i.e RTP or other. o Usage of different media transport protocols, i.e RTP or other.
o Usage of different transport protocols, e.g. UDP, DCCP, TCP o Usage of different transport protocols, e.g. UDP, DCCP, TCP
o Different security solutions, e.g. IPsec, TLS, DTLS, SRTP with o Different security solutions, e.g. IPsec, TLS, DTLS, SRTP with
different keying mechanisms. different keying mechanisms.
In many situations this is resolved by the inclusion of a translator In many situations this is resolved by the inclusion of a translator
between the two peers, as described by Topo-PtP-Translator in between the two peers, as described by Topo-PtP-Translator in
[I-D.westerlund-avtcore-rtp-topologies-update]. The translator's [RFC7667]. The translator's main purpose is to make the peer look to
main purpose is to make the peer look to the other peer like the other peer like something it is compatible with. There can also
something it is compatible with. There can also be other reasons be other reasons than compatibility to insert a translator in the
than compatibility to insert a translator in the form of a middlebox form of a middlebox or gateway, for example a need to monitor the
or gateway, for example a need to monitor the media streams. If the media streams. If the stream transport characteristics are changed
stream transport characteristics are changed by the translator, by the translator, appropriate media handling can require thorough
appropriate media handling can require thorough understanding of the understanding of the application logic, specifically any congestion
application logic, specifically any congestion control or media control or media adaptation.
adaptation.
5.3. Point to Multipoint Using Multicast
The Point to Multi-point topology is using Multicast to interconnect
the session participants. This includes both Topo-ASM and Topo-SSM
in [I-D.westerlund-avtcore-rtp-topologies-update].
Special considerations need to be made as multicast is a one to many
distribution system. For example, the only practical method for
adapting the bit-rate sent towards a given receiver for large groups
is to use a set of multicast groups, where each multicast group
represents a particular bit-rate. Otherwise the whole group gets
media adapted to the participant with the worst conditions. The
media encoding is either scalable, where multiple layers can be
combined, or simulcast, where a single version is selected. By
either selecting or combing multicast groups, the receiver can
control the bit-rate sent on the path to itself. It is also common
that streams that improve transport robustness are sent in their own
multicast group to allow for interworking with legacy or to support
different levels of protection.
The result of this is some common behaviours for RTP multicast:
1. Multicast applications use a group of RTP sessions, not one.
Each endpoint will need to be a member of a number of RTP
sessions in order to perform well.
2. Within each RTP session, the number of RTP Sinks is likely to be
much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement; all of
the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It is to be
noted that SDP offer/answer [RFC3264] is not appropriate for ensuring
this property. The signalling aspects of multicast are not explored
further in this memo.
Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol needs to support group communication. Source authentication
requires special solutions. For more discussion on this please
review Options for Securing RTP Sessions
[I-D.ietf-avtcore-rtp-security-options].
5.4. Point to Multipoint Using an RTP Transport Translator
This mode is described as Topo-Translator in
[I-D.westerlund-avtcore-rtp-topologies-update].
Transport Translators (Relays) result in an RTP session situation
that is very similar to how an ASM group RTP session would behave.
One of the most important aspects with the simple relay is that it is
only rewriting transport headers, no RTP modifications nor media
transcoding occur. The most obvious downside of this basic relaying
is that the translator has no control over how many streams need to
be delivered to a receiver. Nor can it simply select to deliver only
certain streams, as this creates session inconsistencies: If the
translator temporarily stops a stream, this prevents some receivers
from reporting on it. From the sender's perspective it will look
like a transport failure. Applications needing to stop or switch
streams in the central node ought to consider using an RTP mixer to
avoid this issue.
The Transport Translator has the same signalling requirement as
multicast: All participants need to have the same payload type
configuration. Most of the ASM security issues also arise here.
Some alternatives when it comes to solution do exist, as there exists
a central node to communicate with, one that also can enforce some
security policies depending on the level of trust placed in the node.
5.5. Point to Multipoint Using an RTP Mixer
A mixer, described by Topo-Mixer in
[I-D.westerlund-avtcore-rtp-topologies-update], is a centralised node
that selects or mixes content in a conference to optimise the RTP
session so that each endpoint only needs connect to one entity, the
mixer. The media sent from the mixer to the endpoint 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.
Mixers have some downsides, the first is that the mixer has to be a
trusted node as they repacketize the media, and can perform media
transformation operations. When using SRTP, both media operations
and repacketization requires that the mixer verifies integrity,
decrypts the content, performs the operation and forms new RTP
packets, encrypts and integrity-protects them. This applies to all
types of mixers. The second downside is that all these operations
and optimisations of the session requires processing. How much
depends on the implementation, as will become evident below.
A mixer, unlike a pure transport translator, is always application The point to point topology can contain one to many RTP sessions with
specific: the application logic for stream mixing or stream selection one to many media sources per session, each having one or more RTP
has to be embedded within the mixer, and controlled using application sources per media source.
specific signalling. The implementation of a mixer can take several
different forms, as discussed below.
A Mixer can also contain translator functionalities, like a media 3.4. Issues Related to RTP and RTCP Protocol
transcoder to adjust the media bit-rate or codec used for a
particular RTP media stream.
6. RTP Multiplexing: When to Use Multiple RTP Sessions
Using multiple media streams is a well supported feature of RTP. Using multiple media streams is a well supported feature of RTP.
However, it can be unclear for most implementers or people writing However, it can be unclear for most implementers or people writing
RTP/RTCP applications or extensions attempting to apply multiple RTP/RTCP applications or extensions attempting to apply multiple
streams when it is most appropriate to add an additional SSRC in an streams when it is most appropriate to add an additional SSRC in an
existing RTP session and when it is better to use multiple RTP existing RTP session and when it is better to use multiple RTP
sessions. This section tries to discuss the various considerations sessions. This section tries to discuss the various considerations
needed. The next section then concludes with some guidelines. needed.
6.1. RTP and RTCP Protocol Considerations
This section discusses RTP and RTCP aspects worth considering when
selecting between using an additional SSRC and Multiple RTP sessions.
6.1.1. The RTP Specification 3.4.1. The RTP Specification
RFC 3550 contains some recommendations and a bullet list with 5 RFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Let's review arguments for different aspects of RTP multiplexing. Let's review
Section 5.2 of [RFC3550], reproduced below: Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points OCGBPFor efficient protocol processing, the number of multiplexing
should be minimised, as described in the integrated layer processing points should be minimised, as described in the integrated layer
design principle [ALF]. In RTP, multiplexing is provided by the processing design principle [ALF]. In RTP, multiplexing is provided
destination transport address (network address and port number) which by the destination transport address (network address and port
is different for each RTP session. For example, in a teleconference number) which is different for each RTP session. For example, in a
composed of audio and video media encoded separately, each medium teleconference composed of audio and video media encoded separately,
SHOULD be carried in a separate RTP session with its own destination each medium SHOULD be carried in a separate RTP session with its own
transport address. destination transport address.
Separate audio and video streams SHOULD NOT be carried in a single Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems: using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the 1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings. way of identifying which stream had changed encodings.
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RTP session would avoid the first three problems but not the last RTP session would avoid the first three problems but not the last
two. two.
On the other hand, multiplexing multiple related sources of the same On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It might also be treatment is applicable for all of them. It might also be
appropriate to multiplex streams of the same medium using different appropriate to multiplex streams of the same medium using different
SSRC values in other scenarios where the last two problems do not SSRC values in other scenarios where the last two problems do not
apply." apply.OCOe
Let's consider one argument at a time. The first is an argument for Let's consider one argument at a time. The first is an argument for
using different SSRC for each individual media stream, which is very using different SSRC for each individual media stream, which is very
applicable. applicable.
The second argument is advocating against using payload type The second argument is advocating against using payload type
multiplexing, which still stands as can been seen by the extensive multiplexing, which still stands as can been seen by the extensive
list of issues found in Appendix A. list of issues found in Appendix A.
The third argument is yet another argument against payload type The third argument is yet another argument against payload type
multiplexing. multiplexing.
The fourth is an argument against multiplexing media streams that The fourth is an argument against multiplexing media streams that
require different handling into the same session. As we saw in the require different handling into the same session. As we saw in the
discussion of RTP mixers, the RTP mixer has to embed application discussion of RTP mixers, the RTP mixer has to embed application
logic in order to handle streams anyway; the separation of streams logic in order to handle streams anyway; the separation of streams
according to stream type is just another piece of application logic, according to stream type is just another piece of application logic,
which might or might not be appropriate for a particular application. which might or might not be appropriate for a particular application.
A type of application that can mix different media sources "blindly" A type of application that can mix different media sources
is the audio only "telephone" bridge; most other type of application OCGBPblindlyOCOe is the audio only OCGBPtelephoneOCOe bridge; most
needs application-specific logic to perform the mix correctly. other type of application needs application-specific logic to perform
the mix correctly.
The fifth argument discusses network aspects that we will discuss The fifth argument discusses network aspects that we will discuss
more below in Section 6.3. It also goes into aspects of more below in Section 4.2. It also goes into aspects of
implementation, like decomposed endpoints where different processes implementation, like decomposed endpoints where different processes
or inter-connected devices handle different aspects of the whole or inter-connected devices handle different aspects of the whole
multi-media session. multi-media session.
A summary of RFC 3550's view on multiplexing is to use unique SSRCs A summary of RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its own media/packet stream, and to use for anything that is its own media/packet stream, and to use
different RTP sessions for media streams that don't share a media different RTP sessions for media streams that don't share a media
type. This document supports the first point; it is very valid. The type. This document supports the first point; it is very valid. The
later is one thing which is further discussed in this document as later is one thing which needs to be further discussed, as imposing a
something the application developer needs to make a conscious choice single solution on all usages of RTP is inappropriate. Multiple
for, but where imposing a single solution on all usages of RTP is Media Types in an RTP Session specification
inappropriate. [I-D.ietf-avtcore-multi-media-rtp-session] provides a detailed
analysis of the potential issues in having multiple media types in
6.1.1.1. Different Media Types: Recommendations the same RTP session. This document tries to provide an wider scoped
The above quote from RTP [RFC3550] includes a strong recommendation:
"For example, in a teleconference composed of audio and video
media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address."
It was identified in "Why RTP Sessions Should Be Content Neutral"
[I-D.alvestrand-rtp-sess-neutral] that the above statement is poorly
supported by any of the motivations provided in the RTP
specification. This has resulted in the creation of a specification
Multiple Media Types in an RTP Session specification
[I-D.ietf-avtcore-multi-media-rtp-session] which intends to update
this recommendation. That document has a detailed analysis of the
potential issues in having multiple media types in the same RTP
session. This document tries to provide an moreover arching
consideration regarding the usage of RTP session and considers consideration regarding the usage of RTP session and considers
multiple media types in one RTP session as possible choice for the multiple media types in one RTP session as possible choice for the
RTP application designer. RTP application designer.
6.1.2. Multiple SSRCs in a Session 3.4.2. Multiple SSRCs in a Session
Using multiple SSRCs in an RTP session at one endpoint requires Using multiple SSRCs in an RTP session at one endpoint requires
resolving some unclear aspects of the RTP specification. These could resolving some unclear aspects of the RTP specification. These could
potentially lead to some interoperability issues as well as some potentially lead to some interoperability issues as well as some
potential significant inefficencies. These are further discussed in potential significant inefficiencies. These are further discussed in
"RTP Considerations for Endpoints Sending Multiple Media Streams" OCGBPRTP Considerations for Endpoints Sending Multiple Media
[I-D.lennox-avtcore-rtp-multi-stream]. A application designer needs StreamsOCOe [RFC8108]. A application designer needs to consider
to consider these issues and the impact availability or lack of the these issues and the impact availability or lack of the optimization
optimization in the endpoints has on their application. in the endpoints has on their application.
If an application will become affected by the issues described, using If an application will become affected by the issues described, using
Multiple RTP sessions can mitigate these issues. Multiple RTP sessions can mitigate these issues.
6.1.3. Handling Varying Sets of Senders 3.4.3. Binding Related Sources
In some applications, the set of simultaneously active sources varies
within a larger set of session members. A receiver can then possibly
try to use a set of decoding chains that is smaller than the number
of senders, switching the decoding chains between different senders.
As each media decoding chain can contain state, either the receiver
needs to either be able to save the state of swapped-out senders, or
the sender needs to be able to send data that permits the receiver to
reinitialise when it resumes activity.
This behaviour will cause similar issues independent of Additional
SSRC or Multiple RTP session.
6.1.4. Cross Session RTCP Requests
There currently exists no functionality to make truly synchronised
and atomic RTCP messages with some type of request semantics across
multiple RTP Sessions. Instead, separate RTCP messages will have to
be sent in each session. This gives streams in the same RTP session
a slight advantage as RTCP messages for different streams in the same
session can be sent in a compound RTCP packet, thus providing an
atomic operation if different modifications of different streams are
requested at the same time.
When using multiple RTP sessions, the RTCP timing rules in the
sessions and the transport aspects, such as packet loss and jitter,
prevents a receiver from relying on atomic operations, forcing it to
use more robust and forgiving mechanisms.
6.1.5. Binding Related Sources
A common problem in a number of various RTP extensions has been how A common problem in a number of various RTP extensions has been how
to bind related RTP sources and their media streams together. This to bind related RTP sources and their media streams together. This
issue is common to both using additional SSRCs and Multiple RTP issue is common to both using additional SSRCs and Multiple RTP
sessions. sessions.
The solutions can be divided into some groups, RTP/RTCP based, The solutions can be divided into some groups, RTP/RTCP based,
Signalling based (SDP), grouping related RTP sessions, and grouping Signalling based (SDP), grouping related RTP sessions, and grouping
SSRCs within an RTP session. Most solutions are explicit, but some SSRCs within an RTP session. Most solutions are explicit, but some
implicit methods have also been applied to the problem. implicit methods have also been applied to the problem.
skipping to change at page 19, line 51 skipping to change at page 16, line 4
The solutions can be divided into some groups, RTP/RTCP based, The solutions can be divided into some groups, RTP/RTCP based,
Signalling based (SDP), grouping related RTP sessions, and grouping Signalling based (SDP), grouping related RTP sessions, and grouping
SSRCs within an RTP session. Most solutions are explicit, but some SSRCs within an RTP session. Most solutions are explicit, but some
implicit methods have also been applied to the problem. implicit methods have also been applied to the problem.
The SDP-based signalling solutions are: The SDP-based signalling solutions are:
SDP Media Description Grouping: The SDP Grouping Framework [RFC5888] SDP Media Description Grouping: The SDP Grouping Framework [RFC5888]
uses various semantics to group any number of media descriptions. uses various semantics to group any number of media descriptions.
These has previously been considered primarily as grouping RTP These has previously been considered primarily as grouping RTP
sessions, but this might change. sessions, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple
media descriptors as a single RTP session.
SDP SSRC grouping: Source-Specific Media Attributes in SDP [RFC5576] SDP SSRC grouping: Source-Specific Media Attributes in SDP [RFC5576]
includes a solution for grouping SSRCs the same way as the includes a solution for grouping SSRCs the same way as the
Grouping framework groupes Media Descriptions. Grouping framework groups Media Descriptions.
SDP MSID grouping: Media Stream Identifiers [I-D.ietf-mmusic-msid] SDP MSID grouping: Media Stream Identifiers [I-D.ietf-mmusic-msid]
includes a solution for grouping SSRCs that is independent of includes a solution for grouping SSRCs that is independent of
their allocation to RTP sessions. their allocation to RTP sessions.
This supports a lot of use cases. All these solutions have This supports a lot of use cases. All these solutions have
shortcomings in cases where the session's dynamic properties are such shortcomings in cases where the session's dynamic properties are such
that it is difficult or resource consuming to keep the list of that it is difficult or resource consuming to keep the list of
related SSRCs up to date. related SSRCs up to date.
Within RTP/RTCP based solutions when binding to a endpoint or Within RTP/RTCP based solutions when binding to an endpoint or
synchronization context, i.e. the CNAME has not be sufficient and synchronization context, i.e. the CNAME has not been sufficient and
one has multiple RTP sessions has been to using the same SSRC value one way to bind related streams in multiple RTP sessions has been to
across all the RTP sessions. RTP Retransmission [RFC4588] is use the same SSRC value across all the RTP sessions. RTP
multiple RTP session mode, Generic FEC [RFC5109], as well as the RTP Retransmission [RFC4588] is multiple RTP session mode, Generic FEC
payload format for Scalable Video Coding [RFC6190] in Multi Session [RFC5109], as well as the RTP payload format for Scalable Video
Transmission (MST) mode uses this method. This method clearly works Coding [RFC6190] in Multi Session Transmission (MST) mode uses this
but might have some downside in RTP sessions with many participating method. This method clearly works but might have some downside in
SSRCs. The birthday paradox ensures that if you populate a single RTP sessions with many participating SSRCs. The birthday paradox
session with 9292 SSRCs at random, the chances are approximately 1% ensures that if you populate a single session with 9292 SSRCs at
that at least one collision will occur. When a collision occur this random, the chances are approximately 1% that at least one collision
will force one to change SSRC in all RTP sessions and thus will occur. When a collision occur this will force one to change
resynchronizing all of them instead of only the single media stream SSRC in all RTP sessions and thus resynchronizing all of them instead
having the collision. of only the single media stream having the collision. Therefore it
is not recommended to use such method. Using [RFC7656] streams from
the same media source should use the same RTP session.
It can be noted that Section 8.3 of the RTP Specification [RFC3550] It can be noted that Section 8.3 of the RTP Specification [RFC3550]
recommends using a single SSRC space across all RTP sessions for recommends using a single SSRC space across all RTP sessions for
layered coding. layered coding.
Another solution that has been applied to binding SSRCs has been an Another solution that has been applied to binding SSRCs has been an
implicit method used by RTP Retransmission [RFC4588] when doing implicit method used by RTP Retransmission [RFC4588] when doing
retransmissions in the same RTP session as the source RTP media retransmissions in the same RTP session as the source RTP media
stream. This issues an RTP retransmission request, and then await a stream. This issues an RTP retransmission request, and then await a
new SSRC carrying the RTP retransmission payload and where that SSRC new SSRC carrying the RTP retransmission payload and where that SSRC
is from the same CNAME. This limits a requestor to having only one is from the same CNAME. This limits a requestor to having only one
outstanding request on any new source SSRCs per endpoint. outstanding request on any new source SSRCs per endpoint.
There exists no RTP/RTCP based mechanism capable of supporting [I-D.ietf-mmusic-rid] provides an RTP/RTCP based mechanism capable of
explicit association accross multiple RTP sessions as well within an supporting explicit association within an RTP session.
RTP session. A proposed solution for handling this issue is
[I-D.westerlund-avtext-rtcp-sdes-srcname]. If accepted, this can
potentially also be part of an SDP based solution also by reusing the
same identifiers and name space.
6.1.6. Forward Error Correction 3.4.4. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes There exist a number of Forward Error Correction (FEC) based schemes
for how to reduce the packet loss of the original streams. Most of for how to reduce the packet loss of the original streams. Most of
the FEC schemes will protect a single source flow. The protection is the FEC schemes will protect a single source flow. The protection is
achieved by transmitting a certain amount of redundant information achieved by transmitting a certain amount of redundant information
that is encoded such that it can repair one or more packet losses that is encoded such that it can repair one or more packet losses
over the set of packets they protect. This sequence of redundant over the set of packets they protect. This sequence of redundant
information also needs to be transmitted as its own media stream, or information also needs to be transmitted as its own media stream, or
in some cases instead of the original media stream. Thus many of in some cases instead of the original media stream. Thus many of
these schemes create a need for binding related flows as discussed these schemes create a need for binding related flows as discussed
above. Looking at the history of these schemes, there are schemes above. Looking at the history of these schemes, there are schemes
using multiple SSRCs and schemes using multiple RTP sessions, and using multiple SSRCs and schemes using multiple RTP sessions, and
some schemes that support both modes of operation. some schemes that support both modes of operation.
Using multiple RTP sessions supports the case where some set of Using multiple RTP sessions supports the case where some set of
receivers might not be able to utilise the FEC information. By receivers might not be able to utilise the FEC information. By
placing it in a separate RTP session, it can easily be ignored. placing it in a separate RTP session, it can easily be ignored.
In usages involving multicast, having the FEC information on its own In usages involving multicast, having the FEC information on its own
multicast group, and therefore in its own RTP session, allows for multicast group allows for flexibility. This is especially useful
flexibility. This is especially useful when receivers see very when receivers see very heterogeneous packet loss rates. Those
heterogeneous packet loss rates. Those receivers that are not seeing receivers that are not seeing packet loss don't need to join the
packet loss don't need to join the multicast group with the FEC data, multicast group with the FEC data, and so avoid the overhead of
and so avoid the overhead of receiving unnecessary FEC packets, for receiving unnecessary FEC packets, for example.
example.
6.1.7. Transport Translator Sessions
A basic Transport Translator relays any incoming RTP and RTCP packets 4. Particular Considerations for RTP Multiplexing
to the other participants. The main difference between Additional
SSRCs and Multiple RTP Sessions resulting from this use case is that
with Additional SSRCs it is not possible for a particular session
participant to decide to receive a subset of media streams. When
using separate RTP sessions for the different sets of media streams,
a single participant can choose to leave one of the sessions but not
the other.
6.2. Interworking Considerations 4.1. Interworking Considerations
There are several different kinds of interworking, and this section There are several different kinds of interworking, and this section
discusses two related ones. The interworking between different discusses two related ones. The interworking between different
applications and the implications of potentially different choices of applications and the implications of potentially different choices of
usage of RTP's multiplexing points. The second topic relates to what usage of RTP's multiplexing points. The second topic relates to what
limitations have to be considered working with some legacy limitations have to be considered working with some legacy
applications. applications.
6.2.1. Types of Interworking 4.1.1. Types of Interworking
It is not uncommon that applications or services of similar usage, It is not uncommon that applications or services of similar usage,
especially the ones intended for interactive communication, encounter especially the ones intended for interactive communication, encounter
a situation where one want to interconnect two or more of these a situation where one want to interconnect two or more of these
applications. applications.
In these cases one ends up in a situation where one might use a In these cases one ends up in a situation where one might use a
gateway to interconnect applications. This gateway then needs to gateway to interconnect applications. This gateway then needs to
change the multiplexing structure or adhere to limitations in each change the multiplexing structure or adhere to limitations in each
application. application.
There are two fundamental approaches to gatewaying: RTP Translator There are two fundamental approaches to gatewaying: RTP Translator
interworking (RTP bridging), where the gateway acts as an RTP interworking (RTP bridging), where the gateway acts as an RTP
Translator, and the two applications are members of the same RTP Translator, and the two applications are members of the same RTP
session, and Gateway Interworking (with RTP termination), where there session, and Gateway Interworking (with RTP termination), where there
are independent RTP sessions running from each interconnected are independent RTP sessions running from each interconnected
application to the gateway. application to the gateway.
6.2.2. RTP Translator Interworking 4.1.2. RTP Translator Interworking
From an RTP perspective the RTP Translator approach could work if all From an RTP perspective the RTP Translator approach could work if all
the applications are using the same codecs with the same payload the applications are using the same codecs with the same payload
types, have made the same multiplexing choices, have the same types, have made the same multiplexing choices, have the same
capabilities in number of simultaneous media streams combined with capabilities in number of simultaneous media streams combined with
the same set of RTP/RTCP extensions being supported. Unfortunately the same set of RTP/RTCP extensions being supported. Unfortunately
this might not always be true. this might not always be true.
When one is gatewaying via an RTP Translator, a natural requirement When one is gatewaying via an RTP Translator, a natural requirement
is that the two applications being interconnected need to use the is that the two applications being interconnected need to use the
same approach to multiplexing. Furthermore, if one of the same approach to multiplexing. Furthermore, if one of the
applications is capable of working in several modes (such as being applications is capable of working in several modes (such as being
able to use Additional SSRCs or Multiple RTP sessions at will), and able to use Additional SSRCs or Multiple RTP sessions at will), and
the other one is not, successful interconnection depends on locking the other one is not, successful interconnection depends on locking
the more flexible application into the operating mode where the more flexible application into the operating mode where
interconnection can be successful, even if no participants using the interconnection can be successful, even if no participants using the
less flexible application are present when the RTP sessions are being less flexible application are present when the RTP sessions are being
created. created.
6.2.3. Gateway Interworking 4.1.3. Gateway Interworking
When one terminates RTP sessions at the gateway, there are certain When one terminates RTP sessions at the gateway, there are certain
tasks that the gateway has to carry out: tasks that the gateway has to carry out:
o Generating appropriate RTCP reports for all media streams o Generating appropriate RTCP reports for all media streams
(possibly based on incoming RTCP reports), originating from SSRCs (possibly based on incoming RTCP reports), originating from SSRCs
controlled by the gateway. controlled by the gateway.
o Handling SSRC collision resolution in each application's RTP o Handling SSRC collision resolution in each application's RTP
sessions. sessions.
o Signalling, choosing and policing appropriate bit-rates for each o Signalling, choosing and policing appropriate bit-rates for each
session. session.
If either of the applications has any security applied, e.g. in the For applications that uses any security mechanism, e.g. in the form
form of SRTP, the gateway needs to be able to decrypt incoming of SRTP, then the gateway needs to be able to decrypt incoming
packets and re-encrypt them in the other application's security packets and re-encrypt them in the other application's security
context. This is necessary even if all that's needed is a simple context. This is necessary even if all that's needed is a simple
remapping of SSRC numbers. If this is done, the gateway also needs remapping of SSRC numbers. If this is done, the gateway also needs
to be a member of the security contexts of both sides, of course. to be a member of the security contexts of both sides, of course.
Other tasks a gateway might need to apply include transcoding (for Other tasks a gateway might need to apply include transcoding (for
incompatible codec types), rescaling (for incompatible video size incompatible codec types), rescaling (for incompatible video size
requirements), suppression of content that is known not to be handled requirements), suppression of content that is known not to be handled
in the destination application, or the addition or removal of in the destination application, or the addition or removal of
redundancy coding or scalability layers to fit the need of the redundancy coding or scalability layers to fit the need of the
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This fact reveals the potential for these gateways to block evolution This fact reveals the potential for these gateways to block evolution
of the applications by blocking unknown RTP and RTCP extensions that of the applications by blocking unknown RTP and RTCP extensions that
the regular application has been extended with. the regular application has been extended with.
If one uses security functions, like SRTP, they can as seen above If one uses security functions, like SRTP, they can as seen above
incur both additional risk due to the gateway needing to be in incur both additional risk due to the gateway needing to be in
security association between the endpoints, unless the gateway is on security association between the endpoints, unless the gateway is on
the transport level, and additional complexities in form of the the transport level, and additional complexities in form of the
decrypt-encrypt cycles needed for each forwarded packet. SRTP, due decrypt-encrypt cycles needed for each forwarded packet. SRTP, due
to its keying structure, also requires that each RTP session needs to its keying structure, also requires that each RTP session needs
different master keys, as use of the same key in two RTP sessions can different master keys, as use of the same key in two RTP sessions for
result in two-time pads that completely breaks the confidentiality of some ciphers can result in two-time pads that completely breaks the
the packets. confidentiality of the packets.
4.1.4. Multiple SSRC Legacy Considerations
6.2.4. Multiple SSRC Legacy Considerations
Historically, the most common RTP use cases have been point to point Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per endpoint and media type (typically audio than one media source per endpoint and media type (typically audio
and video). Even in conferencing applications, especially voice and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each endpoint and the common to have individual RTP sessions between each endpoint and the
RTP mixer, meaning that the mixer functions as an RTP-terminating RTP mixer, meaning that the mixer functions as an RTP-terminating
gateway. gateway.
skipping to change at page 24, line 33 skipping to change at page 20, line 12
3. Be capable of rendering multiple streams simultaneously. 3. Be capable of rendering multiple streams simultaneously.
This indicates that gateways attempting to interconnect to this class This indicates that gateways attempting to interconnect to this class
of devices has to make sure that only one media stream of each type of devices has to make sure that only one media stream of each type
gets delivered to the endpoint if it's expecting only one, and that gets delivered to the endpoint if it's expecting only one, and that
the multiplexing format is what the device expects. It is highly the multiplexing format is what the device expects. It is highly
unlikely that RTP translator-based interworking can be made to unlikely that RTP translator-based interworking can be made to
function successfully in such a context. function successfully in such a context.
6.3. Network Considerations 4.2. Network Considerations
The multiplexing choice has impact on network level mechanisms that The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor. need to be considered by the implementer.
6.3.1. Quality of Service 4.2.1. Quality of Service
When it comes to Quality of Service mechanisms, they are either flow When it comes to Quality of Service mechanisms, they are either flow
based or marking based. RSVP [RFC2205] is an example of a flow based based or packet marking based. RSVP [RFC2205] is an example of a
mechanism, while Diff-Serv [RFC2474] is an example of a Marking based flow based mechanism, while Diff-Serv [RFC2474] is an example of a
one. For a marking based scheme, the method of multiplexing will not packet marking based one. For a packet marking based scheme, the
affect the possibility to use QoS. method of multiplexing will not affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between However, for a flow based scheme there is a clear difference between
the methods. Additional SSRC will result in all media streams being the methods. Additional SSRC will result in all media streams being
part of the same 5-tuple (protocol, source address, destination part of the same 5-tuple (protocol, source address, destination
address, source port, destination port) which is the most common address, source port, destination port) which is the most common
selector for flow based QoS. Thus, separation of the level of QoS selector for flow based QoS.
between media streams is not possible. That is however possible when
using multiple RTP sessions, where each media stream for which a
separate QoS handling is desired can be in a different RTP session
that can be sent over different 5-tuples.
It also needs to be noted that packet marking based QoS mechanisms It also needs to be noted that packet marking based QoS mechanisms
can have limitations. A general observation is that different DSCP can have limitations. A general observation is that different DSCP
can be assigned to different packets within in a flow as well as can be assigned to different packets within a flow as well as within
within an RTP Media Stream. However, care needs to be taken when an RTP Media Stream. However, care needs to be taken when
considering which forwarding behaviours that are applied on path due considering which forwarding behaviours that are applied on path due
to these DSCPs. In some cases the forwarding behaviour can result in to these DSCPs. In some cases the forwarding behaviour can result in
packet reordering. For more discussion of this see packet reordering. For more discussion of this see [RFC7657].
[I-D.ietf-dart-dscp-rtp].
More specific to the choice between using one or more RTP session can More specific to the choice between using one or more RTP session can
be the method for assigning marking to packets. If this is done be the method for assigning marking to packets. If this is done
using a network ingress function, it can have issues discriminating using a network ingress function, it can have issues discriminating
the different RTP media streams. The network API on the endpoint the different RTP media streams. The network API on the endpoint
also needs to be capable of setting the marking on a per packet basis also needs to be capable of setting the marking on a per packet basis
to reach the full functionality. to reach the full functionality.
6.3.2. NAT and Firewall Traversal 4.2.2. NAT and Firewall Traversal
In today's network there exist a large number of middleboxes. The In today's network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW). Translators (NAT) and Firewalls (FW).
Below we analyze and comment on the impact of requiring more Below we analyse and comment on the impact of requiring more
underlying transport flows in the presence of NATs and Firewalls: underlying transport flows in the presence of NATs and Firewalls:
End-Point Port Consumption: A given IP address only has 65536 End-Point Port Consumption: A given IP address only has 65536
available local ports per transport protocol for all consumers of available local ports per transport protocol for all consumers of
ports that exist on the machine. This is normally never an issue ports that exist on the machine. This is normally never an issue
for an end-user machine. It can become an issue for servers that for an end-user machine. It can become an issue for servers that
handle large number of simultaneous streams. However, if the handle large number of simultaneous streams. However, if the
application uses ICE to authenticate STUN requests, a server can application uses ICE to authenticate STUN requests, a server can
serve multiple endpoints from the same local port, and use the serve multiple endpoints from the same local port, and use the
whole 5-tuple (source and destination address, source and whole 5-tuple (source and destination address, source and
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internal endpoints, available external ports are likely the scarce internal endpoints, available external ports are likely the scarce
resource. Port limitations is primarily a problem for larger resource. Port limitations is primarily a problem for larger
centralised NATs where endpoint independent mapping requires each centralised NATs where endpoint independent mapping requires each
flow to use one port for the external IP address. This affects flow to use one port for the external IP address. This affects
the maximum number of internal users per external IP address. the maximum number of internal users per external IP address.
However, it is worth pointing out that a real-time video However, it is worth pointing out that a real-time video
conference session with audio and video is likely using less than conference session with audio and video is likely using less than
10 UDP flows, compared to certain web applications that can use 10 UDP flows, compared to certain web applications that can use
100+ TCP flows to various servers from a single browser instance. 100+ TCP flows to various servers from a single browser instance.
NAT Traversal Excess Time: Making the NAT/FW traversal takes a NAT Traversal Excess Time: Performing the NAT/FW traversal takes a
certain amount of time for each flow. It also takes time in a certain amount of time for each flow. It also takes time in a
phase of communication between accepting to communicate and the phase of communication between accepting to communicate and the
media path being established which is fairly critical. The best media path being established which is fairly critical. The best
case scenario for how much extra time it takes after finding the case scenario for how much extra time it takes after finding the
first valid candidate pair following the specified ICE procedures first valid candidate pair following the specified ICE procedures
are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing
timer, which ICE specifies to be no smaller than 20 ms. That timer, which ICE specifies to be no smaller than 20 ms. That
assumes a message in one direction, and then an immediate assumes a message in one direction, and then an immediate
triggered check back. The reason it isn't more, is that ICE first triggered check back. The reason it isn't more, is that ICE first
finds one candidate pair that works prior to attempting to finds one candidate pair that works prior to attempting to
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delays, at least 100 ms, which is the minimal retransmission timer delays, at least 100 ms, which is the minimal retransmission timer
for ICE. for ICE.
NAT Traversal Failure Rate: Due to the need to establish more than a NAT Traversal Failure Rate: Due to the need to establish more than a
single flow through the NAT, there is some risk that establishing single flow through the NAT, there is some risk that establishing
the first flow succeeds but that one or more of the additional the first flow succeeds but that one or more of the additional
flows fail. The risk that this happens is hard to quantify, but flows fail. The risk that this happens is hard to quantify, but
ought to be fairly low as one flow from the same interfaces has ought to be fairly low as one flow from the same interfaces has
just been successfully established. Thus only rare events such as just been successfully established. Thus only rare events such as
NAT resource overload, or selecting particular port numbers that NAT resource overload, or selecting particular port numbers that
are filtered etc, ought to be reasons for failure. are filtered etc., ought to be reasons for failure.
Deep Packet Inspection and Multiple Streams: Firewalls differ in how Deep Packet Inspection and Multiple Streams: Firewalls differ in how
deeply they inspect packets. There exist some potential that deeply they inspect packets. There exist some potential that
deeply inspecting firewalls will have similar legacy issues with deeply inspecting firewalls will have similar legacy issues with
multiple SSRCs as some stack implementations. multiple SSRCs as some stack implementations.
Additional SSRC keeps the additional media streams within one RTP Additional SSRC keeps the additional media streams within one RTP
Session and transport flow and does not introduce any additional NAT Session and transport flow and does not introduce any additional NAT
traversal complexities per media stream. This can be compared with traversal complexities per media stream. This can be compared with
normally one or two additional transport flows per RTP session when normally one or two additional transport flows per RTP session when
using multiple RTP sessions. Additional lower layer transport flows using multiple RTP sessions. Additional lower layer transport flows
will be needed, unless an explicit de-multiplexing layer is added will be needed, unless an explicit de-multiplexing layer is added
between RTP and the transport protocol. A proposal for how to between RTP and the transport protocol. At time of writing no such
multiplex multiple RTP sessions over the same single lower layer mechanism was defined.
transport exist in [I-D.westerlund-avtcore-transport-multiplexing].
6.3.3. Multicast 4.2.3. Multicast
Multicast groups provides a powerful semantics for a number of real- Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like time applications, especially the ones that desire broadcast-like
behaviours with one endpoint transmitting to a large number of behaviours with one endpoint transmitting to a large number of
receivers, like in IPTV. But that same semantics do result in a receivers, like in IPTV. But that same semantics do result in a
certain number of limitations. certain number of limitations.
One limitation is that for any group, sender side adaptation to the One limitation is that for any group, sender side adaptation to the
actual receiver properties causes degradation for all participants to actual receiver properties causes degradation for all participants to
what is supported by the receiver with the worst conditions among the what is supported by the receiver with the worst conditions among the
group participants. In most cases this is not acceptable. Instead group participants. In most cases this is not acceptable. Instead
various receiver based solutions are employed to ensure that the various receiver based solutions are employed to ensure that the
receivers achieve best possible performance. By using scalable receivers achieve best possible performance. By using scalable
encoding and placing each scalability layer in a different multicast encoding and placing each scalability layer in a different multicast
group, the receiver can control the amount of traffic it receives. group, the receiver can control the amount of traffic it receives.
To have each scalability layer on a different multicast group, one To have each scalability layer on a different multicast group, one
RTP session per multicast group is used. RTP session per multicast group is used.
In addition, the transport flow considerations in multicast are a bit In addition, the transport flow considerations in multicast are a bit
different from unicast; NATs are not useful in the multicast different from unicast; NATs with port translation are not useful in
environment, meaning that the entire port range of each multicast the multicast environment, meaning that the entire port range of each
address is available for distinguishing between RTP sessions. multicast address is available for distinguishing between RTP
sessions.
Thus it appears easiest and most straightforward to use multiple RTP Thus it appears easiest and most straightforward to use multiple RTP
sessions for sending different media flows used for adapting to sessions for sending different media flows used for adapting to
network conditions. network conditions. It is also common that streams that improve
transport robustness are sent in their own multicast group to allow
for interworking with legacy or to support different levels of
protection.
6.3.4. Multiplexing multiple RTP Session on a Single Transport Here are some common behaviours for RTP multicast:
For applications that don't need flow based QoS and like to save 1. Multicast applications use a group of RTP sessions, not one.
ports and NAT/FW traversal costs and where usage of multiple media Each endpoint will need to be a member of a number of RTP
types in one RTP session is not suitable, there is a proposal for how sessions in order to perform well.
to achieve multiplexing of multiple RTP sessions over the same lower
layer transport [I-D.westerlund-avtcore-transport-multiplexing].
Using such a solution would allow Multiple RTP session without most
of the perceived downsides of Multiple RTP sessions creating a need
for additional transport flows, but this solution would require
support from all functions that handle RTP packets, including
firewalls.
6.4. Security and Key Management Considerations 2. Within each RTP session, the number of RTP Sinks is likely to be
much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement; all of
the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It is to be
noted that SDP offer/answer [RFC3264] is not appropriate for ensuring
this property. The signalling aspects of multicast are not explored
further in this memo.
Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol needs to support group communication. Source authentication
requires special solutions. For more discussion on this please
review Options for Securing RTP Sessions [RFC7201].
4.3. Security and Key Management Considerations
When dealing with point-to-point, 2-member RTP sessions only, there When dealing with point-to-point, 2-member RTP sessions only, there
are few security issues that are relevant to the choice of having one are few security issues that are relevant to the choice of having one
RTP session or multiple RTP sessions. However, there are a few RTP session or multiple RTP sessions. However, there are a few
aspects of multiparty sessions that might warrant consideration. For aspects of multiparty sessions that might warrant consideration. For
general information of possible methods of securing RTP, please general information of possible methods of securing RTP, please
review RTP Security Options [I-D.ietf-avtcore-rtp-security-options]. review RTP Security Options [RFC7201].
6.4.1. Security Context Scope 4.3.1. Security Context Scope
When using SRTP [RFC3711] the security context scope is important and When using SRTP [RFC3711] the security context scope is important and
can be a necessary differentiation in some applications. As SRTP's can be a necessary differentiation in some applications. As SRTP's
crypto suites (so far) are built around symmetric keys, the receiver crypto suites (so far) are built around symmetric keys, the receiver
will need to have the same key as the sender. This results in that will need to have the same key as the sender. This results in that
no one in a multi-party session can be certain that a received packet no one in a multi-party session can be certain that a received packet
really was sent by the claimed sender or by another party having really was sent by the claimed sender or by another party having
access to the key. In most cases this is a sufficient security access to the key. In most cases this is a sufficient security
property, but there are a few cases where this does create issues. property, but there are a few cases where this does create issues.
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SRTP [RFC3711] has no special functions for dealing with different SRTP [RFC3711] has no special functions for dealing with different
sets of master keys for different SSRCs. The key-management sets of master keys for different SSRCs. The key-management
functions have different capabilities to establish different set of functions have different capabilities to establish different set of
keys, normally on a per endpoint basis. For example, DTLS-SRTP keys, normally on a per endpoint basis. For example, DTLS-SRTP
[RFC5764] and Security Descriptions [RFC4568] establish different [RFC5764] and Security Descriptions [RFC4568] establish different
keys for outgoing and incoming traffic from an endpoint. This key keys for outgoing and incoming traffic from an endpoint. This key
usage has to be written into the cryptographic context, possibly usage has to be written into the cryptographic context, possibly
associated with different SSRCs. associated with different SSRCs.
6.4.2. Key Management for Multi-party session 4.3.2. Key Management for Multi-party session
Performing key-management for multi-party session can be a challenge. Performing key-management for multi-party session can be a challenge.
This section considers some of the issues. This section considers some of the issues.
Multi-party sessions, such as transport translator based sessions and Multi-party sessions, such as transport translator based sessions and
multicast sessions, cannot use Security Description [RFC4568] nor multicast sessions, cannot use Security Description [RFC4568] nor
DTLS-SRTP [RFC5764] without an extension as each endpoint provides DTLS-SRTP [RFC5764] without an extension as each endpoint provides
its set of keys. In centralised conferences, the signalling its set of keys. In centralised conferences, the signalling
counterpart is a conference server and the media plane unicast counterpart is a conference server and the media plane unicast
counterpart (to which DTLS messages would be sent) is the transport counterpart (to which DTLS messages would be sent) is the transport
translator. Thus an extension like Encrypted Key Transport translator. Thus an extension like Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution [I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution
that allows for keying all session participants with the same master that allows for keying all session participants with the same master
key. key.
6.4.3. Complexity Implications 4.3.3. Complexity Implications
The usage of security functions can surface complexity implications The usage of security functions can surface complexity implications
of the choice of multiplexing and topology. This becomes especially of the choice of multiplexing and topology. This becomes especially
evident in RTP topologies having any type of middlebox that processes evident in RTP topologies having any type of middlebox that processes
or modifies RTP/RTCP packets. Where there is very small overhead for or modifies RTP/RTCP packets. Where there is very small overhead for
an RTP translator or mixer to rewrite an SSRC value in the RTP packet an RTP translator or mixer to rewrite an SSRC value in the RTP packet
of an unencrypted session, the cost of doing it when using of an unencrypted session, the cost of doing it when using
cryptographic security functions is higher. For example if using cryptographic security functions is higher. For example if using
SRTP [RFC3711], the actual security context and exact crypto key are SRTP [RFC3711], the actual security context and exact crypto key are
determined by the SSRC field value. If one changes it, the determined by the SSRC field value. If one changes it, the
encryption and authentication tag needs to be performed using another encryption and authentication tag needs to be performed using another
key. Thus changing the SSRC value implies a decryption using the old key. Thus changing the SSRC value implies a decryption using the old
SSRC and its security context followed by an encryption using the new SSRC and its security context followed by an encryption using the new
one. one.
7. Archetypes 5. Archetypes
This section discusses some archetypes of how RTP multiplexing can be This section discusses some archetypes of how RTP multiplexing can be
used in applications to achieve certain goals and a summary of their used in applications to achieve certain goals and a summary of their
implications. For each archetype there is discussion of benefits and implications. For each archetype there is discussion of benefits and
downsides. downsides.
7.1. Single SSRC per Session 5.1. Single SSRC per Session
In this archetype each endpoint in a point-to-point session has only In this archetype each endpoint in a point-to-point session has only
a single SSRC, thus the RTP session contains only two SSRCs, one a single SSRC, thus the RTP session contains only two SSRCs, one
local and one remote. This session can be used both unidirectional, local and one remote. This session can be used both unidirectional,
i.e. only a single media stream or bi-directional, i.e. both i.e. only a single media stream or bi-directional, i.e. both
endpoints have one media stream each. If the application needs endpoints have one media stream each. If the application needs
additional media flows between the endpoints, they will have to additional media flows between the endpoints, they will have to
establish additional RTP sessions. establish additional RTP sessions.
The Pros: The Pros:
1. This archetype has great legacy interoperability potential as it 1. This archetype has great legacy interoperability potential as it
will not tax any RTP stack implementations. will not tax any RTP stack implementations.
2. The signalling has good possibilities to negotiate and describe 2. The signalling has good possibilities to negotiate and describe
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b. When the number of RTP sessions grows, the amount of explicit b. When the number of RTP sessions grows, the amount of explicit
state for relating media stream also grows, linearly or possibly state for relating media stream also grows, linearly or possibly
exponentially, depending on how the application needs to relate exponentially, depending on how the application needs to relate
media streams. media streams.
c. The port consumption might become a problem for centralised c. The port consumption might become a problem for centralised
services, where the central node's port consumption grows rapidly services, where the central node's port consumption grows rapidly
with the number of sessions. with the number of sessions.
d. For applications where the media streams are highly dynamic in d. For applications where the media streams are highly dynamic in
their usage, i.e. entering and leaving, the amount of signalling their usage, i.e. entering and leaving, the amount of signalling
can grow high. Issues arising from the timely establishment of can grow high. Issues arising from the timely establishment of
additional RTP sessions can also arise. additional RTP sessions can also arise.
e. Cross session RTCP requests might be needed, and the fact that e. Cross session RTCP requests might be needed, and the fact that
they're impossible can cause issues. they're impossible can cause issues.
f. If the same SSRC value is reused in multiple RTP sessions rather f. If the same SSRC value is reused in multiple RTP sessions rather
than being randomly chosen, interworking with applications that than being randomly chosen, interworking with applications that
uses another multiplexing structure than this application will uses another multiplexing structure than this application will
require SSRC translation. require SSRC translation.
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large number of media descriptions in SDP can also run into issues large number of media descriptions in SDP can also run into issues
with existing implementations. For any application needing a larger with existing implementations. For any application needing a larger
number of media flows, the overhead can become very significant. number of media flows, the overhead can become very significant.
This structure is also not suitable for multi-party sessions, as any This structure is also not suitable for multi-party sessions, as any
given media stream from each participant, although having same usage given media stream from each participant, although having same usage
in the application, needs its own RTP session. In addition, the in the application, needs its own RTP session. In addition, the
dynamic behaviour that can arise in multi-party applications can tax dynamic behaviour that can arise in multi-party applications can tax
the signalling system and make timely media establishment more the signalling system and make timely media establishment more
difficult. difficult.
7.2. Multiple SSRCs of the Same Media Type 5.2. Multiple SSRCs of the Same Media Type
In this archetype, each RTP session serves only a single media type. In this archetype, each RTP session serves only a single media type.
The RTP session can contain multiple media streams, either from a The RTP session can contain multiple media streams, either from a
single endpoint or from multiple endpoints. This commonly creates a single endpoint or from multiple endpoints. This commonly creates a
low number of RTP sessions, typically only one for audio and one for low number of RTP sessions, typically only one for audio and one for
video, with a corresponding need for two listening ports when using video, with a corresponding need for two listening ports when using
RTP/RTCP multiplexing. RTP/RTCP multiplexing.
The Pros: The Pros:
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2. Allows for early de-multiplexing in the processing chain in RTP 2. Allows for early de-multiplexing in the processing chain in RTP
applications where all media streams of the same type have the applications where all media streams of the same type have the
same usage in the application. same usage in the application.
3. Works well with media type de-composite endpoints. 3. Works well with media type de-composite endpoints.
4. Enables Flow-based QoS with different prioritisation between 4. Enables Flow-based QoS with different prioritisation between
media types. media types.
5. For applications with dynamic usage of media streams, i.e. they 5. For applications with dynamic usage of media streams, i.e. they
come and go frequently, having much of the state associated with come and go frequently, having much of the state associated with
the RTP session rather than an individual SSRC can avoid the need the RTP session rather than an individual SSRC can avoid the need
for in-session signalling of meta-information about each SSRC. for in-session signalling of meta-information about each SSRC.
6. Low overhead for security association establishment. 6. Low overhead for security association establishment.
The Cons: The Cons:
a. May have some need for cross session RTCP requests for things a. May have some need for cross session RTCP requests for things
that affect both media types in an asynchronous way. that affect both media types in an asynchronous way.
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share same usage, this structure provides efficiency gains in amount share same usage, this structure provides efficiency gains in amount
of network state used and provides more fate sharing with other media of network state used and provides more fate sharing with other media
flows of the same type. At the same time, it is still maintaining flows of the same type. At the same time, it is still maintaining
almost all functionalities when it comes to negotiation in the almost all functionalities when it comes to negotiation in the
signalling of the properties for the individual media type and also signalling of the properties for the individual media type and also
enabling flow based QoS prioritisation between media types. It enabling flow based QoS prioritisation between media types. It
handles multi-party session well, independently of multicast or handles multi-party session well, independently of multicast or
centralised transport distribution, as additional sources can centralised transport distribution, as additional sources can
dynamically enter and leave the session. dynamically enter and leave the session.
7.3. Multiple Sessions for one Media type 5.3. Multiple Sessions for one Media type
In this archetype one goes one step further than in the above In this archetype one goes one step further than in the above
(Section 7.2) by using multiple RTP sessions also for a single media (Section 5.2) by using multiple RTP sessions also for a single media
type, but still not as far as having a single SSRC per RTP session. type, but still not as far as having a single SSRC per RTP session.
The main reason for going in this direction is that the RTP The main reason for going in this direction is that the RTP
application needs separation of the media streams due to their usage. application needs separation of the media streams due to their usage.
Some typical reasons for going to this archetype are scalability over Some typical reasons for going to this archetype are scalability over
multicast, simulcast, need for extended QoS prioritisation of media multicast, simulcast, need for extended QoS prioritisation of media
streams due to their usage in the application, or the need for fine- streams due to their usage in the application, or the need for fine-
grained signalling using today's tools. grained signalling using today's tools.
The Pros: The Pros:
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will have difficulties establishing such a session. will have difficulties establishing such a session.
For more complex RTP applications that have several different usages For more complex RTP applications that have several different usages
for media streams of the same media type and / or uses scalability or for media streams of the same media type and / or uses scalability or
simulcast, this solution can enable those functions at the cost of simulcast, this solution can enable those functions at the cost of
increased overhead associated with the additional sessions. This increased overhead associated with the additional sessions. This
type of structure is suitable for more advanced applications as well type of structure is suitable for more advanced applications as well
as multicast based applications requiring differentiation to as multicast based applications requiring differentiation to
different participants. different participants.
7.4. Multiple Media Types in one Session 5.4. Multiple Media Types in one Session
This archetype is to use a single RTP session for multiple different This archetype is to use a single RTP session for multiple different
media types, like audio and video, and possibly also transport media types, like audio and video, and possibly also transport
robustness mechanisms like FEC or Retransmission. Each media stream robustness mechanisms like FEC or Retransmission. Each media stream
will use its own SSRC and a given SSRC value from a particular will use its own SSRC and a given SSRC value from a particular
endpoint will never use the SSRC for more than a single media type. endpoint will never use the SSRC for more than a single media type.
The Pros: The Pros:
1. Single RTP session which implies: 1. Single RTP session which implies:
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g. Additional concern with legacy implementations that do not g. Additional concern with legacy implementations that do not
support the RTP specification fully when it comes to handling support the RTP specification fully when it comes to handling
multiple SSRC per endpoint, as also multiple simultaneous media multiple SSRC per endpoint, as also multiple simultaneous media
types needs to be handled. types needs to be handled.
h. If the applications need finer control over which session h. If the applications need finer control over which session
participants that are included in different sets of security participants that are included in different sets of security
associations, most key-management will have difficulties associations, most key-management will have difficulties
establishing such a session. establishing such a session.
7.5. Summary 5.5. Summary
There are some clear relations between these archetypes. Both the There are some clear relations between these archetypes. Both the
"single SSRC per RTP session" and the "multiple media types in one OCGBPsingle SSRC per RTP sessionOCOe and the OCGBPmultiple media
session" are cases which require full explicit signalling of the types in one sessionOCOe are cases which require full explicit
media stream relations. However, they operate on two different signalling of the media stream relations. However, they operate on
levels where the first primarily enables session level binding, and two different levels where the first primarily enables session level
the second needs to do it all on SSRC level. From another binding, and the second needs to do it all on SSRC level. From
perspective, the two solutions are the two extreme points when it another perspective, the two solutions are the two extreme points
comes to number of RTP sessions needed. when it comes to number of RTP sessions needed.
The two other archetypes "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for one Media Type" are examples of two other
cases that first of all allows for some implicit mapping of the role
or usage of the media streams based on which RTP session they appear
in. It thus potentially allows for less signalling and in particular
reduced need for real-time signalling in dynamic sessions. They also
represent points in between the first two when it comes to amount of
RTP sessions established, i.e. representing an attempt to reduce the
amount of sessions as much as possible without compromising the
functionality the session provides both on network level and on
signalling level.
8. Summary considerations and guidelines The two other archetypes OCGBPMultiple SSRCs of the Same Media
TypeOCOe and OCGBPMultiple Sessions for one Media TypeOCOe are
examples of two other cases that first of all allows for some
implicit mapping of the role or usage of the media streams based on
which RTP session they appear in. It thus potentially allows for
less signalling and in particular reduced need for real-time
signalling in dynamic sessions. They also represent points in
between the first two when it comes to amount of RTP sessions
established, i.e. representing an attempt to reduce the amount of
sessions as much as possible without compromising the functionality
the session provides both on network level and on signalling level.
8.1. Guidelines 6. Summary considerations and guidelines
6.1. Guidelines
This section contains a number of recommendations for implementors or This section contains a number of recommendations for implementers or
specification writers when it comes to handling multi-stream. specification writers when it comes to handling multi-stream.
Do not Require the same SSRC across Sessions: As discussed in Do not Require the same SSRC across Sessions: As discussed in
Section 6.1.5 there exist drawbacks in using the same SSRC in Section 3.4.3 there exist drawbacks in using the same SSRC in
multiple RTP sessions as a mechanism to bind related media streams multiple RTP sessions as a mechanism to bind related media streams
together. It is instead suggested that a mechanism to explicitly together. It is instead suggested that a mechanism to explicitly
signal the relation is used, either in RTP/RTCP or in the used signal the relation is used, either in RTP/RTCP or in the used
signalling mechanism that establishes the RTP session(s). signalling mechanism that establishes the RTP session(s).
Use additional SSRCs additional Media Sources: In the cases where an Use additional SSRCs for additional Media Sources: In the cases
RTP endpoint needs to transmit additional media streams of the where an RTP endpoint needs to transmit additional media streams
same media type in the application, with the same processing of the same media type in the application, with the same
requirements at the network and RTP layers, it is suggested to processing requirements at the network and RTP layers, it is
send them as additional SSRCs in the same RTP session. For suggested to send them as additional SSRCs in the same RTP
example a telepresence room where there are three cameras, and session. For example a telepresence room where there are three
each camera captures 2 persons sitting at the table, sending each cameras, and each camera captures 2 persons sitting at the table,
camera as its own SSRC within a single RTP session is suggested. sending each camera as its own SSRC within a single RTP session is
suggested.
Use additional RTP sessions for streams with different requirements: Use additional RTP sessions for streams with different requirements:
When media streams have different processing requirements from the When media streams have different processing requirements from the
network or the RTP layer at the endpoints, it is suggested that network or the RTP layer at the endpoints, it is suggested that
the different types of streams are put in different RTP sessions. the different types of streams are put in different RTP sessions.
This includes the case where different participants want different This includes the case where different participants want different
subsets of the set of RTP streams. subsets of the set of RTP streams.
When using multiple RTP Sessions use grouping: When using Multiple When using multiple RTP Sessions use grouping: When using Multiple
RTP session solutions, it is suggested to explicitly group the RTP session solutions, it is suggested to explicitly group the
involved RTP sessions when needed using the signalling mechanism, involved RTP sessions when needed using the signalling mechanism,
for example The Session Description Protocol (SDP) Grouping for example The Session Description Protocol (SDP) Grouping
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the different types of streams are put in different RTP sessions. the different types of streams are put in different RTP sessions.
This includes the case where different participants want different This includes the case where different participants want different
subsets of the set of RTP streams. subsets of the set of RTP streams.
When using multiple RTP Sessions use grouping: When using Multiple When using multiple RTP Sessions use grouping: When using Multiple
RTP session solutions, it is suggested to explicitly group the RTP session solutions, it is suggested to explicitly group the
involved RTP sessions when needed using the signalling mechanism, involved RTP sessions when needed using the signalling mechanism,
for example The Session Description Protocol (SDP) Grouping for example The Session Description Protocol (SDP) Grouping
Framework. [RFC5888], using some appropriate grouping semantics. Framework. [RFC5888], using some appropriate grouping semantics.
RTP/RTCP Extensions May Support Additional SSRCs as well as Multiple RTP sessions: RTP/RTCP Extensions May Support Additional SSRCs as well as Multiple
RTP sessions:
When defining an RTP or RTCP extension, the creator needs to When defining an RTP or RTCP extension, the creator needs to
consider if this extension is applicable to usage with additional consider if this extension is applicable to usage with additional
SSRCs and Multiple RTP sessions. Any extension intended to be SSRCs and Multiple RTP sessions. Any extension intended to be
generic is suggested to support both. Applications that are not generic is suggested to support both. Applications that are not
as generally applicable will have to consider if interoperability as generally applicable will have to consider if interoperability
is better served by defining a single solution or providing both is better served by defining a single solution or providing both
options. options.
Transport Support Extensions: When defining new RTP/RTCP extensions Transport Support Extensions: When defining new RTP/RTCP extensions
intended for transport support, like the retransmission or FEC intended for transport support, like the retransmission or FEC
mechanisms, they are expected to include support for both mechanisms, they are expected to include support for both
additional SSRCs and multiple RTP sessions so that application additional SSRCs and multiple RTP sessions so that application
developers can choose freely from the set of mechanisms without developers can choose freely from the set of mechanisms without
concerning themselves with which of the multiplexing choices a concerning themselves with which of the multiplexing choices a
particular solution supports. particular solution supports.
9. IANA Considerations 7. Open Issues
There are currently some issues that needs to be resolved before this
document is ready to be published:
1. Use of RFC 2119 language is section on SSRC (3.2.2)
2. Better align source and sink terminolgy with Taxonomy
(Section 3.2.2)
3. Section on Binding Related Sources (Section 3.4.3) needs more
text on usage of the RID and other SDES based mechanisms created.
4. Does the MSID text need to be updated and clarified based on the
evoulsion of MSID since previous version. Section 3.4.3.
5. Section 4.1.2 (RTP Translator Interworking) needs to be updated.
It is not obvious that it is a natural requirement that the same
multiplexing is used. This needs better discussion.
6. Refernce to Ta for ICE being 20 ms will need to be updated due to
ICE update.
7. In Section 4.3.2 (Key Management for Multi-party session) the
reference to EKT needs to be updated, question is if draft-ietf-
perc-ekt-diet is appropriate here?
8. Can we find a more approriate term than archetypes?
9.
8. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section can be removed on publication as an Note to RFC Editor: this section can be removed on publication as an
RFC. RFC.
10. Security Considerations 9. Security Considerations
There is discussion of the security implications of choosing SSRC vs There is discussion of the security implications of choosing SSRC vs
Multiple RTP session in Section 6.4. Multiple RTP session in Section 4.3.
11. References 10. References
11.1. Normative References 10.1. Normative References
[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, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
11.2. Informative References [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>.
10.2. Informative References
[ALF] Clark, D. and D. Tennenhouse, "Architectural [ALF] Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols", SIGCOMM Considerations for a New Generation of Protocols", SIGCOMM
Symposium on Communications Architectures and Protocols Symposium on Communications Architectures and
(Philadelphia, Pennsylvania), pp. 200--208, IEEE Computer Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
Communications Review, Vol. 20(4), September 1990. Computer Communications Review, Vol. 20(4), September
1990.
[I-D.alvestrand-rtp-sess-neutral]
Alvestrand, H., "Why RTP Sessions Should Be Content
Neutral", draft-alvestrand-rtp-sess-neutral-01 (work in
progress), June 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-6222bis]
Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
(work in progress), July 2013.
[I-D.ietf-avtcore-multi-media-rtp-session] [I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft- Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-05 (work in ietf-avtcore-multi-media-rtp-session-13 (work in
progress), February 2014. progress), December 2015.
[I-D.ietf-avtcore-rtp-security-options]
Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", draft-ietf-avtcore-rtp-security-options-10
(work in progress), January 2014.
[I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", draft-ietf-avtext-
multiple-clock-rates-11 (work in progress), November 2013.
[I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-07 (work in progress), September 2014.
[I-D.ietf-mmusic-msid] [I-D.ietf-mmusic-msid]
Alvestrand, H., "WebRTC MediaStream Identification in the Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", draft-ietf-mmusic-msid-06 Session Description Protocol", draft-ietf-mmusic-msid-16
(work in progress), June 2014. (work in progress), February 2017.
[I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
Roach, A., and B. Campen, "RTP Payload Format
Restrictions", draft-ietf-mmusic-rid-11 (work in
progress), July 2017.
[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,
"Negotiating Media Multiplexing Using the Session "Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-11 (work in progress), September 2014. negotiation-39 (work in progress), August 2017.
[I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-13 (work in progress),
January 2014.
[I-D.lennox-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "RTP
Considerations for Endpoints Sending Multiple Media
Streams", draft-lennox-avtcore-rtp-multi-stream-02 (work
in progress), February 2013.
[I-D.lennox-mmusic-sdp-source-selection] [I-D.lennox-mmusic-sdp-source-selection]
Lennox, J. and H. Schulzrinne, "Mechanisms for Media Lennox, J. and H. Schulzrinne, "Mechanisms for Media
Source Selection in the Session Description Protocol Source Selection in the Session Description Protocol
(SDP)", draft-lennox-mmusic-sdp-source-selection-05 (work (SDP)", draft-lennox-mmusic-sdp-source-selection-05 (work
in progress), October 2012. in progress), October 2012.
[I-D.westerlund-avtcore-max-ssrc]
Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling",
draft-westerlund-avtcore-max-ssrc-02 (work in progress),
July 2012.
[I-D.westerlund-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
westerlund-avtcore-rtp-topologies-update-02 (work in
progress), February 2013.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
Sessions onto a Single Lower-Layer Transport", draft-
westerlund-avtcore-transport-multiplexing-07 (work in
progress), October 2013.
[I-D.westerlund-avtext-rtcp-sdes-srcname]
Westerlund, M., "RTCP Source Description Item SRCNAME to
Label Individual Media Sources", draft-westerlund-avtext-
rtcp-sdes-srcname-03 (work in progress), October 2013.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997. DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S., and S. [RFC2205] Braden, R., Ed., Zhang, L., Berson, S., Herzog, S., and S.
Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1 Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
Functional Specification", RFC 2205, September 1997. Functional Specification", RFC 2205, DOI 10.17487/RFC2205,
September 1997, <https://www.rfc-editor.org/info/rfc2205>.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2474] Nichols, K., Blake, S., Baker, F., and D.L. Black, [RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS "Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474, December Field) in the IPv4 and IPv6 Headers", RFC 2474,
1998. DOI 10.17487/RFC2474, December 1998,
<https://www.rfc-editor.org/info/rfc2474>.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session [RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000. Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
October 2000, <https://www.rfc-editor.org/info/rfc2974>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261, Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002. DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June with Session Description Protocol (SDP)", RFC 3264,
2002. DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002. Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://www.rfc-editor.org/info/rfc3389>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004. DOI 10.17487/RFC3830, August 2004,
<https://www.rfc-editor.org/info/rfc3830>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005. Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006. Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<https://www.rfc-editor.org/info/rfc4568>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
January 2008. Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)", RFC
5583, July 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010. Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010. Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<https://www.rfc-editor.org/info/rfc5888>.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A. [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
Eleftheriadis, "RTP Payload Format for Scalable Video "RTP Payload Format for Scalable Video Coding", RFC 6190,
Coding", RFC 6190, May 2011. DOI 10.17487/RFC6190, May 2011,
<https://www.rfc-editor.org/info/rfc6190>.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for [RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
Choosing RTP Control Protocol (RTCP) Canonical Names time Transport Protocol (RTP) Header Extension for Mixer-
(CNAMEs)", RFC 6222, April 2011. to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<https://www.rfc-editor.org/info/rfc6465>.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax, [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
"Unicast-Based Rapid Acquisition of Multicast RTP Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
Sessions", RFC 6285, June 2011. <https://www.rfc-editor.org/info/rfc7201>.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
Transport Protocol (RTP) Header Extension for Mixer-to- (Diffserv) and Real-Time Communication", RFC 7657,
Client Audio Level Indication", RFC 6465, December 2011. DOI 10.17487/RFC7657, November 2015,
<https://www.rfc-editor.org/info/rfc7657>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC7826] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, Ed., "Real-Time Streaming Protocol
Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December
2016, <https://www.rfc-editor.org/info/rfc7826>.
[RFC8088] Westerlund, M., "How to Write an RTP Payload Format",
RFC 8088, DOI 10.17487/RFC8088, May 2017,
<https://www.rfc-editor.org/info/rfc8088>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
Appendix A. Dismissing Payload Type Multiplexing Appendix A. Dismissing Payload Type Multiplexing
This section documents a number of reasons why using the payload type This section documents a number of reasons why using the payload type
as a multiplexing point for most things related to multiple streams as a multiplexing point for most things related to multiple streams
is unsuitable. If one attempts to use Payload type multiplexing is unsuitable. If one attempts to use Payload type multiplexing
beyond it's defined usage, that has well known negative effects on beyond it's defined usage, that has well known negative effects on
RTP. To use Payload type as the single discriminator for multiple RTP. To use Payload type as the single discriminator for multiple
streams implies that all the different media streams are being sent streams implies that all the different media streams are being sent
with the same SSRC, thus using the same timestamp and sequence number with the same SSRC, thus using the same timestamp and sequence number
skipping to change at page 43, line 31 skipping to change at page 39, line 49
is currently possible for RTP Sessions and for SSRC, but there is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types. is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements 10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing. per media stream when using Payload Type Multiplexing.
11. Most existing SDP media level attributes cannot be applied on a 11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that per Payload Type level and would require re-definition in that
context. context.
12. A legacy endpoint that doesn't understand the indication that 12. A legacy endpoint that does not understand the indication that
different RTP payload types are different media streams might be different RTP payload types are different media streams might be
slightly confused by the large amount of possibly overlapping or slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types. identically defined RTP Payload Types.
Appendix B. Proposals for Future Work Appendix B. Signalling considerations
The above discussion and guidelines indicates that a small set of
extension mechanisms could greatly improve the situation when it
comes to using multiple streams independently of Multiple RTP session
or Additional SSRC. These extensions are:
Media Source Identification: A Media source identification that can
be used to bind together media streams that are related to the
same media source. A proposal
[I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
item SRCNAME that also can be used with the a=ssrc SDP attribute
to provide signalling layer binding information.
MSID: A Media Stream identification scheme that can be used to
signal relationships between SSRCs that can be in the same or in
different RTP sessions. Described in [I-D.ietf-mmusic-msid]
SSRC limitations within RTP sessions: By providing a signalling
solution that allows the signalling peers to explicitly express
both support and limitations on how many simultaneous media
streams an endpoint can handle within a given RTP Session. That
ensures that usage of Additional SSRC occurs when supported and
without overloading an endpoint. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc].
Appendix C. Signalling considerations
Signalling is not an architectural consideration for RTP itself, so Signalling is not an architectural consideration for RTP itself, so
this discussion has been moved to an appendix. However, it is hugely this discussion has been moved to an appendix. However, it is hugely
important for anyone building complete applications, so it is important for anyone building complete applications, so it is
deserving of discussion. deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with The issues raised here need to be addressed in the WGs that deal with
signalling; they cannot be addressed by tweaking, extending or signalling; they cannot be addressed by tweaking, extending or
profiling RTP. profiling RTP.
C.1. Signalling Aspects B.1. Signalling Aspects
There exist various signalling solutions for establishing RTP There exist various signalling solutions for establishing RTP
sessions. Many are SDP [RFC4566] based, however SDP functionality is sessions. Many are SDP [RFC4566] based, however SDP functionality is
also dependent on the signalling protocols carrying the SDP. Where also dependent on the signalling protocols carrying the SDP. Where
RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative RTSP [RFC7826] and SAP [RFC2974] both use SDP in a declarative
fashion, while SIP [RFC3261] uses SDP with the additional definition fashion, while SIP [RFC3261] uses SDP with the additional definition
of Offer/Answer [RFC3264]. The impact on signalling and especially of Offer/Answer [RFC3264]. The impact on signalling and especially
SDP needs to be considered as it can greatly affect how to deploy a SDP needs to be considered as it can greatly affect how to deploy a
certain multiplexing point choice. certain multiplexing point choice.
C.1.1. Session Oriented Properties B.1.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused around One aspect of the existing signalling is that it is focused around
sessions, or at least in the case of SDP the media description. sessions, or at least in the case of SDP the media description.
There are a number of things that are signalled on a session level/ There are a number of things that are signalled on a session level/
media description but those are not necessarily strictly bound to an media description but those are not necessarily strictly bound to an
RTP session and could be of interest to signal specifically for a RTP session and could be of interest to signal specifically for a
particular media stream (SSRC) within the session. The following particular media stream (SSRC) within the session. The following
properties have been identified as being potentially useful to signal properties have been identified as being potentially useful to signal
not only on RTP session level: not only on RTP session level:
skipping to change at page 45, line 13 skipping to change at page 41, line 4
limiting or RTCP signalling using TMMBR is used. limiting or RTCP signalling using TMMBR is used.
o Which SSRC that will use which RTP Payload Types (this will be o Which SSRC that will use which RTP Payload Types (this will be
visible from the first media packet, but is sometimes useful to visible from the first media packet, but is sometimes useful to
know before packet arrival). know before packet arrival).
Some of these issues are clearly SDP's problem rather than RTP Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy an solution using limitations. However, if the aim is to deploy an solution using
additional SSRCs that contains several sets of media streams with additional SSRCs that contains several sets of media streams with
different properties (encoding/packetization parameter, bit-rate, different properties (encoding/packetization parameter, bit-rate,
etc), putting each set in a different RTP session would directly etc.), putting each set in a different RTP session would directly
enable negotiation of the parameters for each set. If insisting on enable negotiation of the parameters for each set. If insisting on
additional SSRC only, a number of signalling extensions are needed to additional SSRC only, a number of signalling extensions are needed to
clarify that there are multiple sets of media streams with different clarify that there are multiple sets of media streams with different
properties and that they need in fact be kept different, since a properties and that they need in fact be kept different, since a
single set will not satisfy the application's requirements. single set will not satisfy the application's requirements.
For some parameters, such as resolution and framerate, a SSRC-linked For some parameters, such as resolution and framerate, a SSRC-linked
mechanism has been proposed: mechanism has been proposed:
[I-D.lennox-mmusic-sdp-source-selection]. [I-D.lennox-mmusic-sdp-source-selection].
C.1.2. SDP Prevents Multiple Media Types B.1.2. SDP Prevents Multiple Media Types
SDP chose to use the m= line both to delineate an RTP session and to SDP chose to use the m= line both to delineate an RTP session and to
specify the top level of the MIME media type; audio, video, text, specify the top level of the MIME media type; audio, video, text,
image, application. This media type is used as the top-level media image, application. This media type is used as the top-level media
type for identifying the actual payload format bound to a particular type for identifying the actual payload format bound to a particular
payload type using the rtpmap attribute. This binding has to be payload type using the rtpmap attribute. This binding has to be
loosened in order to use SDP to describe RTP sessions containing loosened in order to use SDP to describe RTP sessions containing
multiple MIME top level types. multiple MIME top level types.
There is an accepted WG item in the MMUSIC WG to define how multiple There is an accepted WG item in the MMUSIC WG to define how multiple
media lines describe a single underlying transport media lines describe a single underlying transport
[I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible [I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible
in SDP to define one RTP session with media types having different in SDP to define one RTP session with media types having different
MIME top level types. MIME top level types.
C.1.3. Signalling Media Stream Usage B.1.3. Signalling Media Stream Usage
Media streams being transported in RTP has some particular usage in Media streams being transported in RTP has some particular usage in
an RTP application. This usage of the media stream is in many an RTP application. This usage of the media stream is in many
applications so far implicitly signalled. For example, an applications so far implicitly signalled. For example, an
application might choose to take all incoming audio RTP streams, mix application might choose to take all incoming audio RTP streams, mix
them and play them out. However, in more advanced applications that them and play them out. However, in more advanced applications that
use multiple media streams there will be more than a single usage or use multiple media streams there will be more than a single usage or
purpose among the set of media streams being sent or received. RTP purpose among the set of media streams being sent or received. RTP
applications will need to signal this usage somehow. The signalling applications will need to signal this usage somehow. The signalling
used will have to identify the media streams affected by their RTP- used will have to identify the media streams affected by their RTP-
skipping to change at page 46, line 25 skipping to change at page 42, line 17
If this signalling affects how any RTP central node, like an RTP If this signalling affects how any RTP central node, like an RTP
mixer or translator that selects, mixes or processes streams, treats mixer or translator that selects, mixes or processes streams, treats
the streams, the node will also need to receive the same signalling the streams, the node will also need to receive the same signalling
to know how to treat media streams with different usage in the right to know how to treat media streams with different usage in the right
fashion. fashion.
Authors' Addresses Authors' Addresses
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Torshamsgatan 23
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
Phone: +46 10 714 82 87 Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com Email: magnus.westerlund@ericsson.com
Bo Burman Bo Burman
Ericsson Ericsson
Farogatan 6 Farogatan 6
SE-164 80 Kista SE-164 80 Kista
skipping to change at page 47, line 4 skipping to change at page 42, line 40
Phone: +46 10 714 13 11 Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com Email: bo.burman@ericsson.com
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
Harald Tveit Alvestrand Harald Tveit Alvestrand
Google Google
Kungsbron 2 Kungsbron 2
Stockholm 11122 Stockholm 11122
Sweden Sweden
Email: harald@alvestrand.no Email: harald@alvestrand.no
Roni Even
Huawei
Email: roni.even@huawei.com
Hui Zheng
Huawei
Email: marvin.zhenghui@huawei.com
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