draft-ietf-avtcore-multiplex-guidelines-05.txt   draft-ietf-avtcore-multiplex-guidelines-06.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: May 3, 2018 C. Perkins Expires: January 3, 2019 C. Perkins
University of Glasgow University of Glasgow
H. Alvestrand H. Alvestrand
Google Google
R. Even R. Even
H. Zheng
Huawei Huawei
October 30, 2017 July 2, 2018
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-05 draft-ietf-avtcore-multiplex-guidelines-06
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
skipping to change at page 1, line 45 skipping to change at page 1, line 44
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This Internet-Draft will expire on May 3, 2018. This Internet-Draft will expire on January 3, 2019.
Copyright Notice Copyright Notice
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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 . . . . . . . . . . . . . . . . . . 5 2.2. Subjects Out of Scope . . . . . . . . . . . . . . . . . . 5
3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5 3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5
3.1. Reasons for Multiplexing and Grouping RTP Media Streams . 5 3.1. Reasons for Multiplexing and Grouping RTP Streams . . . . 5
3.2. RTP Multiplexing Points . . . . . . . . . . . . . . . . . 6 3.2. RTP Multiplexing Points . . . . . . . . . . . . . . . . . 6
3.2.1. RTP Session . . . . . . . . . . . . . . . . . . . . . 7 3.2.1. RTP Session . . . . . . . . . . . . . . . . . . . . . 7
3.2.2. Synchronisation Source (SSRC) . . . . . . . . . . . . 8 3.2.2. Synchronisation Source (SSRC) . . . . . . . . . . . . 8
3.2.3. Contributing Source (CSRC) . . . . . . . . . . . . . 10 3.2.3. Contributing Source (CSRC) . . . . . . . . . . . . . 10
3.2.4. RTP Payload Type . . . . . . . . . . . . . . . . . . 10 3.2.4. RTP Payload Type . . . . . . . . . . . . . . . . . . 10
3.3. Issues Related to RTP Topologies . . . . . . . . . . . . 11 3.3. Issues Related to RTP Topologies . . . . . . . . . . . . 11
3.4. Issues Related to RTP and RTCP Protocol . . . . . . . . . 13 3.4. Issues Related to RTP and RTCP Protocol . . . . . . . . . 12
3.4.1. The RTP Specification . . . . . . . . . . . . . . . . 13 3.4.1. The RTP Specification . . . . . . . . . . . . . . . . 13
3.4.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 15 3.4.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 15
3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 15 3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 15
3.4.4. Forward Error Correction . . . . . . . . . . . . . . 17 3.4.4. Forward Error Correction . . . . . . . . . . . . . . 17
4. Particular Considerations for RTP Multiplexing . . . . . . . 17 4. Considerations for RTP Multiplexing . . . . . . . . . . . . . 17
4.1. Interworking Considerations . . . . . . . . . . . . . . . 17 4.1. Interworking Considerations . . . . . . . . . . . . . . . 17
4.1.1. Types of Interworking . . . . . . . . . . . . . . . . 17 4.1.1. Application Interworking . . . . . . . . . . . . . . 18
4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 18 4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 18
4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 18 4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 19
4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 19 4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 20
4.2. Network Considerations . . . . . . . . . . . . . . . . . 20 4.2. Network Considerations . . . . . . . . . . . . . . . . . 20
4.2.1. Quality of Service . . . . . . . . . . . . . . . . . 20 4.2.1. Quality of Service . . . . . . . . . . . . . . . . . 20
4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 20 4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 21
4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 22 4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 23
4.3. Security and Key Management Considerations . . . . . . . 23 4.3. Security and Key Management Considerations . . . . . . . 24
4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24 4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24
4.3.2. Key Management for Multi-party session . . . . . . . 24 4.3.2. Key Management for Multi-party sessions . . . . . . . 25
4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25 4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25
5. Archetypes . . . . . . . . . . . . . . . . . . . . . . . . . 25 5. RTP Multiplexing Design Choices . . . . . . . . . . . . . . . 26
5.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 25 5.1. Single SSRC per Endpoint . . . . . . . . . . . . . . . . 26
5.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 27 5.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 27
5.3. Multiple Sessions for one Media type . . . . . . . . . . 28 5.3. Multiple Sessions for one Media type . . . . . . . . . . 28
5.4. Multiple Media Types in one Session . . . . . . . . . . . 30 5.4. Multiple Media Types in one Session . . . . . . . . . . . 30
5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 31 5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 31
6. Summary considerations and guidelines . . . . . . . . . . . . 31 6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 31
6.1. Guidelines . . . . . . . . . . . . . . . . . . . . . . . 32
7. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 33 7. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 33
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
9. Security Considerations . . . . . . . . . . . . . . . . . . . 34 9. Security Considerations . . . . . . . . . . . . . . . . . . . 33
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 34 10. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 33
10.1. Normative References . . . . . . . . . . . . . . . . . . 34 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
10.2. Informative References . . . . . . . . . . . . . . . . . 34 11.1. Normative References . . . . . . . . . . . . . . . . . . 33
11.2. Informative References . . . . . . . . . . . . . . . . . 34
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 38 Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 38
Appendix B. Signalling considerations . . . . . . . . . . . . . 40 Appendix B. Signalling Considerations . . . . . . . . . . . . . 40
B.1. Signalling Aspects . . . . . . . . . . . . . . . . . . . 40 B.1. Session Oriented Properties . . . . . . . . . . . . . . . 40
B.1.1. Session Oriented Properties . . . . . . . . . . . . . 40 B.2. SDP Prevents Multiple Media Types . . . . . . . . . . . . 41
B.1.2. SDP Prevents Multiple Media Types . . . . . . . . . . 41 B.3. Signalling RTP stream Usage . . . . . . . . . . . . . . . 41
B.1.3. Signalling Media Stream Usage . . . . . . . . . . . . 41
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42
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 was from the beginning designed for multiple applications. RTP was from the beginning designed for multiple
participants in a communication session. It supports many paradigms participants in a communication session. It supports many topology
of topologies and usages, as defined in [RFC7667]. RTP has several paradigms and usages, as defined in [RFC7667]. RTP has several
multiplexing points designed for different purposes. These enable multiplexing points designed for different purposes. These enable
support of multiple media streams and switching between different support of multiple RTP streams and switching between different
encoding or packetization of the media. By using multiple RTP encoding or packetization of the media. By using multiple RTP
sessions, sets of media streams can be structured for efficient sessions, sets of RTP streams can be structured for efficient
processing or identification. Thus the question for any RTP processing or identification. Thus, the question for any RTP
application designer is how to best use the RTP session, the SSRC and application designer is how to best use the RTP session, the RTP
the payload type to meet the application's needs. stream identifier (SSRC), and the RTP payload type to meet the
application's needs.
There have been increased interest in more advanced usage of RTP, for There have been increased interest in more advanced usage of RTP.
example, multiple streams can occur when a single endpoint have For example, multiple RTP streams can be used when a single endpoint
multiple media sources, like multiple cameras or microphones that has multiple media sources (like multiple cameras or microphones)
need to be sent simultaneously. Consequently, questions are raised that need to be sent simultaneously. Consequently, questions are
regarding the most appropriate RTP usage. The limitations in some raised regarding the most appropriate RTP usage. The limitations in
implementations, RTP/RTCP extensions, and signalling has also been some implementations, RTP/RTCP extensions, and signalling has also
exposed. The authors also hope that clarification on the usefulness been exposed. The authors also hope that clarification on the
of some functionalities in RTP will result in more complete usefulness of some functionalities in RTP will result in more
implementations in the future. 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.
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 designs of RTP usage are discussed.
Finally, some recommendations and examples are provided. Finally, some guidelines and examples are provided.
2. Definitions 2. Definitions
2.1. Terminology 2.1. Terminology
The definitions in Section 3 of [RFC3550] are referenced normatively. The definitions in Section 3 of [RFC3550] are referenced normatively.
The taxonomy defined in [RFC7656] is referenced normatively. The taxonomy defined in [RFC7656] is referenced normatively.
The following terms and abbreviations are used in this document: The following terms and abbreviations are used in this document:
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
than two endpoints communicate. more than two endpoints communicate.
RTP Source: The originator or source of a particular Media Stream. RTP Source: The originator or source of a particular RTP stream sent
Identified using an SSRC in a particular RTP session. An RTP from an endpoint. Identified using an SSRC in a particular RTP
source is the source of a single media stream, and is associated session. An RTP source is the source of a single RTP stream, and
with a single endpoint and a single Media Source. An RTP Source is associated with a single endpoint and a single media source.
is just called a Source in RFC 3550. An RTP Source is just called a Source in RFC 3550. An endpoint
can have multiple RTP sources.
RTP Sink: A recipient of a Media Stream. The Media Sink is RTP Sink: An endpoint that receives RTP streams. The RTP Sink is
identified using one or more SSRCs. There can be more than one identified using one or more SSRCs. The SSRCs used by an RTP sink
RTP Sink for one RTP source. can be both RTP source ones, as well as used soley to represent
the RTP sink. There can be more than one RTP Sink for one RTP
source.
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 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; i.e. SSRCs that map to a common CNAME can be assumed
to have RTCP SR timing information derived from a common clock
such that they can be synchronised for playout.
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 applications that don't send RTCP, they do not conform to the RTP
the RTP specification, and thus can be regarded as reusing the RTP specification, and thus can be regarded as reusing the RTP packet
packet format but not implementing the RTP protocol. format but not implementing the RTP protocol.
3. RTP Multiplexing Overview 3. RTP Multiplexing Overview
3.1. Reasons for Multiplexing and Grouping RTP Media Streams 3.1. Reasons for Multiplexing and Grouping RTP Streams
The reasons why an endpoint might choose to send multiple media There are several reasons why an endpoint might choose to send
streams are widespread. In the below discussion, please keep in mind multiple media streams. 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 RTP streams vary and include but
but are not limited to the following: 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 RTP streams might be needed to represent one media source
Source (for instance when using layered encodings) (for instance when using layered encodings)
o A Retransmission stream might repeat the content of another Media o A retransmission stream might repeat some parts of the content of
Stream another RTP stream
o An FEC stream might provide material that can be used to repair o An FEC stream might provide material that can be used to repair
another Media Stream another RTP stream
o Alternative Encodings, for instance different codecs for the same o Alternative encodings, for instance using different codecs for the
audio stream same 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 reasons, it is necessary to decide if each
stream gets its own SSRC multiplexed within a RTP Session, or if it additional RTP stream is sent within the same RTP session as the
is necessary to use additional RTP sessions to group the media other RTP streams, or if it is necessary to use additional RTP
streams. The choice between these made due to one reason might not sessions to group the RTP streams. The choice suitable for one
be the choice suitable for another reason. The clearest reason, might not be the choice suitable for another reason. The
understanding is associated with multiple media sources of the same clearest understanding is associated with multiplexing multiple media
media type. However, all warrant discussion and clarification on how sources of the same media type. However, all reasons 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
of SSRC or RTP session multiplexing solutions. To utilise RTP well
and as efficiently as possible, both are needed. The real issue is and as efficiently as possible, both are needed. The real issue is
finding the right guidance on when to create RTP sessions and when finding the right guidance on when to create additional RTP sessions
additional SSRCs in an RTP session is the right choice. and when additional RTP streams in the same RTP session is the right
choice.
3.2. 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 RTP streams and groups of
media streams. Figure 1 outlines the process of demultiplexing RTP streams. Figure 1 outlines the process of demultiplexing
incoming RTP streams: incoming RTP streams:
| |
| packets | packets
+-- v +-- v
| +------------+ | +------------+
| | Socket | Transport Protocol Demultiplexing
| | 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)
| || || ||
| || || || | || || ||
RTP | +--+ +--+ +--+
| || || || Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, etc.
| +--+ +--+ +--+
Media | +--+ +--+ +--+ +-- | | |
+-- | / |
Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, FEC, etc. | +----+ |
| / | |
| +--+ +--+ +--+ Payload | +---+ +---+ +---+
Formats | |Dec| |Dec| |Dec| Decoders
+-- | | | | +---+ +---+ +---+
(pick rendering context based on PT)
+-- | / |
| +---+ |
| / | |
Payload | +--+ +--+ +--+
Formats | |CR| |CR| |CR| Codecs and rendering
| +--+ +--+ +--+
+-- +--
Figure 1: RTP Demultiplexing Process Figure 1: RTP Demultiplexing Process
3.2.1. 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 RTP does not contain a session identifier, yet RTP sessions must be
that share a single synchronisation source space [RFC3550]. That is, possible to separate both across different endpoints and within a
if a group of participants are each aware of the synchronisation single endpoint.
source identifiers belonging to the other participants, then those
participants are in a single RTP session. A participant can become
aware of a synchronisation source identifier by receiving an RTP
packet containing it in the SSRC field or CSRC list, by receiving an
RTCP packet mentioning it in an SSRC field, or through signalling
(e.g., the SDP "a=ssrc:" attribute). Thus, the scope of an RTP
session is determined by the participants' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by the endpoints and any middleboxes, and by the signalling.
RTP does not contain a session identifier. Rather, it relies on the For RTP session separation across endpoints, the set of participants
underlying transport layer to separate different sessions, and on the that form an RTP session is defined as those that share a single
signalling to identify sessions in a manner that is meaningful to the synchronisation source space [RFC3550]. That is, if a group of
application. The signalling layer might give sessions an explicit participants are each aware of the synchronisation source identifiers
identifier, or their identification might be implicit based on the belonging to the other participants, then those participants are in a
addresses and ports used. Accordingly, a single RTP Session can have single RTP session. A participant can become aware of a
synchronisation source identifier by receiving an RTP packet
containing it in the SSRC field or CSRC list, by receiving an RTCP
packet mentioning it in an SSRC field, or through signalling (e.g.,
the Session Description Protocol (SDP) [RFC4566] "a=ssrc:" attribute
[RFC5576]). Thus, the scope of an RTP session is determined by the
participants' network interconnection topology, in combination with
RTP and RTCP forwarding strategies deployed by the endpoints and any
middleboxes, and by the signalling.
For RTP session separation within a single endpoint, RTP relies on
the underlying transport layer, and on the signalling to identify RTP
sessions in a manner that is meaningful to the application. A single
endpoint can have one or more transport flows for the same RTP
session. The signalling layer might give RTP sessions an explicit
identifier, or the identification might be implicit based on the
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 by receiving the multiple UDP flows used as identified
UDP port numbers. Another example is when using SDP grouping based on their UDP source and destination IP addresses and UDP port
framework [RFC5888] which uses an identifier per "m="-line; if there numbers. Another example is SDP media descriptions (the "m=" line
is a one-to-one mapping between "m="-lines and RTP sessions, that and the following associated lines) signals the transport flow and
grouping framework identifier will identify an RTP Session. RTP session configuration for the endpoints part of the RTP session.
[I-D.ietf-mmusic-sdp-bundle-negotiation] extends the "m-"-line for SDP grouping framework [RFC5888] allows labeling of the media
bundled media, which adds complexity to demultiplexing media stream. descriptions, for example used so that RTP Session Groups can be
Section 10.2 of [I-D.ietf-mmusic-sdp-bundle-negotiation] provides created. With Negotiating Media Multiplexing Using the Session
information about how RTP/RTCP streams are associated with SDP media Description Protocol (SDP)[I-D.ietf-mmusic-sdp-bundle-negotiation],
description. multiple media descriptions where each represents the RTP streams
sent or received for a media source are part of a common RTP session.
RTP sessions are globally unique, but their identity can only be The RTP protocol makes no normative statements about the relationship
determined by the communication context at an endpoint of the between different RTP sessions, however the applications that use
session, or by a middlebox that is aware of the session context. The more than one RTP session will have some higher layer understanding
relationship between RTP sessions depending on the underlying of the relationship between the sessions they create.
application, transport, and signalling protocol. The RTP protocol
makes no normative statements about the relationship between
different RTP sessions, however the applications that use more than
one RTP session will have some higher layer understanding of the
relationship between the sessions they create.
3.2.2. 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 has at least one SSRC identifier, even if it
identifier, even if it does not send media (endpoints that are only does not send RTP packets. RTP endpoints that are only RTP sinks
RTP sinks still send RTCP, and use their synchronisation source still send RTCP and use their SSRC identifiers in the RTCP packets
identifier in the RTCP packets they send). An endpoint can have they send. An endpoint can have multiple SSRC identifiers if it
multiple synchronisation sources identifiers if it contains multiple contains multiple RTP sources (i.e., if it sends multiple RTP
RTP sources (i.e., if it sends multiple media streams). Endpoints streams). Endpoints that are both RTP sources and RTP sinks use the
that are both RTP sources and RTP sinks use the same synchronisation same SSRC in both roles. At any given time, an RTP source has one
sources in both roles. At any given time, a RTP source has one and and only one SSRC - although that can change over the lifetime of the
only one SSRC - although that can change over the lifetime of the RTP RTP source or sink.
source or sink.
The synchronisation Source identifier is a 32-bit unsigned integer.
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.
Unless pre-signalled using the SDP "a=ssrc:" attribute [RFC5576], the
synchronisation source identifier is chosen at random. It is not
dependent on the network address of the endpoint, and is intended to
be unique within an RTP session. Synchronisation source identifier
collisions can occur, and are handled as specified in [RFC3550] and
[RFC5576], resulting in the synchronisation source identifier of the The SSRC is a 32-bit identifier. It is present in every RTP and RTCP
affecting RTP sources and/or sinks changing. An RTP source that packet header, and in the payload of some RTCP packet types. It can
changes its RTP Session identifier (e.g. source transport address) also be present in SDP signalling. Unless pre-signalled, e.g. using
during a session has to choose a new SSRC identifier to avoid being the SDP "a=ssrc:" attribute [RFC5576], the SSRC is chosen at random.
interpreted as looped source. It is not dependent on the network address of the endpoint, and is
intended to be unique within an RTP session. SSRC collisions can
occur, and are handled as specified in [RFC3550] and [RFC5576],
resulting in the SSRC of the colliding RTP sources and/or sinks
changing. An RTP source that changes its network transport address
during a session have to choose a new SSRC identifier to avoid being
interpreted as looped source, unless the transport layer mechansism,
e.g ICE [RFC5245], handle such changes
Synchronisation source identifiers that belong to the same SSRC identifiers that belong to the same synchronisation context
synchronisation context (i.e., that represent media streams that can (i.e., that represent RTP streams that can be synchronised using
be synchronised using information in RTCP SR packets) are indicated information in RTCP SR packets) use identical CNAME chunks in
by use of identical CNAME chunks in corresponding RTCP SDES packets. corresponding RTCP SDES packets. SDP signalling can also be used to
SDP signalling can also be used to provide explicit grouping of provide explicit SSRC grouping [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 RTP retransmission
Transmission of scalable video [RFC6190]. This is to be avoided [RFC4588]. This is to be avoided since there is no guarantee that
since there is no guarantee of uniqueness in SSRC values across SSRC values are unique across RTP sessions. For the RTP
RTP sessions. retransmission [RFC4588] case it is recommended to use explicit
binding of the source RTP stream and the redundancy stream, e.g.
using the RepairedRtpStreamId RTCP SDES item [I-D.ietf-avtext-rid].
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 SSRC. Each RTP source will have a different SSRC, and the
synchronisation source, and the corresponding media stream will have corresponding RTP stream will have a separate RTP sequence number and
a separate RTP sequence number and timestamp space. 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 "physical" media source, for
example a camera or microphone. example a camera or microphone.
Processed Media Source: A source with some attributed property Conceptual 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 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 an SSRC for use as source in RTCP
reports. reports.
Note that an endpoint that generates more than one media type, e.g. Note that an endpoint that generates more than one media type, e.g. a
a conference participant sending both audio and video, need not (and conference participant sending both audio and video, need not (and
commonly does not) use the same SSRC value across RTP sessions. RTCP should 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.
3.2.3. 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 an SSRC identifier is listed as a CSRC in the RTP header of a packet
header of a packet generated by an RTP mixer if the corresponding generated by an RTP mixer, if the corresponding SSRC was in the
SSRC was in the header of one of the packets that contributed to 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 an RTP 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 receiver's 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.
3.2.4. RTP Payload Type 3.2.4. RTP Payload Type
Each Media Stream utilises one or more RTP payload formats. An RTP Each RTP 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 (PT) field in the RTP packet header.
The combination therefore identifies a specific Media Stream encoding The combination of SSRC and PT therefore identifies a specific RTP
format. The format definition can be taken from [RFC3551] for stream encoding format. The format definition can be taken from
statically allocated payload types, but ought to be explicitly [RFC3551] for statically allocated payload types, but ought to be
defined in signalling, such as SDP, both for static and dynamic explicitly defined in signalling, such as SDP, both for static and
Payload Types. The term "format" here includes whatever can be dynamic payload types. The term "format" 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 "format" includes media type, RTP timestamp sampling rate, codec,
configuration, payload format configurations, and various robustness codec configuration, payload format configurations, and various
mechanisms such as redundant encodings [RFC2198]. robustness mechanisms such as redundant encodings [RFC2198].
The payload type is scoped by sending endpoint within an RTP Session. The RTP payload type is scoped by the sending endpoint within an RTP
All synchronisation sources sent from a single endpoint share the session. PT has the same meaning across all RTP streams in an RTP
same payload types definitions. The RTP Payload Type is designed session. All SSRCs sent from a single endpoint share the same
such that only a single Payload Type is valid at any time instant in payload type definitions. The RTP payload type is designed such that
the RTP source's RTP timestamp time line, effectively time- only a single payload type is valid at any time instant in the RTP
multiplexing different Payload Types if any change occurs. The source's RTP timestamp time line, effectively time-multiplexing
payload type used can change on a per-packet basis for an SSRC, for different payload types if any change occurs. The payload type used
example a speech codec making use of generic comfort noise [RFC3389]. can change on a per-packet basis for an SSRC, for example a speech
If there is a true need to send multiple Payload Types for the same codec making use of generic comfort noise [RFC3389]. If there is a
SSRC that are valid for the same instant, then redundant encodings true need to send multiple payload types for the same SSRC that are
[RFC2198] can be used. Several additional constraints than the ones valid for the same instant, then redundant encodings [RFC2198] can be
mentioned above need to be met to enable this use, one of which is used. Several additional constraints than the ones mentioned above
that the combined payload sizes of the different Payload Types ought need to be met to enable this use, one of which is that the combined
not exceed the transport MTU. payload sizes of the different payload types ought not exceed the
transport MTU. If it is acceptable to send multiple formats of the
same media source as separate RTP streams (with separate SSRC),
simulcast [I-D.ietf-mmusic-sdp-simulcast] can be used.
Other aspects of RTP payload format use are described in RTP Payload Other aspects of RTP payload format use are described in How to Write
HowTo [RFC8088]. an RTP Payload Format [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 consume and decode an RTP stream.
be viewed as selecting a rendering context. The rendering context The RTP payload type number is sometimes used to associate an RTP
can be defined by the signalling, and the RTP payload type number is stream with the signalling; this is not recommended since a specific
sometimes used to associate an RTP media stream with the signalling. payload type value can be used in multiple bundled "m=" sections
This association is possible provided unique RTP payload type numbers [I-D.ietf-mmusic-sdp-bundle-negotiation]. This association is only
are used in each context. For example, an RTP media stream can be possible if unique RTP payload type numbers are used in each context.
associated with an SDP "m=" line by comparing the RTP payload type
numbers used by the media stream with payload types signalled in the
"a=rtpmap:" lines in the media sections of the SDP. If RTP media
streams are being associated with signalling contexts based on the
RTP payload type, then the assignment of RTP payload type numbers
needs to be unique across signalling contexts; if the same RTP
payload format configuration is used in multiple contexts, then a
different RTP payload type number has to be assigned in each context
to ensure uniqueness. If the RTP payload type number is not being
used to associated RTP media streams with a signalling context, then
the same RTP payload type number can be used to indicate the exact
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.
3.3. Issues Related to RTP Topologies 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 [RFC7667]. by RTP Topology [RFC7667].
Even the most basic use case, denoted Topo-Point-to-Point in Even the most basic use case, denoted Topo-Point-to-Point in
[RFC7667], raises a number of considerations that are discussed in [RFC7667], raises a number of considerations that are discussed in
detail in following sections. They range over such aspects as: detail in following sections. They 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 per RTP session?
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 RTP streams for RTP retransmission or
or FEC. FEC?
o etc. o etc.
For some Point to Multi-point topologies (e.g. Topo-ASM and Topo-SSM For some point to multi-point topologies (e.g. Topo-ASM and Topo-SSM
in [RFC7667]), multicast is used to interconnect the session in [RFC7667]), multicast is used to interconnect the session
participants. Special considerations (documented in Section 4.2.3) participants. Special considerations (documented in Section 4.2.3)
need to be made as multicast is a one to many distribution system. are then needed as multicast is a one-to-many distribution system.
Sometimes an RTP 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 communicating peers are not compatible 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
[RFC7667]. The translator's main purpose is to make the peer look to [RFC7667]. The translator's main purpose is to make the peers look
the other peer like something it is compatible with. There can also compatible to each other. There can also be other reasons than
be other reasons than compatibility to insert a translator in the compatibility to insert a translator in the form of a middlebox or
form of a middlebox or gateway, for example a need to monitor the gateway, for example a need to monitor the RTP 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.
The point to point topology can contain one to many RTP sessions with The point to point topology can contain one to many RTP sessions with
one to many media sources per session, each having one or more RTP one to many media sources per session, each having one or more RTP
sources per media source. sources per media source.
3.4. Issues Related to RTP and RTCP Protocol 3.4. Issues Related to RTP and RTCP Protocol
Using multiple media streams is a well supported feature of RTP. Using multiple RTP streams is a well-supported feature of RTP.
However, it can be unclear for most implementers or people writing However, for most implementers or people writing RTP/RTCP
RTP/RTCP applications or extensions attempting to apply multiple applications or extensions attempting to apply multiple streams, it
streams when it is most appropriate to add an additional SSRC in an can be unclear when it is most appropriate to add an additional RTP
existing RTP session and when it is better to use multiple RTP stream in an existing RTP session and when it is better to use
sessions. This section tries to discuss the various considerations multiple RTP sessions. This section discusses the various
needed. considerations needed.
3.4.1. 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 "For efficient protocol processing, the number of multiplexing points
should be minimised, as described in the integrated layer processing should be minimised, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the design principle [ALF]. In RTP, multiplexing is provided by the
skipping to change at page 14, line 29 skipping to change at page 14, line 18
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."
Let's consider one argument at a time. The first is an argument for Let's consider one argument at a time. The first argument is for
using different SSRC for each individual media stream, which is very using different SSRC for each individual RTP stream, which is very
applicable. basic.
The second argument is advocating against using payload type The second argument is advocating against demultiplexing RTP streams
multiplexing, which still stands as can been seen by the extensive within a session based on their RTP payload type numbers, which still
list of issues found in Appendix A. stands as can been seen by the extensive 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 argument is against multiplexing RTP packets that require
require different handling into the same session. As we saw in the 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 must embed application logic
logic in order to handle streams anyway; the separation of streams to handle streams anyway; the separation of streams according to
according to stream type is just another piece of application logic, stream type is just another piece of application logic, which might
which might or might not be appropriate for a particular application. or might not be appropriate for a particular application. One type
A type of application that can mix different media sources "blindly" of application that can mix different media sources "blindly" is the
is the audio only "telephone" bridge; most other type of application audio-only "telephone" bridge; most other types of applications need
needs application-specific logic to perform the mix correctly. 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 4.2. 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 Split Component Terminal (see Section 3.10 of
or inter-connected devices handle different aspects of the whole [RFC7667]) endpoints where different processes or inter-connected
multi-media session. devices handle different aspects of the whole 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 needs to be further discussed, as imposing a latter needs further discussion, as imposing a single solution on all
single solution on all usages of RTP is inappropriate. Multiple usages of RTP is inappropriate. Multiple Media Types in an RTP
Media Types in an RTP Session specification Session specification [I-D.ietf-avtcore-multi-media-rtp-session]
[I-D.ietf-avtcore-multi-media-rtp-session] provides a detailed provides a detailed analysis of the potential issues in having
analysis of the potential issues in having multiple media types in multiple media types in the same RTP session. This document provides
the same RTP session. This document tries to provide an wider scoped a wider scope for an RTP session and considers multiple media types
consideration regarding the usage of RTP session and considers in one RTP session as a possible choice for the RTP application
multiple media types in one RTP session as possible choice for the designer.
RTP application designer.
3.4.2. 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 at one endpoint in an RTP session 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 inefficiencies. These are further discussed in potential significant inefficiencies, as further discussed in "RTP
"RTP Considerations for Endpoints Sending Multiple Media Streams" Considerations for Endpoints Sending Multiple Media Streams"
[RFC8108]. A application designer needs to consider these issues and [RFC8108]. An RTP application designer should consider these issues
the impact availability or lack of the optimization in the endpoints and the possible applicaiton impact from lack of appropriate RTP
has on their application. handling or optimization in the peer endpoints.
If an application will become affected by the issues described, using Using multiple RTP sessions can potentially mitigate application
Multiple RTP sessions can mitigate these issues. issues caused by multiple SSRCs in an RTP session.
3.4.3. Binding Related Sources 3.4.3. 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 RTP 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 a few groups:
Signalling based (SDP), grouping related RTP sessions, and grouping
SSRCs within an RTP session. Most solutions are explicit, but some o RTP/RTCP based
implicit methods have also been applied to the problem.
o Signalling based (SDP)
o grouping related RTP sessions
o grouping SSRCs within an RTP session
Most solutions are explicit, but some 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, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple sessions, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple
media descriptors as a single RTP session. media descriptions 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 groups 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 specifies a Session Description Protocol (SDP) Grouping mechanism
their allocation to RTP sessions. for RTP streams that can be used to specify relations between RTP
streams. This mechanism is used to signal the association between
the SDP concept of "media description" and the WebRTC concept of
"MediaStream" / "MediaStreamTrack" (Corresponds to the [RFC7656]
term "Source Stream") using SDP signalling.
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 an endpoint or An RTP/RTCP-based solution is to use the RTCP SDES CNAME to bind the
synchronization context, i.e. the CNAME has not been sufficient and RTP streams to an endpoint or synchronization context. For
one way to bind related streams in multiple RTP sessions has been to applications with a single RTP stream per type (Media, Source or
use the same SSRC value across all the RTP sessions. RTP Redundancy) this is sufficient independent if one or more RTP
Retransmission [RFC4588] is multiple RTP session mode, Generic FEC sessions are used. However, some applications choose not to use it
[RFC5109], as well as the RTP payload format for Scalable Video because of perceived complexity or a desire not to implement RTCP and
Coding [RFC6190] in Multi Session Transmission (MST) mode uses this instead use the same SSRC value to bind related RTP streams across
method. This method clearly works but might have some downside in multiple RTP sessions. RTP Retransmission [RFC4588] in multiple RTP
RTP sessions with many participating SSRCs. The birthday paradox session mode and Generic FEC [RFC5109] both use this method. This
ensures that if you populate a single session with 9292 SSRCs at method may work but might have some downsides in RTP sessions with
random, the chances are approximately 1% that at least one collision many participating SSRCs. When an SSRC collision occurs, this will
will occur. When a collision occur this will force one to change force one to change SSRC in all RTP sessions and thus resynchronize
SSRC in all RTP sessions and thus resynchronizing all of them instead all of them instead of only the single media stream having the
of only the single media stream having the collision. Therefore it collision. Therefore, it is not recommended to use identical SSRC
is not recommended to use such method. Using [RFC7656] streams from values to relate RTP streams.
the same media source should use the same RTP session.
It can be noted that Section 8.3 of the RTP Specification [RFC3550]
recommends using a single SSRC space across all RTP sessions for
layered coding.
Another solution that has been applied to binding SSRCs has been an Another solution to bind SSRCs is an implicit method used by RTP
implicit method used by RTP Retransmission [RFC4588] when doing Retransmission [RFC4588] when doing retransmissions in the same RTP
retransmissions in the same RTP session as the source RTP media session as the source RTP stream. The receiver missing a packet
stream. This issues an RTP retransmission request, and then await a issues an RTP retransmission request, and then awaits a new SSRC
new SSRC carrying the RTP retransmission payload and where that SSRC carrying the RTP retransmission payload and where that SSRC is from
is from the same CNAME. This limits a requestor to having only one 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.
[I-D.ietf-mmusic-rid] provides an RTP/RTCP based mechanism capable of RTP Payload Format Restrictions [I-D.ietf-mmusic-rid] provides an
supporting explicit association within an RTP session. RTP/RTCP based mechanism to unambiguously identify the RTP streams
within an RTP session and restrict the streams' payload format
parameters in a codec-agnostic way beyond what is provided with the
regular Payload Types. The mapping is done by specifying an "a=rid"
value in the SDP offer/answer signalling and having the corresponding
"rtp-stream-id" value as an SDES item and an RTP header extension.
The RID solution also includes a solution for binding redundancy RTP
streams to their original source RTP streams, given that those use
RID identifiers.
It can be noted that Section 8.3 of the RTP Specification [RFC3550]
recommends using a single SSRC space across all RTP sessions for
layered coding. Based on the experience so far however, we recommend
to use a solution doing explicit binding between the RTP streams so
what the used SSRC values are do not matter. That way solutions
using multiple RTP streams in a single RTP session and multiple RTP
sessions uses the same solution.
3.4.4. 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 the redundant information protects. This
information also needs to be transmitted as its own media stream, or sequence of redundant information also needs to be transmitted as its
in some cases instead of the original media stream. Thus many of own media stream, or in some cases, instead of the original media
these schemes create a need for binding related flows as discussed stream. Thus, many of these schemes create a need for binding
above. Looking at the history of these schemes, there are schemes related flows as discussed above. Looking at the history of these
using multiple SSRCs and schemes using multiple RTP sessions, and schemes, there are schemes using multiple SSRCs and schemes using
some schemes that support both modes of operation. multiple RTP sessions, and 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 and if separating RTP sessions
on transport level, FEC can easily be ignored already on transport
level.
In usages involving multicast, having the FEC information on its own In usages involving multicast, having the FEC information on its own
multicast group allows for flexibility. This is especially useful multicast group allows for similar flexibility. This is especially
when receivers see very heterogeneous packet loss rates. Those useful when receivers see very heterogeneous packet loss rates.
receivers that are not seeing packet loss don't need to join the Those receivers that are not seeing packet loss don't need to join
multicast group with the FEC data, and so avoid the overhead of the multicast group with the FEC data, and so avoid the overhead of
receiving unnecessary FEC packets, for example. receiving unnecessary FEC packets, for example.
4. Particular Considerations for RTP Multiplexing 4. Considerations for RTP Multiplexing
4.1. 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; interworking between different applications including
applications and the implications of potentially different choices of the implications of potentially different RTP multiplexing point
usage of RTP's multiplexing points. The second topic relates to what choices and limitations that have to be considered when working with
limitations have to be considered working with some legacy some legacy applications.
applications.
4.1.1. Types of Interworking 4.1.1. Application Interworking
It is not uncommon that applications or services of similar usage, It is not uncommon that applications or services of similar but not
especially the ones intended for interactive communication, encounter identical usage, especially the ones intended for interactive
a situation where one want to interconnect two or more of these communication, encounter a situation where one want to interconnect
applications. two or more of these 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 must then either
change the multiplexing structure or adhere to limitations in each change the multiplexing structure or adhere to the respective
application. limitations in each 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, with the two applications being 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.
4.1.2. 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, and have the same
capabilities in number of simultaneous media streams combined with capabilities in number of simultaneous RTP streams combined with the
the same set of RTP/RTCP extensions being supported. Unfortunately same set of RTP/RTCP extensions being supported. Unfortunately, this
this might not always be true. might not always be true.
When one is gatewaying via an RTP Translator, a natural requirement When one is gatewaying via an RTP Translator, an important
is that the two applications being interconnected need to use the consideration is if the two applications being interconnected need to
same approach to multiplexing. Furthermore, if one of the use the same approach to multiplexing. If one side is using RTP
applications is capable of working in several modes (such as being session multiplexing and the other is using SSRC multiplexing with
able to use Additional SSRCs or Multiple RTP sessions at will), and bundle, the mapping of SDP "m=" lines between both sides requires
the other one is not, successful interconnection depends on locking that the order in bundled and not bundled sides will be the same to
the more flexible application into the operating mode where allow routing without mapping, it is possible for the RTP translator
interconnection can be successful, even if no participants using the to map the RTP streams between both sides. There are also challenges
less flexible application are present when the RTP sessions are being with SSRC collision handling since there may be a collision on the
SSRC multiplexing side but the RTP session multiplexing side will not
be aware of any collision unless SSRC translation is applied on the
RTP translator. Furthermore, if one of the applications is capable
of working in several modes (such as being able to use additional RTP
streams in one RTP session or multiple RTP sessions at will), and the
other one is not, successful interconnection depends on locking the
more flexible application into the operating mode where
interconnection can be successful, even if no participants are using
the less flexible application when the RTP sessions are being
created. created.
4.1.3. 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 RTP streams (possibly
(possibly based on incoming RTCP reports), originating from SSRCs based on incoming RTCP reports), originating from SSRCs controlled
controlled by the gateway. 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.
For applications that uses any security mechanism, e.g. in the form For applications that uses any security mechanism, e.g. in the form
of SRTP, then the gateway needs to be able to decrypt incoming of SRTP, the gateway needs to be able to decrypt incoming packets and
packets and re-encrypt them in the other application's security re-encrypt them in the other application's security context. This is
context. This is necessary even if all that's needed is a simple necessary even if all that's needed is a simple remapping of SSRC
remapping of SSRC numbers. If this is done, the gateway also needs numbers. If this is done, the gateway also needs to be a member of
to be a member of the security contexts of both sides, of course. 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), media-level adaptations that cannot be
requirements), suppression of content that is known not to be handled solved through media negotiation (such as rescaling for incompatible
in the destination application, or the addition or removal of video size requirements), suppression of content that is known not to
redundancy coding or scalability layers to fit the need of the be handled in the destination application, or the addition or removal
of redundancy coding or scalability layers to fit the needs of the
destination domain. destination domain.
From the above, we can see that the gateway needs to have an intimate From the above, we can see that the gateway needs to have an intimate
knowledge of the application requirements; a gateway is by its nature knowledge of the application requirements; a gateway is by its nature
application specific, not a commodity product. application specific, not a commodity product.
This fact reveals the potential for these gateways to block evolution This fact reveals the potential for these gateways to block
of the applications by blocking unknown RTP and RTCP extensions that application evolution by blocking RTP and RTCP extensions that the
the regular application has been extended with. applications have been extended with but that are unknown to the
gateway.
If one uses security functions, like SRTP, they can as seen above If one uses security functions, like SRTP, and as can be seen from
incur both additional risk due to the gateway needing to be in above, they incur both additional risk due to the requirement to have
security association between the endpoints, unless the gateway is on the gateway in the security association between the endpoints (unless
the transport level, and additional complexities in form of the the gateway is on the transport level), and additional complexities
decrypt-encrypt cycles needed for each forwarded packet. SRTP, due in form of the decrypt-encrypt cycles needed for each forwarded
to its keying structure, also requires that each RTP session needs packet. SRTP, due to its keying structure, also requires that each
different master keys, as use of the same key in two RTP sessions for RTP session needs different master keys, as use of the same key in
some ciphers can result in two-time pads that completely breaks the two RTP sessions can for some ciphers result in two-time pads that
confidentiality of the packets. completely breaks the confidentiality of the packets.
4.1.4. Multiple SSRC Legacy Considerations 4.1.4. 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.
When establishing RTP sessions that can contain endpoints that aren't When establishing RTP sessions that can contain endpoints that aren't
updated to handle multiple streams following these recommendations, a updated to handle multiple streams following these recommendations, a
particular application can have issues with multiple SSRCs within a particular application can have issues with multiple SSRCs within a
single session. These issues include: single session. These issues include:
1. Need to handle more than one stream simultaneously rather than 1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one. replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously. 2. Be capable of decoding multiple streams simultaneously.
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 RTP 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.
4.2. Network Considerations 4.2. Network Considerations
The multiplexing choice has impact on network level mechanisms that The RTP multiplexing choice has impact on network level mechanisms
need to be considered by the implementer. that need to be considered by the implementer.
4.2.1. 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 packet marking based. RSVP [RFC2205] is an example of a based or packet marking based. RSVP [RFC2205] is an example of a
flow based mechanism, while Diff-Serv [RFC2474] is an example of a flow based mechanism, while Diff-Serv [RFC2474] is an example of a
packet marking based one. For a packet marking based scheme, the packet marking based one. For a packet marking based scheme, the
method of multiplexing will not 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 multiplexing methods. Additional SSRC will result in all RTP
part of the same 5-tuple (protocol, source address, destination streams being part of the same 5-tuple (protocol, source address,
address, source port, destination port) which is the most common destination address, source port, destination port) which is the most
selector for flow based QoS. common selector for flow based QoS.
It also needs to be noted that packet marking based QoS mechanisms It must also be noted that packet marking based QoS mechanisms can
can have limitations. A general observation is that different DSCP have limitations. A general observation is that different
can be assigned to different packets within a flow as well as within Differentiated Services Code Points (DSCP) can be assigned to
an RTP Media Stream. However, care needs to be taken when different packets within a flow as well as within an RTP stream.
considering which forwarding behaviours that are applied on path due However, care must be taken when considering which forwarding
to these DSCPs. In some cases the forwarding behaviour can result in behaviours that are applied on path due to these DSCPs. In some
packet reordering. For more discussion of this see [RFC7657]. cases the forwarding behaviour can result in packet reordering. For
more discussion of this see [RFC7657].
More specific to the choice between using one or more RTP session can The method for assigning marking to packets can impact what number of
be the method for assigning marking to packets. If this is done RTP sessions to choose. If this marking is done using a network
using a network ingress function, it can have issues discriminating ingress function, it can have issues discriminating the different RTP
the different RTP media streams. The network API on the endpoint streams. The network API on the endpoint also needs to be capable of
also needs to be capable of setting the marking on a per packet basis setting the marking on a per-packet basis to reach the full
to reach the full functionality. functionality.
4.2.2. 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 analyse 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:
skipping to change at page 21, line 39 skipping to change at page 22, line 18
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: Performing the NAT/FW traversal takes a NAT Traversal Extra Delay: 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. That assumes a message in one direction, and then an
assumes a message in one direction, and then an immediate immediate triggered check back. The reason it isn't more, is that
triggered check back. The reason it isn't more, is that ICE first ICE first finds one candidate pair that works prior to attempting
finds one candidate pair that works prior to attempting to to establish multiple flows. Thus, there is no extra time until
establish multiple flows. Thus, there is no extra time until one one has found a working candidate pair. Based on that working
has found a working candidate pair. Based on that working pair pair the needed extra time is to in parallel establish the, in
the needed extra time is to in parallel establish the, in most most cases 2-3, additional flows. However, packet loss causes
cases 2-3, additional flows. However, packet loss causes extra extra delays, at least 100 ms, which is the minimal retransmission
delays, at least 100 ms, which is the minimal retransmission timer 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 Using additional RTP streams in the same RTP session and transport
Session and transport flow and does not introduce any additional NAT flow does not introduce any additional NAT traversal complexities per
traversal complexities per media stream. This can be compared with RTP stream. This can be compared with normally one or two additional
normally one or two additional transport flows per RTP session when transport flows per RTP session when using multiple RTP sessions.
using multiple RTP sessions. Additional lower layer transport flows Additional lower layer transport flows will be needed, unless an
will be needed, unless an explicit de-multiplexing layer is added explicit de-multiplexing layer is added between RTP and the transport
between RTP and the transport protocol. At time of writing no such protocol. At time of writing no such mechanism was defined.
mechanism was defined.
4.2.3. Multicast 4.2.3. Multicast
Multicast groups provides a powerful semantics for a number of real- Multicast groups provides a powerful tool for a number of real-time
time applications, especially the ones that desire broadcast-like 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. There are also the RTP/RTCP extension to
certain number of limitations. better support Source Specific Multicast (SSM) [RFC5760]. Another
application is the Many to Many communication, which RTP [RFC3550]
was originally built to support. But the multicast semantics do
result in a 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. For broadcast type of applications this is not
various receiver based solutions are employed to ensure that the acceptable. Instead, various receiver-based solutions are employed
receivers achieve best possible performance. By using scalable to ensure that the receivers achieve best possible performance. By
encoding and placing each scalability layer in a different multicast using scalable encoding and placing each scalability layer in a
group, the receiver can control the amount of traffic it receives. different multicast group, the receiver can control the amount of
To have each scalability layer on a different multicast group, one traffic it receives. To have each scalability layer on a different
RTP session per multicast group is used. multicast group, one 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 with port translation are not useful in different from unicast; NATs with port translation are not useful in
the multicast environment, meaning that the entire port range of each the multicast environment, meaning that the entire port range of each
multicast address is available for distinguishing between RTP multicast address is available for distinguishing between RTP
sessions. sessions.
Thus it appears easiest and most straightforward to use multiple RTP Thus, when using broadcast applications it appears easiest and most
sessions for sending different media flows used for adapting to straightforward to use multiple RTP sessions for sending different
network conditions. It is also common that streams that improve media flows used for adapting to network conditions. It is also
transport robustness are sent in their own multicast group to allow common that streams that improve transport robustness are sent in
for interworking with legacy or to support different levels of their own multicast group to allow for interworking with legacy or to
protection. support different levels of protection.
Here are some common behaviours for RTP multicast: For many to many applications there is different needs. Here it will
depend on how the actual application is realized what is the most
appropriate choice. With sender side congestion control there might
not exist any benefit with using multiple RTP session.
1. Multicast applications use a group of RTP sessions, not one. The properties of a broadcast application using RTP multicast:
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 1. Uses 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. much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the 3. The applications need signalling functions to identify the
relationships between RTP sessions. relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the 4. The applications need signalling or RTP/RTCP functions to
relationships between SSRCs in different RTP sessions. identify the relationships between SSRCs in different RTP
sessions when needs beyond CNAME exists.
All multicast configurations share a signalling requirement; all of Both broadcast and many to many multicast applications do share a
the participants will need to have the same RTP and payload type signalling requirement; all of the participants will need to have the
configuration. Otherwise, A could for example be using payload type same RTP and payload type configuration. Otherwise, A could for
97 as the video codec H.264 while B thinks it is MPEG-2. It is to be example be using payload type 97 as the video codec H.264 while B
noted that SDP offer/answer [RFC3264] is not appropriate for ensuring thinks it is MPEG-2. It is to be noted that SDP offer/answer
this property. The signalling aspects of multicast are not explored [RFC3264] is not appropriate for ensuring this property in broadcast/
further in this memo. multicast context. The signalling aspects of broadcast/multicast are
not explored further in this memo.
Security solutions for this type of group communications are also Security solutions for this type of group communications are also
challenging. First of all the key-management and the security challenging. First, the key-management and the security protocol
protocol needs to support group communication. Source authentication needs to support group communication. Second, source authentication
requires special solutions. For more discussion on this please requires special solutions. For more discussion on this please
review Options for Securing RTP Sessions [RFC7201]. review Options for Securing RTP Sessions [RFC7201].
4.3. Security and Key Management Considerations 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 [RFC7201]. review RTP Security Options [RFC7201].
4.3.1. 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 are (so far) 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 and not by another party having
access to the key. In most cases this is a sufficient security access to the key. At least unless TESLA source authentication
property, but there are a few cases where this does create issues. [RFC4383], which adds delay to achieve source authentication. In
most cases symmetric ciphers provide sufficient security properties,
but there are a few cases where this does create issues.
The first case is when someone leaves a multi-party session and one The first case is when someone leaves a multi-party session and one
wants to ensure that the party that left can no longer access the wants to ensure that the party that left can no longer access the RTP
media streams. This requires that everyone re-keys without streams. This requires that everyone re-keys without disclosing the
disclosing the keys to the excluded party. keys to the excluded party.
A second case is when using security as an enforcing mechanism for A second case is when using security as an enforcing mechanism for
differentiation. Take for example a scalable layer or a high quality differentiation. Take for example a scalable layer or a high quality
simulcast version which only premium users are allowed to access. simulcast version that only premium users are allowed to access. The
The mechanism preventing a receiver from getting the high quality mechanism preventing a receiver from getting the high quality stream
stream can be based on the stream being encrypted with a key that can be based on the stream being encrypted with a key that user can't
user can't access without paying premium, having the key-management access without paying premium, having the key-management limit access
limit access to the key. to the key.
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 sets 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.
4.3.2. Key Management for Multi-party session 4.3.2. Key Management for Multi-party sessions
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-perc-srtp-ekt-diet] 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 is needed.
4.3.3. 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 from the choice of multiplexing and topology. This becomes
evident in RTP topologies having any type of middlebox that processes especially evident in RTP topologies having any type of middlebox
or modifies RTP/RTCP packets. Where there is very small overhead for that processes or modifies RTP/RTCP packets. Where there is very
an RTP translator or mixer to rewrite an SSRC value in the RTP packet small overhead for an RTP translator or mixer to rewrite an SSRC
of an unencrypted session, the cost of doing it when using value in the RTP packet of an unencrypted session, the cost is higher
cryptographic security functions is higher. For example if using when using cryptographic security functions. 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 SSRC, the
encryption and authentication tag needs to be performed using another encryption and authentication must use another key. Thus, changing
key. Thus changing the SSRC value implies a decryption using the old the SSRC value implies a decryption using the old SSRC and its
SSRC and its security context followed by an encryption using the new security context, followed by an encryption using the new one.
one.
5. Archetypes 5. RTP Multiplexing Design Choices
This section discusses some archetypes of how RTP multiplexing can be This section discusses how some RTP multiplexing design choices can
used in applications to achieve certain goals and a summary of their be used in applications to achieve certain goals, and a summary of
implications. For each archetype there is discussion of benefits and the implications of such choices. For each design there is
downsides. discussion of benefits and downsides.
5.1. Single SSRC per Session 5.1. Single SSRC per Endpoint
In this archetype each endpoint in a point-to-point session has only In this design each endpoint in a point-to-point session has only a
a single SSRC, thus the RTP session contains only two SSRCs, one single SSRC, thus the RTP session contains only two SSRCs, one local
local and one remote. This session can be used both unidirectional, and one remote. This session can be used both unidirectional, i.e.
i.e. only a single media stream or bi-directional, i.e. both only a single RTP stream or bi-directional, i.e. both endpoints have
endpoints have one media stream each. If the application needs one RTP stream each. If the application needs additional media flows
additional media flows between the endpoints, they will have to between the endpoints, they will have to establish additional RTP
establish additional RTP sessions. sessions.
The Pros: The Pros:
1. This archetype has great legacy interoperability potential as it 1. This design 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
the exact formats and bit-rates for each media stream, especially the exact formats and bit-rates for each RTP stream, especially
using today's tools in SDP. using today's tools in SDP.
3. It does not matter if usage or purpose of the media stream is 3. It is possible to control security association per RTP stream
signalled on media stream level or session level as there is no with current key-management, since each RTP stream is directly
difference. related to an RTP session, and the most used keying mechanisms
operates on a per-session basis.
4. It is possible to control security association per RTP media
stream with current key-management, since each media stream is
directly related to an RTP session, and the keying operates on a
per-session basis.
The Cons: The Cons:
a. The number of RTP sessions grows directly in proportion with the a. The number of RTP sessions grows directly in proportion with the
number of media streams, which has the implications: number of RTP streams, which has the implications:
* Linear growth of the amount of NAT/FW state with number of * Linear growth of the amount of NAT/FW state with number of RTP
media streams. streams.
* Increased delay and resource consumption from NAT/FW * Increased delay and resource consumption from NAT/FW
traversal. traversal.
* Likely larger signalling message and signalling processing * Likely larger signalling message and signalling processing
requirement due to the amount of session related information. requirement due to the amount of session related information.
* Higher potential for a single media stream to fail during * Higher potential for a single RTP stream to fail during
transport between the endpoints. transport between the endpoints.
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 RTP stream also grows, linearly, depending on
exponentially, depending on how the application needs to relate how the application needs to relate RTP 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 RTP stream usage is highly dynamic,
their usage, i.e. entering and leaving, the amount of signalling i.e. entering and leaving, the amount of signalling can grow
can grow high. Issues arising from the timely establishment of high. Issues can also arise from the timely establishment of
additional RTP sessions can also arise. additional RTP sessions.
e. Cross session RTCP requests might be needed, and the fact that
they're impossible can cause issues.
f. If the same SSRC value is reused in multiple RTP sessions rather
than being randomly chosen, interworking with applications that
uses another multiplexing structure than this application will
require SSRC translation.
g. Cannot be used with Any Source Multicast (ASM) as one cannot
guarantee that only two endpoints participate as packet senders.
Using SSM, it is possible to restrict to these requirements if no
RTCP feedback is injected back into the SSM group.
h. For most security mechanisms, each RTP session or transport flow e. If, against the recommendation, the same SSRC value is reused in
requires individual key-management and security association multiple RTP sessions rather than being randomly chosen,
establishment thus increasing the overhead. interworking with applications that use a different multiplexing
structure will require SSRC translation.
RTP applications that need to inter-work with legacy RTP RTP applications that need to interwork with legacy RTP applications
applications, like most deployed VoIP and video conferencing can potentially benefit from this structure. However, a large number
solutions, can potentially benefit from this structure. However, a of media descriptions in SDP can also run into issues with existing
large number of media descriptions in SDP can also run into issues implementations. For any application needing a larger number of
with existing implementations. For any application needing a larger media flows, the overhead can become very significant. This
number of media flows, the overhead can become very significant. structure is also not suitable for multi-party sessions, as any given
This structure is also not suitable for multi-party sessions, as any RTP stream from each participant, although having same usage in the
given media stream from each participant, although having same usage application, needs its own RTP session. In addition, the dynamic
in the application, needs its own RTP session. In addition, the behaviour that can arise in multi-party applications can tax the
dynamic behaviour that can arise in multi-party applications can tax signalling system and make timely media establishment more difficult.
the signalling system and make timely media establishment more
difficult.
5.2. 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 design, 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 RTP 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:
1. Low number of RTP sessions needed compared to single SSRC case. 1. Low number of RTP sessions needed compared to Single SSRC per
This implies: Endpoint case. This implies:
* Reduced NAT/FW state * Reduced NAT/FW state
* Lower NAT/FW Traversal Cost in both processing and delay. * Lower NAT/FW Traversal Cost in both processing and delay.
2. Allows for early de-multiplexing in the processing chain in RTP 2. Works well with Split Component Terminal (see Section 3.10 of
applications where all media streams of the same type have the [RFC7667]) where the split is per media type.
same usage in the application.
3. Works well with media type de-composite endpoints.
4. Enables Flow-based QoS with different prioritisation between 3. Enables Flow-based QoS with different prioritisation between
media types. media types.
5. For applications with dynamic usage of media streams, i.e. they 4. For applications with dynamic usage of RTP streams, i.e.
come and go frequently, having much of the state associated with frequently added and removed, having much of the state associated
the RTP session rather than an individual SSRC can avoid the need with the RTP session rather than per individual SSRC can avoid
for in-session signalling of meta-information about each SSRC. the need for in-session signalling of meta-information about each
SSRC.
6. Low overhead for security association establishment. 5. Low overhead for security association establishment.
The Cons: The Cons:
a. May have some need for cross session RTCP requests for things a. Some potential for concern with legacy implementations that don't
that affect both media types in an asynchronous way. support the RTP specification fully when it comes to handling
b. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per endpoint. multiple SSRC per endpoint.
c. Will not be able to control security association for sets of b. Not possible to control security association for sets of RTP
media streams within the same media type with today's key- streams within the same media type with today's key- management
management mechanisms, unless these are split into different RTP mechanisms, unless these are split into different RTP sessions.
sessions.
For RTP applications where all media streams of the same media type For RTP applications where all RTP streams of the same media type
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 enables 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.
5.3. 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 This design goes one step further than above (Section 5.2) by using
(Section 5.2) by using multiple RTP sessions also for a single media multiple RTP sessions also for a single media type. The main reason
type, but still not as far as having a single SSRC per RTP session. for going in this direction is that the RTP application needs
The main reason for going in this direction is that the RTP separation of the RTP streams due to their usage. Some typical
application needs separation of the media streams due to their usage. reasons for going to this design are scalability over multicast,
Some typical reasons for going to this archetype are scalability over simulcast, need for extended QoS prioritisation of RTP streams due to
multicast, simulcast, need for extended QoS prioritisation of media their usage in the application, or the need for fine- grained
streams due to their usage in the application, or the need for fine- signalling using today's tools.
grained signalling using today's tools.
The Pros: The Pros:
1. More suitable for Multicast usage where receivers can 1. More suitable for multicast usage where receivers can
individually select which RTP sessions they want to participate individually select which RTP sessions they want to participate
in, assuming each RTP session has its own multicast group. in, assuming each RTP session has its own multicast group.
2. Indication of the application's usage of the media stream, where 2. The application can indicate its usage of the RTP streams on RTP
multiple different usages exist. session level, in case multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media 3. Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely signalling. stream and thus reduced need for explicit and timely signalling.
4. Enables detailed QoS prioritisation for flow based mechanisms. 4. Enables detailed QoS prioritisation for flow-based mechanisms.
5. Works well with de-composite endpoints.
6. Handles dynamic usage of media streams well.
7. For transport translator based multi-party sessions, this 5. Works well with Split Component Terminal (see Section 3.10 of
structure allows for improved control of which type of media [RFC7667]).
streams an endpoint receives.
8. The scope for who is included in a security association can be 6. The scope for who is included in a security association can be
structured around the different RTP sessions, thus enabling such structured around the different RTP sessions, thus enabling such
functionality with existing key-management. functionality with existing key-management.
The Cons: The Cons:
a. Increases the amount of RTP sessions compared to Multiple SSRCs a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type. of the Same Media Type.
b. Increased amount of session configuration state. b. Increased amount of session configuration state.
c. May need synchronised cross-session RTCP requests and require c. For RTP streams that are part of scalability, simulcast or
some consideration due to this. transport robustness, a method to bind sources across multiple
RTP sessions is needed.
d. For media streams that are part of scalability, simulcast or
transport robustness it will be needed to bind sources, which
need to support multiple RTP sessions.
e. Some potential for concern with legacy implementations that does d. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling not support the RTP specification fully when it comes to handling
multiple SSRC per endpoint. multiple SSRC per endpoint.
f. Higher overhead for security association establishment. e. Higher overhead for security association establishment due to the
increased number of RTP sessions.
g. If the applications need finer control than on media type level f. If the applications need finer control than on RTP session level
over which session participants that are included in different over which participants that are included in different sets of
sets of security associations, most of today's key-management security associations, most of today's key-management will have
will have difficulties establishing such a session. 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 RTP 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.
5.4. 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 design uses a single RTP session for multiple different media
media types, like audio and video, and possibly also transport types, like audio and video, and possibly also transport robustness
robustness mechanisms like FEC or Retransmission. Each media stream mechanisms like FEC or Retransmission. An endpoint can have zero,
will use its own SSRC and a given SSRC value from a particular one or more media sources per media type. Resulting in a number of
endpoint will never use the SSRC for more than a single media type. RTP streams of various media types and both source and redundancy
type.
The Pros: The Pros:
1. Single RTP session which implies: 1. Single RTP session which implies:
* Minimal NAT/FW state. * Minimal NAT/FW state.
* Minimal NAT/FW Traversal Cost. * Minimal NAT/FW Traversal Cost.
* Fate-sharing for all media flows. * Fate-sharing for all media flows.
2. Enables separation of the different media types based on the 2. Can handle dynamic allocations of RTP streams well on an RTP
payload types so media type specific endpoint or central
processing can still be supported despite single session.
3. Can handle dynamic allocations of media streams well on an RTP
level. Depends on the application's needs for explicit level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be indication of the stream usage and how timely that can be
signalled. signalled.
4. Minimal overhead for security association establishment. 3. Minimal overhead for security association establishment.
The Cons: The Cons:
a. Less suitable for interworking with other applications that uses a. Less suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a individual RTP sessions per media type or multiple sessions for a
single media type, due to need of SSRC translation. single media type, due to the potential need of SSRC translation.
b. Negotiation of bandwidth for the different media types is b. Negotiation of bandwidth for the different media types is
currently not possible in SDP. This requires SDP extensions to currently only possible using RID [I-D.ietf-mmusic-rid] in SDP.
enable payload or source specific bandwidth. Likely to be a
problem due to media type asymmetry in needed bandwidth.
c. Not suitable for de-composite endpoints. c. Not suitable for Split Component Terminal (see Section 3.10 of
[RFC7667]).
d. Flow based QoS cannot provide separate treatment to some media d. Flow-based QoS cannot provide separate treatment of RTP streams
streams compared to others in the single RTP session. compared to others in the single RTP session.
e. If there is significant asymmetry between the media streams' RTCP e. If there is significant asymmetry between the RTP streams' RTCP
reporting needs, there are some challenges in configuration and reporting needs, there are some challenges in configuration and
usage to avoid wasting RTCP reporting on the media stream that usage to avoid wasting RTCP reporting on the RTP stream that does
does not need that frequent reporting. not need that frequent reporting.
f. Not suitable for applications where some receivers like to f. Not suitable for applications where some receivers like to
receive only a subset of the media streams, especially if receive only a subset of the RTP streams, especially if multicast
multicast or transport translator is being used. or transport translator is being used.
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.
5.5. Summary 5.5. Summary
There are some clear relations between these archetypes. Both the There are some clear similarities between these designs. Both the
"single SSRC per RTP session" and the "multiple media types in one "Single SSRC per Endpoint" and the "Multiple Media Types in one
session" are cases which require full explicit signalling of the Session" are cases that require full explicit signalling of the media
media stream relations. However, they operate on two different stream relations. However, they operate on two different levels
levels where the first primarily enables session level binding, and where the first primarily enables session level binding, and the
the second needs to do it all on SSRC level. From another second needs SSRC level binding. From another perspective, the two
perspective, the two solutions are the two extreme points when it solutions are the two extreme points when it comes to number of RTP
comes to number of RTP sessions needed. sessions needed.
The two other archetypes "Multiple SSRCs of the Same Media Type" and The two other designs "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for one Media Type" are examples of two other "Multiple Sessions for one Media Type" are two examples that
cases that first of all allows for some implicit mapping of the role primarily allows for some implicit mapping of the role or usage of
or usage of the media streams based on which RTP session they appear the RTP streams based on which RTP session they appear in. It thus
in. It thus potentially allows for less signalling and in particular potentially allows for less signalling and in particular reduces the
reduced need for real-time signalling in dynamic sessions. They also need for real-time signalling in dynamic sessions. They also
represent points in between the first two when it comes to amount of represent points in between the first two designs when it comes to
RTP sessions established, i.e. representing an attempt to reduce the amount of RTP sessions established, i.e. representing an attempt to
amount of sessions as much as possible without compromising the balance the amount of RTP sessions with the functionality the
functionality the session provides both on network level and on communication session provides both on network level and on
signalling level. signalling level.
6. Summary considerations and guidelines 6. Guidelines
6.1. Guidelines
This section contains a number of recommendations for implementers or This section contains a number of multi-stream guidelines for
specification writers when it comes to handling multi-stream. implementers or specification writers.
Do not Require the same SSRC across Sessions: As discussed in Do not use the same SSRC across RTP sessions: As discussed in
Section 3.4.3 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 RTP streams
together. It is instead suggested that a mechanism to explicitly together. It is instead recommended to use a mechanism to
signal the relation is used, either in RTP/RTCP or in the used explicitly signal the relation, either in RTP/RTCP or in the
signalling mechanism that establishes the RTP session(s). signalling mechanism used to establish the RTP session(s).
Use additional SSRCs for additional Media Sources: In the cases Use additional RTP streams for additional media sources: In the
where an RTP endpoint needs to transmit additional media streams cases where an RTP endpoint needs to transmit additional RTP
of the same media type in the application, with the same streams of the same media type in the application, with the same
processing requirements at the network and RTP layers, it is processing requirements at the network and RTP layers, it is
suggested to send them as additional SSRCs in the same RTP suggested to send them in the same RTP session. For example a
session. For example a telepresence room where there are three telepresence room where there are three cameras, and each camera
cameras, and each camera captures 2 persons sitting at the table, captures 2 persons sitting at the table, sending each camera as
sending each camera as its own SSRC within a single RTP session is its own RTP stream within a single RTP session is suggested.
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 RTP 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 a 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/RTCP Extensions Support Multiple RTP Streams as well as Multiple
RTP sessions: 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 use 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 must support both. Extensions that are not as generally
as generally applicable will have to consider if interoperability applicable will have to consider if interoperability is better
is better served by defining a single solution or providing both 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 must include support for both multiple RTP
additional SSRCs and multiple RTP sessions so that application streams in the same RTP sessions and multiple RTP sessions, such
developers can choose freely from the set of mechanisms without that application developers can choose freely from the set of
concerning themselves with which of the multiplexing choices a mechanisms without concerning themselves with which of the
particular solution supports. multiplexing choices a particular solution supports.
7. Open Issues 7. Open Issues
There are currently some issues that needs to be resolved before this There are currently some issues that needs to be resolved before this
document is ready to be published: document is ready to be published:
1. Use of RFC 2119 language is section on SSRC (3.2.2) 1. Does the MSID text need to be updated and clarified based on the
evolution of MSID since previous version. Section 3.4.3.
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. 2. Changed definitions needs review and consideration.
8. IANA Considerations 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.
9. Security Considerations 9. Security Considerations
There is discussion of the security implications of choosing SSRC vs The security considerations of the RTP specification [RFC3550] and
Multiple RTP session in Section 4.3. any applicable RTP profile [RFC3551],[RFC4585],[RFC3711], the
extensions for sending multiple media types in a single RTP session
[I-D.ietf-avtcore-multi-media-rtp-session], MSID
[I-D.ietf-mmusic-msid], RID [I-D.ietf-mmusic-rid], BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation], [RFC5760], [RFC5761], apply
if selected and thus needs to be considered in the evaluation.
10. References There is discussion of the security implications of choosing multiple
SSRC vs multiple RTP sessions in Section 4.3.
10.1. Normative References 10. Contributors
Hui Zheng (Marvin) from Huawei contributed to WG draft versions -04
and -05 of the document.
11. References
11.1. 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, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>. July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656, for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015, DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>. <https://www.rfc-editor.org/info/rfc7656>.
10.2. Informative References 11.2. 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 Symposium on Communications Architectures and
Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
Computer Communications Review, Vol. 20(4), September Computer Communications Review, Vol. 20(4), September
1990. 1990.
[I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011.
[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-13 (work in ietf-avtcore-multi-media-rtp-session-13 (work in
progress), December 2015. progress), December 2015.
[I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", draft-ietf-avtext-
rid-09 (work in progress), October 2016.
[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-16 Session Description Protocol", draft-ietf-mmusic-msid-16
(work in progress), February 2017. (work in progress), February 2017.
[I-D.ietf-mmusic-rid] [I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B., Roach, A., "RTP Payload Format Restrictions", draft-ietf-
Roach, A., and B. Campen, "RTP Payload Format mmusic-rid-15 (work in progress), May 2018.
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-39 (work in progress), August 2017. negotiation-52 (work in progress), May 2018.
[I-D.lennox-mmusic-sdp-source-selection] [I-D.ietf-mmusic-sdp-simulcast]
Lennox, J. and H. Schulzrinne, "Mechanisms for Media Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
Source Selection in the Session Description Protocol "Using Simulcast in SDP and RTP Sessions", draft-ietf-
(SDP)", draft-lennox-mmusic-sdp-source-selection-05 (work mmusic-sdp-simulcast-13 (work in progress), June 2018.
in progress), October 2012.
[I-D.ietf-perc-srtp-ekt-diet]
Jennings, C., Mattsson, J., McGrew, D., Wing, D., and F.
Andreasen, "Encrypted Key Transport for DTLS and Secure
RTP", draft-ietf-perc-srtp-ekt-diet-07 (work in progress),
March 2018.
[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,
DOI 10.17487/RFC2198, September 1997, DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>. <https://www.rfc-editor.org/info/rfc2198>.
[RFC2205] Braden, R., Ed., 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, DOI 10.17487/RFC2205, Functional Specification", RFC 2205, DOI 10.17487/RFC2205,
skipping to change at page 36, line 33 skipping to change at page 36, line 19
[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,
DOI 10.17487/RFC3830, August 2004, DOI 10.17487/RFC3830, August 2004,
<https://www.rfc-editor.org/info/rfc3830>. <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, DOI 10.17487/RFC4103, June 2005, Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>. <https://www.rfc-editor.org/info/rfc4103>.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383,
DOI 10.17487/RFC4383, February 2006,
<https://www.rfc-editor.org/info/rfc4383>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566, Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/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, DOI 10.17487/RFC4568, July 2006, Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<https://www.rfc-editor.org/info/rfc4568>. <https://www.rfc-editor.org/info/rfc4568>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[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,
DOI 10.17487/RFC4588, July 2006, DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>. <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, DOI 10.17487/RFC5104, with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>. February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>. 2007, <https://www.rfc-editor.org/info/rfc5109>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<https://www.rfc-editor.org/info/rfc5245>.
[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, DOI 10.17487/RFC5576, June 2009, (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>. <https://www.rfc-editor.org/info/rfc5576>.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760,
DOI 10.17487/RFC5760, February 2010,
<https://www.rfc-editor.org/info/rfc5760>.
[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, Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010, DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>. <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, Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010, DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>. <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, Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010, DOI 10.17487/RFC5888, June 2010,
<https://www.rfc-editor.org/info/rfc5888>. <https://www.rfc-editor.org/info/rfc5888>.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
DOI 10.17487/RFC6190, May 2011,
<https://www.rfc-editor.org/info/rfc6190>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real- [RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer- time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465, to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011, DOI 10.17487/RFC6465, December 2011,
<https://www.rfc-editor.org/info/rfc6465>. <https://www.rfc-editor.org/info/rfc6465>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<https://www.rfc-editor.org/info/rfc7201>. <https://www.rfc-editor.org/info/rfc7201>.
skipping to change at page 38, line 22 skipping to change at page 38, line 26
<https://www.rfc-editor.org/info/rfc8088>. <https://www.rfc-editor.org/info/rfc8088>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, [RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session", "Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017, RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>. <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 is unsuitable for most things related to
is unsuitable. If one attempts to use Payload type multiplexing multiple RTP streams. If one attempts to use Payload type
beyond it's defined usage, that has well known negative effects on multiplexing beyond its defined usage, that has well known negative
RTP. To use Payload type as the single discriminator for multiple effects on RTP. To use payload type as the single discriminator for
streams implies that all the different media streams are being sent multiple streams implies that all the different RTP streams are being
with the same SSRC, thus using the same timestamp and sequence number sent with the same SSRC, thus using the same timestamp and sequence
space. This has many effects: number space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed 1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP media. For example, RTP streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not streams are forced to use the same RTP timestamp rate. When
possible, Payload Type multiplexing cannot be used. this is not possible, payload type multiplexing cannot be used.
2. Many RTP payload formats can fragment a media object over 2. Many RTP payload formats can fragment a media object over
multiple packets, like parts of a video frame. These payload multiple RTP packets, like parts of a video frame. These
formats need to determine the order of the fragments to payload formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all correctly decode them. Thus, it is important to ensure that all
fragments related to a frame or a similar media object are fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in by ensuring that the fragments of each RTP stream are sent in
sequence. sequence.
3. Some media formats require uninterrupted sequence number space 3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/ a repair mechanism within a single media context. The text/
T140 payload format [RFC4103] is an example of such a format. T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before function between RTP and the individual RTP stream before being
being used with Payload Type multiplexing. used with payload type multiplexing.
4. Sending multiple streams in the same sequence number space makes 4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which it impossible to determine which payload type, which stream a
stream a packet loss relates to. packet loss relates to, and thus to which stream to potentially
apply packet loss concealment or other stream-specific loss
mitigation mechanisms.
5. If RTP Retransmission [RFC4588] is used and there is a loss, it 5. If RTP Retransmission [RFC4588] is used and there is a loss, it
is possible to ask for the missing packet(s) by SSRC and is possible to ask for the missing packet(s) by SSRC and
sequence number, not by Payload Type. If only some of the sequence number, not by payload type. If only some of the
Payload Type multiplexed streams are of interest, there is no payload type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets need be requested, wasting stream(s) and all lost packets need be requested, wasting
bandwidth. bandwidth.
6. The current RTCP feedback mechanisms are built around providing 6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet feedback on RTP streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is (sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported, almost never a field to indicate which payload type is reported,
so sending feedback for a specific media stream is difficult so sending feedback for a specific RTP payload type is difficult
without extending existing RTCP reporting. without extending existing RTCP reporting.
7. The current RTCP media control messages [RFC5104] specification 7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e. is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing. would need to be redefined to support payload type multiplexing.
8. The number of payload types are inherently limited. 8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number Accordingly, using payload type multiplexing limits the number
of streams that can be multiplexed and does not scale. This of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP. blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this 9. At times, there is a need to group multiplexed streams and this
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 RTP 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 does not 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 RTP 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. Signalling considerations Appendix B. 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.
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. RTSP
RTSP [RFC7826] and SAP [RFC2974] both use SDP in a declarative [RFC7826] and SAP [RFC2974] both use SDP in a declarative fashion,
fashion, while SIP [RFC3261] uses SDP with the additional definition while SIP [RFC3261] uses SDP with the additional definition of Offer/
of Offer/Answer [RFC3264]. The impact on signalling and especially Answer [RFC3264]. The impact on signalling and especially SDP needs
SDP needs to be considered as it can greatly affect how to deploy a to be considered as it can greatly affect how to deploy a certain
certain multiplexing point choice. multiplexing point choice.
B.1.1. Session Oriented Properties B.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. RTP 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 media description
media description but those are not necessarily strictly bound to an level but those are not necessarily strictly bound to an RTP session
RTP session and could be of interest to signal specifically for a and could be of interest to signal specifically for a particular RTP
particular media stream (SSRC) within the session. The following stream (SSRC) within the session. The following properties have been
properties have been identified as being potentially useful to signal identified as being potentially useful to signal not only on RTP
not only on RTP session level: session level:
o Bitrate/Bandwidth exist today only at aggregate or a common any o Bitrate/Bandwidth exist today only at aggregate or as a common
media stream limit, unless either codec-specific bandwidth "any RTP stream" limit, unless either codec-specific bandwidth
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 RTP 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 RTP 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 RTP payload type, resolution and
mechanism has been proposed: framerate, a SSRC-linked mechanism has been proposed in
[I-D.lennox-mmusic-sdp-source-selection]. [I-D.ietf-mmusic-rid]
B.1.2. SDP Prevents Multiple Media Types B.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 and is bound to a
payload type using the rtpmap attribute. This binding has to be particular payload type using the rtpmap attribute. This binding has
loosened in order to use SDP to describe RTP sessions containing to be loosened in order to use SDP to describe RTP sessions
multiple MIME top level types. containing multiple MIME top level types.
There is an accepted WG item in the MMUSIC WG to define how multiple [I-D.ietf-mmusic-sdp-bundle-negotiation] describes how to let
media lines describe a single underlying transport multiple SDP media descriptions use a single underlying transport in
[I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible SDP, which allows to define one RTP session with media types having
in SDP to define one RTP session with media types having different different MIME top level types.
MIME top level types.
B.1.3. Signalling Media Stream Usage B.3. Signalling RTP stream Usage
Media streams being transported in RTP has some particular usage in RTP streams being transported in RTP has some particular usage in an
an RTP application. This usage of the media stream is in many RTP application. This usage of the RTP 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 RTP 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 RTP 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 RTP streams affected by their RTP-
level identifiers, which means that they have to be identified either level identifiers, which means that they have to be identified either
by their session or by their SSRC + session. by their session or by their SSRC + session.
In some applications, the receiver cannot utilise the media stream at In some applications, the receiver cannot utilise the RTP stream at
all before it has received the signalling message describing the all before it has received the signalling message describing the RTP
media stream and its usage. In other applications, there exists a stream and its usage. In other applications, there exists a default
default handling that is appropriate. handling that is appropriate.
If all media streams in an RTP session are to be treated in the same If all RTP streams in an RTP session are to be treated in the same
way, identifying the session is enough. If SSRCs in a session are to way, identifying the session is enough. If SSRCs in a session are to
be treated differently, signalling needs to identify both the session be treated differently, signalling needs to identify both the session
and the SSRC. and the SSRC.
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 RTP streams with different usage in the right
fashion. fashion.
Authors' Addresses Authors' Addresses
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Torshamsgatan 23 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 Gronlandsgatan 31
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
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 Roni Even
Huawei Huawei
Email: roni.even@huawei.com Email: roni.even@huawei.com
Hui Zheng
Huawei
Email: marvin.zhenghui@huawei.com
 End of changes. 261 change blocks. 
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