draft-ietf-avtcore-multiplex-guidelines-08.txt   draft-ietf-avtcore-multiplex-guidelines-09.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: June 17, 2019 C. Perkins Expires: January 23, 2020 C. Perkins
University of Glasgow University of Glasgow
H. Alvestrand H. Alvestrand
Google Google
R. Even R. Even
Huawei Huawei
December 14, 2018 July 22, 2019
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-08 draft-ietf-avtcore-multiplex-guidelines-09
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
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/. Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on June 17, 2019. This Internet-Draft will expire on January 23, 2020.
Copyright Notice Copyright Notice
Copyright (c) 2018 IETF Trust and the persons identified as the Copyright (c) 2019 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5 3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5
3.1. Reasons for Multiplexing and Grouping RTP 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 . . . . . . . . . 12 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 . . . . . . . . . . . . . 14
3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 15 3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 14
3.4.4. Forward Error Correction . . . . . . . . . . . . . . 17 3.4.4. Forward Error Correction . . . . . . . . . . . . . . 16
4. Considerations for RTP Multiplexing . . . . . . . . . . . . . 17 4. Considerations for RTP Multiplexing . . . . . . . . . . . . . 17
4.1. Interworking Considerations . . . . . . . . . . . . . . . 17 4.1. Interworking Considerations . . . . . . . . . . . . . . . 17
4.1.1. Application Interworking . . . . . . . . . . . . . . 17 4.1.1. Application Interworking . . . . . . . . . . . . . . 17
4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 18 4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 17
4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 18 4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 18
4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 19 4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 19
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 . . . . . . . . . . . . . 21 4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 20
4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 22 4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 22
4.3. Security and Key Management Considerations . . . . . . . 24 4.3. Security and Key Management Considerations . . . . . . . 23
4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24 4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24
4.3.2. Key Management for Multi-party sessions . . . . . . . 25 4.3.2. Key Management for Multi-party Sessions . . . . . . . 24
4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25 4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25
5. RTP Multiplexing Design Choices . . . . . . . . . . . . . . . 26 5. RTP Multiplexing Design Choices . . . . . . . . . . . . . . . 25
5.1. Multiple Media Types in one Session . . . . . . . . . . . 26 5.1. Multiple Media Types in One Session . . . . . . . . . . . 25
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. Single SSRC per Endpoint . . . . . . . . . . . . . . . . 29 5.4. Single SSRC per Endpoint . . . . . . . . . . . . . . . . 29
5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 31 5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 31
6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 31 6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 31
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 32 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 32
8. Security Considerations . . . . . . . . . . . . . . . . . . . 33 8. Security Considerations . . . . . . . . . . . . . . . . . . . 32
9. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 33 9. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 33
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 33 10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 33
10.1. Normative References . . . . . . . . . . . . . . . . . . 33 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
10.2. Informative References . . . . . . . . . . . . . . . . . 33 11.1. Normative References . . . . . . . . . . . . . . . . . . 33
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 37 11.2. Informative References . . . . . . . . . . . . . . . . . 35
Appendix B. Signalling Considerations . . . . . . . . . . . . . 39 Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 38
Appendix B. Signalling Considerations . . . . . . . . . . . . . 40
B.1. Session Oriented Properties . . . . . . . . . . . . . . . 40 B.1. Session Oriented Properties . . . . . . . . . . . . . . . 40
B.2. SDP Prevents Multiple Media Types . . . . . . . . . . . . 40 B.2. SDP Prevents Multiple Media Types . . . . . . . . . . . . 41
B.3. Signalling RTP stream Usage . . . . . . . . . . . . . . . 41 B.3. Signalling RTP Stream Usage . . . . . . . . . . . . . . . 41
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 41 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 topology participants in a communication session. It supports many topology
paradigms 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 RTP 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 RTP 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, an RTP application designer
application designer is how to best use the RTP session, the RTP needs to understand how to best use the RTP session, the RTP stream
stream identifier (SSRC), and the RTP payload type to meet the identifier (SSRC), and the RTP payload type to meet the application's
application's needs. needs.
There have been increased interest in more advanced usage of RTP. There have been increased interest in more advanced usage of RTP.
For example, multiple RTP streams can be used when a single endpoint For example, multiple RTP streams can be used when a single endpoint
has multiple media sources (like multiple cameras or microphones) has multiple media sources (like multiple cameras or microphones)
that need to be sent simultaneously. Consequently, questions are that need to be sent simultaneously. Consequently, questions are
raised regarding the most appropriate RTP usage. The limitations in raised regarding the most appropriate RTP usage. The limitations in
some implementations, RTP/RTCP extensions, and signalling has also some implementations, RTP/RTCP extensions, and signalling have also
been exposed. The authors also hope that clarification on the been exposed. The authors hope that clarification on the usefulness
usefulness of some functionalities in RTP will result in more of some functionalities in RTP will result in more complete
complete implementations in the future. 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 arising
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 provide some guidelines and recommend against some usages as
for particular purposes. being unsuitable, in general or 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 designs of RTP usage are discussed. and their impacts. Some designs of RTP usage are discussed.
Finally, some guidelines and examples are provided. Finally, some guidelines and examples are provided.
2. Definitions 2. Definitions
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In this document, it will be used to refer to situations where In this document, it will be used to refer to situations where
more than two endpoints communicate. more than two endpoints communicate.
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 Receiver: An Endpoint or Middlebox receiving RTP streams and RTP Receiver: An Endpoint or Middlebox receiving RTP streams and
RTCP messages. It uses at least one SSRC to send RTCP messages. RTCP messages. It uses at least one SSRC to send RTCP messages.
An RTP Receiver may also be an RTP sender. An RTP Receiver may also be an RTP Sender.
RTP Sender: An Endpoint sending one or more RTP streams, but also RTP Sender: An Endpoint sending one or more RTP streams, but also
sending RTCP messages. sending RTCP messages.
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; i.e. SSRCs that map to a common CNAME can be assumed 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 to have RTCP SR timing information derived from a common clock
such that they can be synchronised for playout. 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.
Note: The above definitions of RTP Receiver and RTP sender are Note: The above definitions of RTP Receiver and RTP Sender are
intended to be consistent with the usage in [RFC3550]. consistent with the usage in [RFC3550].
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 [RFC3261],
some other protocol is in use for session configuration, the Jingle [JINGLE] or some other protocol is in use for session
particular syntaxes used to define RTP session properties, or the configuration, the particular syntaxes used to define RTP session
constraints imposed by particular choices in the signalling properties, or the constraints imposed by particular choices in the
protocols, are mentioned only as examples in order to describe the signalling protocols, are mentioned only as examples in order to
RTP issues more precisely. describe the 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 applications that don't send RTCP, they do not conform to the RTP are applications that don't send RTCP, they do not conform to the RTP
specification, and thus can be regarded as reusing the RTP packet specification, and thus can be regarded as reusing the RTP 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 Streams 3.1. Reasons for Multiplexing and Grouping RTP Streams
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are not limited to the following: are not limited to the following:
o Multiple media sources o Multiple media sources
o Multiple RTP streams might be needed to represent one media source o Multiple RTP streams might be needed to represent one media source
(for instance when using layered encodings) (for instance when using layered encodings)
o A retransmission stream might repeat some parts of the content of o A retransmission stream might repeat some parts of the content of
another RTP stream another RTP stream
o An FEC stream might provide material that can be used to repair o A Forward Error Correction (FEC) stream might provide material
another RTP stream that can be used to repair another RTP stream
o Alternative encodings, for instance using different codecs for the o Alternative encodings, for instance using different codecs for the
same 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 reasons, it is necessary to decide if each For each of these reasons, it is necessary to decide if each
additional RTP stream is sent within the same RTP session as the additional RTP stream is sent within the same RTP session as the
other RTP streams, or if it is necessary to use additional RTP other RTP streams, or if it is necessary to use additional RTP
sessions to group the RTP streams. The choice suitable for one sessions to group the RTP streams. The choice suitable for one
reason, might not be the choice suitable for another reason. The reason, might not be the choice suitable for another reason. The
clearest understanding is associated with multiplexing multiple media clearest understanding is associated with multiplexing multiple media
sources of the same media type. However, all reasons warrant sources of the same media type. However, all reasons warrant
discussion and clarification on how to deal with them. As the discussion and clarification on how to deal with them. As the
discussion below will show, in reality we cannot choose a single one discussion below will show, in reality we cannot choose a single one
of SSRC or RTP session multiplexing solutions. To utilise RTP well of SSRC or RTP session multiplexing solutions for all purposes. To
and as efficiently as possible, both are needed. The real issue is utilise RTP well and as efficiently as possible, both are needed.
finding the right guidance on when to create additional RTP sessions The real issue is finding the right guidance on when to create
and when additional RTP streams in the same RTP session is the right additional RTP sessions and when additional RTP streams in the same
choice. 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 RTP streams and groups of protocol that can be used to distinguish RTP streams and groups of
RTP streams. Figure 1 outlines the process of demultiplexing RTP streams. Figure 1 outlines the process of demultiplexing
incoming RTP streams: incoming RTP streams starting already at the socket representing
reception of one or transport flows, e.g. an UDP destination port.
It also demultiplexes RTP/RTCP from any other protocols, such as STUN
[RFC5389] and DTLS-SRTP [RFC5764] on the same transport as described
in [RFC7983].
| |
| packets | packets
+-- v +-- v
| +------------+ | +------------+
| | Socket | Transport Protocol Demultiplexing | | Socket | Transport Protocol Demultiplexing
| +------------+ | +------------+
| || || | || ||
RTP | RTP/ || |+-----> SCTP ( ...and any other protocols) RTP | RTP/ || |+-----> DTLS (SRTP Keying, SCTP, etc)
Session | RTCP || +------> STUN (multiplexed using same port) Session | RTCP || +------> STUN (multiplexed using same port)
+-- || +-- ||
+-- || +-- ||
| (split by SSRC) | (split by SSRC)
| || || || | || || ||
| || || || | || || ||
RTP | +--+ +--+ +--+ RTP | +--+ +--+ +--+
Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, etc. Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, etc.
| +--+ +--+ +--+ | +--+ +--+ +--+
+-- | | | +-- | | |
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| / | | | / | |
Payload | +---+ +---+ +---+ Payload | +---+ +---+ +---+
Formats | |Dec| |Dec| |Dec| Decoders Formats | |Dec| |Dec| |Dec| Decoders
| +---+ +---+ +---+ | +---+ +---+ +---+
+-- +--
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.
RTP does not contain a session identifier, yet RTP sessions must be RTP does not contain a session identifier, yet different RTP sessions
possible to separate both across different endpoints and within a must be possible to identify both across different endpoints and
single endpoint. within a single endpoint.
For RTP session separation across endpoints, the set of participants For RTP session separation across endpoints, the set of participants
that form an RTP session is defined as those that share a single that form an RTP session is defined as those that share a single
synchronisation source space [RFC3550]. That is, if a group of synchronisation source space [RFC3550]. That is, if a group of
participants are each aware of the synchronisation source identifiers participants are each aware of the synchronisation source identifiers
belonging to the other participants, then those participants are in a belonging to the other participants, then those participants are in a
single RTP session. A participant can become aware of a single RTP session. A participant can become aware of a
synchronisation source identifier by receiving an RTP packet synchronisation source identifier by receiving an RTP packet
containing it in the SSRC field or CSRC list, by receiving an RTCP 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., packet mentioning it in an SSRC field, or through signalling (e.g.,
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[RFC5576]). Thus, the scope of an RTP session is determined by the [RFC5576]). Thus, the scope of an RTP session is determined by the
participants' network interconnection topology, in combination with participants' network interconnection topology, in combination with
RTP and RTCP forwarding strategies deployed by the endpoints and any RTP and RTCP forwarding strategies deployed by the endpoints and any
middleboxes, and by the signalling. middleboxes, and by the signalling.
For RTP session separation within a single endpoint, RTP relies on For RTP session separation within a single endpoint, RTP relies on
the underlying transport layer, and on the signalling to identify RTP the underlying transport layer, and on the signalling to identify RTP
sessions in a manner that is meaningful to the application. A single sessions in a manner that is meaningful to the application. A single
endpoint can have one or more transport flows for the same RTP endpoint can have one or more transport flows for the same RTP
session, and a single RTP session can span multiple transport layer session, and a single RTP session can therefore span multiple
flows. The signalling layer might give RTP sessions an explicit transport layer flows even if all endpoints use a single transport
identifier, or the identification might be implicit based on the layer flow per endpoint for that RTP session. The signalling layer
addresses and ports used. Accordingly, a single RTP session can have might give RTP sessions an explicit identifier, or the identification
multiple associated identifiers, explicit and implicit, belonging to might be implicit based on the addresses and ports used.
different contexts. For example, when running RTP on top of UDP/IP, Accordingly, a single RTP session can have multiple associated
an endpoint can identify and delimit an RTP session from other RTP identifiers, explicit and implicit, belonging to different contexts.
sessions by receiving the multiple UDP flows used as identified based For example, when running RTP on top of UDP/IP, an endpoint can
on their UDP source and destination IP addresses and UDP port identify and delimit an RTP session from other RTP sessions by their
numbers. Another example is SDP media descriptions (the "m=" line UDP source and destination IP addresses and UDP port numbers.
and the following associated lines) signals the transport flow and Independently if an endpoint has one or more IP addresses, a single
RTP session configuration for the endpoints part of the RTP session. RTP session can be using multiple IP/UDP flows for receiving and/or
SDP grouping framework [RFC5888] allows labeling of the media sending RTP packets to other endpoints or middleboxes. Another
descriptions, for example used so that RTP Session Groups can be example is SDP media descriptions (the "m=" line and the following
created. With Negotiating Media Multiplexing Using the Session associated lines) that signals the transport flow and RTP session
Description Protocol (SDP)[I-D.ietf-mmusic-sdp-bundle-negotiation], configuration for the endpoint's part of the RTP session. The SDP
multiple media descriptions where each represents the RTP streams grouping framework [RFC5888] allows labeling of the media
sent or received for a media source are part of a common RTP session. descriptions to be used so that RTP Session Groups can be created.
Through use of Negotiating Media Multiplexing Using the Session
Description Protocol (SDP) [I-D.ietf-mmusic-sdp-bundle-negotiation],
multiple media descriptions become part of a common RTP session where
each media description represents the RTP streams sent or received
for a media source.
The RTP protocol makes no normative statements about the relationship The RTP protocol makes no normative statements about the relationship
between different RTP sessions, however the applications that use between different RTP sessions, however the applications that use
more than one RTP session will have some higher layer understanding more than one RTP session will have some higher layer understanding
of the relationship between the sessions they create. 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 source of an RTP stream A synchronisation source (SSRC) identifies a source of an RTP stream,
or an RTP receiver when sending RTCP. Every endpoint has at least or an RTP receiver when sending RTCP. Every endpoint has at least
one SSRC identifier, even if it does not send RTP packets. RTP one SSRC identifier, even if it does not send RTP packets. RTP
endpoints that are only RTP receivers still send RTCP and use their endpoints that are only RTP receivers still send RTCP and use their
SSRC identifiers in the RTCP packets they send. An endpoint can have SSRC identifiers in the RTCP packets they send. An endpoint can have
multiple SSRC identifiers if it sends multiple RTP streams. multiple SSRC identifiers if it sends multiple RTP streams.
Endpoints that are both RTP sender and RTP receiver use the same SSRC Endpoints that are both RTP sender and RTP receiver use the same SSRC
in both roles. in both roles.
The SSRC is a 32-bit identifier. It is present in every RTP and RTCP The SSRC is a 32-bit identifier. It is present in every RTP and RTCP
packet header, and in the payload of some RTCP packet types. It can packet header, and in the payload of some RTCP packet types. It can
skipping to change at page 9, line 4 skipping to change at page 9, line 9
endpoints that are only RTP receivers still send RTCP and use their endpoints that are only RTP receivers still send RTCP and use their
SSRC identifiers in the RTCP packets they send. An endpoint can have SSRC identifiers in the RTCP packets they send. An endpoint can have
multiple SSRC identifiers if it sends multiple RTP streams. multiple SSRC identifiers if it sends multiple RTP streams.
Endpoints that are both RTP sender and RTP receiver use the same SSRC Endpoints that are both RTP sender and RTP receiver use the same SSRC
in both roles. in both roles.
The SSRC is a 32-bit identifier. It is present in every RTP and RTCP The SSRC is a 32-bit identifier. It is present in every RTP and RTCP
packet header, and in the payload of some RTCP packet types. It can packet header, and in the payload of some RTCP packet types. It can
also be present in SDP signalling. Unless pre-signalled, e.g. using also be present in SDP signalling. Unless pre-signalled, e.g. using
the SDP "a=ssrc:" attribute [RFC5576], the SSRC is chosen at random. the SDP "a=ssrc:" attribute [RFC5576], the SSRC is chosen at random.
It is not dependent on the network address of the endpoint, and is It is not dependent on the network address of the endpoint, and is
intended to be unique within an RTP session. SSRC collisions can intended to be unique within an RTP session. SSRC collisions can
occur, and are handled as specified in [RFC3550] and [RFC5576], occur, and are handled as specified in [RFC3550] and [RFC5576],
resulting in the SSRC of the colliding RTP streams or receivers resulting in the SSRC of the colliding RTP streams or receivers
changing. An endpoint that changes its network transport address changing. An endpoint that changes its network transport address
during a session have to choose a new SSRC identifier to avoid being during a session has to choose a new SSRC identifier to avoid being
interpreted as looped source, unless the transport layer mechanism, interpreted as looped source, unless the transport layer mechanism,
e.g ICE [RFC8445], handles such changes. e.g ICE [RFC8445], handles such changes.
SSRC identifiers that belong to the same synchronisation context SSRC identifiers that belong to the same synchronisation context
(i.e., that represent RTP streams that can be synchronised using (i.e., that represent RTP streams that can be synchronised using
information in RTCP SR packets) use identical CNAME chunks in information in RTCP SR packets) use identical CNAME chunks in
corresponding RTCP SDES packets. SDP signalling can also be used to corresponding RTCP SDES packets. SDP signalling can also be used to
provide explicit SSRC grouping [RFC5576]. provide explicit SSRC grouping [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 RTP retransmission streams in two different RTP sessions, such as in RTP retransmission
[RFC4588]. This is to be avoided since there is no guarantee that [RFC4588]. This is to be avoided since there is no guarantee that
SSRC values are unique across RTP sessions. For the RTP SSRC values are unique across RTP sessions. For the RTP
retransmission [RFC4588] case it is recommended to use explicit retransmission [RFC4588] case it is recommended to use explicit
binding of the source RTP stream and the redundancy stream, e.g. binding of the source RTP stream and the redundancy stream, e.g.
using the RepairedRtpStreamId RTCP SDES item [I-D.ietf-avtext-rid]. 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
SSRC and thus specific per RTP stream. SSRC and thus specific per RTP stream.
Different types of entities use a SSRC to identify themselves, as Different types of entities use an SSRC to identify themselves, as
follows: follows:
A real media source: Uses the SSRC to identify a "physical" media A real media source: Uses the SSRC to identify a "physical" media
source. source.
A conceptual media source: Uses the SSRC to identify the result of A conceptual media source: Uses the SSRC to identify the result of
applying some filtering function in a network node, for example a applying some filtering function in a network node, for example a
filtering function in an RTP mixer that provides the most active filtering function in an RTP mixer that provides the most active
speaker based on some criteria, or a mix representing a set of speaker based on some criteria, or a mix representing a set of
other sources. other sources.
An RTP receiver: Uses the SSRC to identify itself as the source of An RTP receiver: Uses the SSRC to identify itself as the source of
its RTCP reports. its RTCP reports.
Note that an endpoint that generates more than one media type, e.g. 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,
should not) use the same SSRC value across RTP sessions. RTCP indeed, should not) use the same SSRC value across RTP sessions.
compound packets containing the CNAME SDES item is the designated
method to bind an SSRC to a CNAME, effectively cross-correlating RTCP compound packets containing the CNAME SDES item is the
SSRCs within and between RTP Sessions as coming from the same designated method to bind an SSRC to a CNAME, effectively cross-
endpoint. The main property attributed to SSRCs associated with the correlating SSRCs within and between RTP Sessions as coming from the
same CNAME is that they are from a particular synchronisation context same endpoint. The main property attributed to SSRCs associated with
and can be synchronised at playback. the same CNAME is that they are from a particular synchronisation
context 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, making the new SSRC 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
an SSRC identifier is listed as a CSRC in the RTP header of a packet an SSRC identifier is listed as a CSRC in the RTP header of a packet
generated by an RTP mixer, if the corresponding SSRC was in the generated by an RTP mixer, if the corresponding SSRC was in the
header of one of the packets that contributed to the mix. header of one of the packets that contributed to the 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
skipping to change at page 10, line 38 skipping to change at page 10, line 42
Mixer-to-Client Audio Level Indication [RFC6465] expands on the Mixer-to-Client Audio Level Indication [RFC6465] expands on the
receiver's 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 RTP 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 is
identified by the payload type (PT) field in the RTP packet header. identified by the payload type (PT) field in the RTP packet header.
The combination of SSRC and PT therefore identifies a specific RTP The combination of SSRC and PT therefore identifies a specific RTP
stream encoding format. The format definition can be taken from stream in a specific encoding format. The format definition can be
[RFC3551] for statically allocated payload types, but ought to be taken from [RFC3551] for statically allocated payload types, but
explicitly defined in signalling, such as SDP, both for static and ought to be explicitly defined in signalling, such as SDP, both for
dynamic payload types. The term "format" here includes whatever can static and dynamic payload types. The term "format" here includes
be described by out-of-band signalling means. In SDP, the term those aspects described by out-of-band signalling means; in SDP, the
"format" includes media type, RTP timestamp sampling rate, codec, term "format" includes media type, RTP timestamp sampling rate,
codec configuration, payload format configurations, and various codec, codec configuration, payload format configurations, and
robustness mechanisms such as redundant encodings [RFC2198]. various robustness mechanisms such as redundant encodings [RFC2198].
The RTP payload type is scoped by the sending endpoint within an RTP The RTP payload type is scoped by the sending endpoint within an RTP
session. PT has the same meaning across all RTP streams in an RTP session. PT has the same meaning across all RTP streams in an RTP
session. All SSRCs sent from a single endpoint share the same session. All SSRCs sent from a single endpoint share the same
payload type definitions. The RTP payload type is designed such that payload type definitions. The RTP payload type is designed such that
only a single payload type is valid at any time instant in the RTP only a single payload type is valid at any time instant in the RTP
stream's timestamp time line, effectively time-multiplexing different stream's timestamp time line, effectively time-multiplexing different
payload types if any change occurs. The payload type used can change payload types if any change occurs. The payload type can change on a
on a per-packet basis for an SSRC, for example a speech codec making per-packet basis for an SSRC, for example a speech codec making use
use of generic comfort noise [RFC3389]. If there is a true need to of generic comfort noise [RFC3389]. If there is a true need to send
send multiple payload types for the same SSRC that are valid for the multiple payload types for the same SSRC that are valid for the same
same instant, then redundant encodings [RFC2198] can be used. instant, then redundant encodings [RFC2198] can be used. Several
Several additional constraints than the ones mentioned above need to additional constraints than the ones mentioned above need to be met
be met to enable this use, one of which is that the combined payload to enable this use, one of which is that the combined payload sizes
sizes of the different payload types ought not exceed the transport of the different payload types ought not exceed the transport MTU.
MTU. If it is acceptable to send multiple formats of the same media If it is acceptable to send multiple formats of the same media source
source as separate RTP streams (with separate SSRC), simulcast as separate RTP streams (with separate SSRC), simulcast
[I-D.ietf-mmusic-sdp-simulcast] can be used. [I-D.ietf-mmusic-sdp-simulcast] can be used.
Other aspects of RTP payload format use are described in How to Write Other aspects of RTP payload format use are described in How to Write
an RTP Payload Format [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 consume and decode an RTP stream. however, used to determine how to consume and decode an RTP stream.
The RTP payload type number is sometimes used to associate an RTP The RTP payload type number is sometimes used to associate an RTP
skipping to change at page 12, line 32 skipping to change at page 12, line 32
o Usage of different transport protocols, e.g., UDP, DCCP, or TCP. o Usage of different transport protocols, e.g., UDP, DCCP, or TCP.
o Different security solutions, e.g., IPsec, TLS, DTLS, or SRTP with o Different security solutions, e.g., IPsec, TLS, DTLS, or 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 peers look [RFC7667]. The translator's main purpose is to make the peers look
compatible to each other. There can also be other reasons than compatible to each other. There can also be other reasons than
compatibility to insert a translator in the form of a middlebox or compatibility to insert a translator in the form of a middlebox or
gateway, for example a need to monitor the RTP streams. If the gateway, for example a need to monitor the RTP streams. Beware that
stream transport characteristics are changed by the translator, changing the stream transport characteristics in the translator, can
appropriate media handling can require thorough understanding of the require thorough understanding of the application logic, specifically
application logic, specifically any congestion control or media any congestion control or media adaptation to ensure appropriate
adaptation. media handling.
The point to point topology can contain one to many RTP sessions with Within the uses enabled by the RTP standard the point to point
one to many media sources per session, each having one or more RTP topology can contain one to many RTP sessions with one to many media
streams per media source. sources per session, each having one or more RTP streams per media
source.
3.4. Issues Related to RTP and RTCP Protocol 3.4. Issues Related to RTP and RTCP Protocol
Using multiple RTP streams is a well-supported feature of RTP. Using multiple RTP streams is a well-supported feature of RTP.
However, for most implementers or people writing RTP/RTCP However, for most implementers or people writing RTP/RTCP
applications or extensions attempting to apply multiple streams, it applications or extensions attempting to apply multiple streams, it
can be unclear when it is most appropriate to add an additional RTP can be unclear when it is most appropriate to add an additional RTP
stream in an existing RTP session and when it is better to use stream in an existing RTP session and when it is better to use
multiple RTP sessions. This section discusses the various multiple RTP sessions. This section discusses the various
considerations 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. Please review
Section 5.2 of [RFC3550], reproduced below: Section 5.2 of [RFC3550]. Five important aspects are quoted below.
"For efficient protocol processing, the number of multiplexing points
should be minimised, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference
composed of audio and video media encoded separately, each medium
SHOULD be carried in a separate RTP session with its own destination
transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings.
2. An SSRC is defined to identify a single timing and sequence
number space. Interleaving multiple payload types would require
different timing spaces if the media clock rates differ and would
require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
Using a different SSRC for each medium but sending them in the same 1. If, say, two audio streams shared the same RTP session and the
RTP session would avoid the first three problems but not the last same SSRC value, and one were to change encodings and thus acquire
two. a different RTP payload type, there would be no general way of
identifying which stream had changed encodings.
On the other hand, multiplexing multiple related sources of the same The first argument is to use different SSRC for each individual RTP
medium in one RTP session using different SSRC values is the norm for stream, which is fundamental to RTP operation.
multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It might also be
appropriate to multiplex streams of the same medium using different
SSRC values in other scenarios where the last two problems do not
apply."
Let's consider one argument at a time. The first argument is for 2. An SSRC is defined to identify a single timing and sequence number
using different SSRC for each individual RTP stream, which is space. Interleaving multiple payload types would require
fundamental to RTP operation. different timing spaces if the media clock rates differ and would
require different sequence number spaces to tell which payload
type suffered packet loss.
The second argument is advocating against demultiplexing RTP streams The second argument is advocating against demultiplexing RTP streams
within a session based on their RTP payload type numbers, which still within a session based only on their RTP payload type numbers, which
stands as can been seen by the extensive list of issues found in still stands as can been seen by the extensive list of issues found
Appendix A. in Appendix A.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
The third argument is yet another argument against payload type The third argument is yet another argument against payload type
multiplexing. multiplexing.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
The fourth argument is against multiplexing RTP packets that require The fourth argument is against multiplexing RTP packets that require
different handling into the same session. As we saw in the different handling into the same session. In most cases the RTP
discussion of RTP mixers, the RTP mixer must embed application logic mixer must embed application logic to handle streams; the separation
to handle streams anyway; the separation of streams according to of streams according to stream type is just another piece of
stream type is just another piece of application logic, which might application logic, which might or might not be appropriate for a
or might not be appropriate for a particular application. One type particular application. One type of application that can mix
of application that can mix different media sources "blindly" is the different media sources blindly is the audio-only telephone bridge,
audio-only "telephone" bridge; most other types of applications need although the ability to do that comes from the well-defined scenario
application-specific logic to perform the mix correctly. that is aided by use of a single media type, even though individual
streams may use incompatible codec types; most other types of
applications need application-specific logic to perform the mix
correctly.
The fifth argument discusses network aspects that we will discuss 5. Carrying multiple media in one RTP session precludes: the use of
more below in Section 4.2. It also goes into aspects of different network paths or network resource allocations if
implementation, like Split Component Terminal (see Section 3.10 of appropriate; reception of a subset of the media if desired, for
[RFC7667]) endpoints where different processes or inter-connected example just audio if video would exceed the available bandwidth;
devices handle different aspects of the whole multi-media session. and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
The fifth argument discusses network aspects that are described in
Section 4.2. It also goes into aspects of implementation, like Split
Component Terminal (see Section 3.10 of [RFC7667]) endpoints where
different processes or inter-connected 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
latter needs further discussion, as imposing a single solution on all latter needs further discussion, as imposing a single solution on all
usages of RTP is inappropriate. Multiple Media Types in an RTP usages of RTP is inappropriate. Multiple Media Types in an RTP
Session specification [I-D.ietf-avtcore-multi-media-rtp-session] Session specification [I-D.ietf-avtcore-multi-media-rtp-session]
provides a detailed analysis of the potential issues in having provides a detailed analysis of the potential issues in having
multiple media types in the same RTP session. This document provides multiple media types in the same RTP session. This document provides
skipping to change at page 15, line 47 skipping to change at page 15, line 20
o Grouping SSRCs within an RTP session o Grouping SSRCs within an RTP session
Most solutions are explicit, but some implicit methods have also been Most solutions are explicit, but some implicit methods have also been
applied to the problem. 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 This has previously been considered primarily as grouping RTP
sessions, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple sessions, but [I-D.ietf-mmusic-sdp-bundle-negotiation] groups
media descriptions as a single RTP session. multiple 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.
This supports a lot of use cases. All these solutions have The above grouping constructs support many use cases. Those
shortcomings in cases where the session's dynamic properties are such solutions have shortcomings in cases where the session's dynamic
that it is difficult or resource consuming to keep the list of properties are such that it is difficult or a drain on resources to
related SSRCs up to date. keep the list of related SSRCs up to date.
An RTP/RTCP-based solution is to use the RTCP SDES CNAME to bind the An RTP/RTCP-based grouping solution is to use the RTCP SDES CNAME to
RTP streams to an endpoint or synchronization context. For bind related RTP streams to an endpoint or to a synchronization
applications with a single RTP stream per type (Media, Source or context. For applications with a single RTP stream per type (media,
Redundancy) this is sufficient independent if one or more RTP source or redundancy stream), CNAME is sufficient for that purpose
sessions are used. However, some applications choose not to use it independent if one or more RTP sessions are used. However, some
because of perceived complexity or a desire not to implement RTCP and applications choose not to use CNAME because of perceived complexity
instead use the same SSRC value to bind related RTP streams across or a desire not to implement RTCP and instead use the same SSRC value
multiple RTP sessions. RTP Retransmission [RFC4588] in multiple RTP to bind related RTP streams across multiple RTP sessions. RTP
session mode and Generic FEC [RFC5109] both use this method. This Retransmission [RFC4588] in multiple RTP session mode and Generic FEC
method may work but might have some downsides in RTP sessions with [RFC5109] both use the CNAME method to relate the RTP streams, which
many participating SSRCs. When an SSRC collision occurs, this will may work but might have some downsides in RTP sessions with many
force one to change SSRC in all RTP sessions and thus resynchronize participating SSRCs. It is not recommended to use identical SSRC
all of them instead of only the single media stream having the values across RTP sessions to relate RTP streams; When an SSRC
collision. Therefore, it is not recommended to use identical SSRC collision occurs, this will force change of that SSRC in all RTP
values to relate RTP streams. sessions and thus resynchronize all of them instead of only the
single media stream having the collision.
Another solution to bind SSRCs is an implicit method used by RTP Another method to implicitly bind SSRCs is used by RTP Retransmission
Retransmission [RFC4588] when doing retransmissions in the same RTP [RFC4588] when using the same RTP session as the source RTP stream
session as the source RTP stream. The receiver missing a packet for retransmissions. The receiver missing a packet issues an RTP
issues an RTP retransmission request, and then awaits a new SSRC retransmission request, and then awaits a new SSRC carrying the RTP
carrying the RTP retransmission payload and where that SSRC is from retransmission payload and where that SSRC is from the same CNAME.
the same CNAME. This limits a requester to having only one This limits a requester to having only one outstanding retransmission
outstanding request on any new source SSRCs per endpoint. request on any new source SSRCs per endpoint.
RTP Payload Format Restrictions [I-D.ietf-mmusic-rid] provides an RTP Payload Format Restrictions [I-D.ietf-mmusic-rid] provides an
RTP/RTCP based mechanism to unambiguously identify the RTP streams RTP/RTCP based mechanism to unambiguously identify the RTP streams
within an RTP session and restrict the streams' payload format within an RTP session and restrict the streams' payload format
parameters in a codec-agnostic way beyond what is provided with the parameters in a codec-agnostic way beyond what is provided with the
regular Payload Types. The mapping is done by specifying an "a=rid" regular payload types. The mapping is done by specifying an "a=rid"
value in the SDP offer/answer signalling and having the corresponding value in the SDP offer/answer signalling and having the corresponding
"rtp-stream-id" value as an SDES item and an RTP header extension. RtpStreamId value as an SDES item and an RTP header extension. The
The RID solution also includes a solution for binding redundancy RTP RID solution also includes a solution for binding redundancy RTP
streams to their original source RTP streams, given that those use streams to their original source RTP streams, given that those use
RID identifiers. RID identifiers.
It can be noted that Section 8.3 of the RTP Specification [RFC3550] Section 8.3 of the RTP Specification [RFC3550] recommends using a
recommends using a single SSRC space across all RTP sessions for single SSRC space across all RTP sessions for layered coding. Based
layered coding. Based on the experience so far however, we recommend on the experience so far however, we recommend to use a solution with
to use a solution doing explicit binding between the RTP streams so explicit binding between the RTP streams that is agnostic to the used
what the used SSRC values are do not matter. That way solutions SSRC values. That way, solutions using multiple RTP streams in a
using multiple RTP streams in a single RTP session and multiple RTP single RTP session and in multiple RTP sessions will use the same
sessions uses the same solution. type of binding.
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 protects 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 the redundant information protects. This over the set of packets the redundant information protects. This
sequence of redundant information also needs to be transmitted as its sequence of redundant information needs to be transmitted as its own
own media stream, or in some cases, instead of the original media media stream, or in some cases, instead of the original media stream.
stream. Thus, many of these schemes create a need for binding Thus, many of these schemes create a need for binding related flows
related flows as discussed above. Looking at the history of these as discussed above. Looking at the history of these schemes, there
schemes, there are schemes using multiple SSRCs and schemes using are schemes using multiple SSRCs and schemes using multiple RTP
multiple RTP sessions, and some schemes that support both modes of sessions, and some schemes that support both modes of operation.
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 and if separating RTP sessions placing it in a separate RTP session and if separating RTP sessions
on transport level, FEC can easily be ignored already on transport on transport level, FEC can easily be ignored already on transport
level. level, without considering any RTP layer information.
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 similar flexibility. This is especially multicast group allows for similar flexibility. This is especially
useful when receivers see very heterogeneous packet loss rates. useful when receivers see heterogeneous packet loss rates. A
Those receivers that are not seeing packet loss don't need to join receiver can based on measurment of experienced packet loss decide to
the multicast group with the FEC data, and so avoid the overhead of join a multicast group with the suitable FEC data repair
receiving unnecessary FEC packets, for example. capabilities.
4. 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; interworking between different applications including discusses two; interworking directly between different applications,
the implications of potentially different RTP multiplexing point and interworking of applications through an RTP Translator. The
choices and limitations that have to be considered when working with discussion includes the implications of potentially different RTP
some legacy applications. multiplexing point choices and limitations that have to be considered
when working with some legacy applications.
4.1.1. Application Interworking 4.1.1. Application Interworking
It is not uncommon that applications or services of similar but not It is not uncommon that applications or services of similar but not
identical usage, especially the ones intended for interactive identical usage, especially the ones intended for interactive
communication, encounter a situation where one want to interconnect communication, encounter a situation where one want to interconnect
two or more of these 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 must then either gateway to interconnect applications. This gateway must then either
change the multiplexing structure or adhere to the respective change the multiplexing structure or adhere to the respective
limitations in each application. limitations in each application.
There are two fundamental approaches to building a gateway: using an There are two fundamental approaches to building a gateway: using RTP
RTP Translator interworking (RTP bridging), where the gateway acts as Translator interworking (RTP bridging), where the gateway acts as an
an RTP Translator, with the two applications being members of the RTP Translator with the two interconnected applications being members
same RTP session; or Gateway Interworking with RTP termination, where of the same RTP session; or using Gateway Interworking with RTP
there are independent RTP sessions running from each interconnected termination, where there are independent RTP sessions between each
application to the gateway. interconnected application and 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 From an RTP perspective, the RTP Translator approach could work if
all the applications are using the same codecs with the same payload all the applications are using the same codecs with the same payload
types, have made the same multiplexing choices, and have the same types, have made the same multiplexing choices, and have the same
capabilities in number of simultaneous RTP streams combined with the capabilities in number of simultaneous RTP streams combined with the
same set of RTP/RTCP extensions being supported. Unfortunately, this same set of RTP/RTCP extensions being supported. Unfortunately, this
might not always be true. might not always be true.
When a gateway is implemented via an RTP Translator, an important When a gateway is implemented via an RTP Translator, an important
consideration is if the two applications being interconnected need to consideration is if the two applications being interconnected need to
use the same approach to multiplexing. If one side is using RTP use the same approach to multiplexing. If one side is using RTP
session multiplexing and the other is using SSRC multiplexing with session multiplexing and the other is using SSRC multiplexing with
bundle, it is possible for the RTP translator to map the RTP streams BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], it is possible for
between both sides if the order of SDP "m=" lines between both sides the RTP translator to map the RTP streams between both sides using
are the same. There are also challenges with SSRC collision handling some method, e.g. if the number and order of SDP "m=" lines between
since there may be a collision on the SSRC multiplexing side but the both sides are the same. There are also challenges with SSRC
RTP session multiplexing side will not be aware of any collision collision handling since, unless SSRC translation is applied on the
unless SSRC translation is applied on the RTP translator. RTP translator, there may be a collision on the SSRC multiplexing
side that the RTP session multiplexing side will not be aware of.
Furthermore, if one of the applications is capable of working in Furthermore, if one of the applications is capable of working in
several modes (such as being able to use additional RTP streams 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 one RTP session or multiple RTP sessions at will), and the other one
is not, successful interconnection depends on locking the more is not, successful interconnection depends on locking the more
flexible application into the operating mode where interconnection flexible application into the operating mode where interconnection
can be successful, even if no participants are using the less can be successful, even if none of the participants are using the
flexible application when the RTP sessions are being created. less flexible application when the RTP sessions are being 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 RTP streams (possibly o Generating appropriate RTCP reports for all RTP streams (possibly
based on incoming RTCP reports), originating from SSRCs controlled based on incoming RTCP reports), originating from SSRCs 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 use any security mechanism, e.g., in the form
of SRTP, the gateway needs to be able to decrypt incoming packets and of SRTP, the gateway needs to be able to decrypt and verify source
re-encrypt them in the other application's security context. This is integrity of the incoming packets, and re-encrypt, integrity protect,
necessary even if all that's needed is a simple remapping of SSRC and sign the packets as peer in the other application's security
numbers. If this is done, the gateway also needs to be a member of context. This is necessary even if all that's needed is a simple
the security contexts of both sides, of course. remapping of SSRC numbers. If this is done, the gateway also needs
to be a member of the security contexts of both sides, of course.
Other tasks a gateway might need to apply include transcoding (for The gateway might also need to apply transcoding (for incompatible
incompatible codec types), media-level adaptations that cannot be codec types), media-level adaptations that cannot be solved through
solved through media negotiation (such as rescaling for incompatible media negotiation (such as rescaling for incompatible video size
video size requirements), suppression of content that is known not to requirements), suppression of content that is known not to be handled
be handled in the destination application, or the addition or removal in the destination application, or the addition or removal of
of redundancy coding or scalability layers to fit the needs of the 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 These gateways might therefore potentially block application
application evolution by blocking RTP and RTCP extensions that the evolution by blocking RTP and RTCP extensions that the applications
applications have been extended with but that are unknown to the have been extended with but that are unknown to the gateway.
gateway.
If one uses security functions, like SRTP, and as can be seen from If one uses security functions, like SRTP, and as can be seen from
above, they incur both additional risk due to the requirement to have above, they incur both additional risk due to the requirement to have
the gateway in the security association between the endpoints (unless the gateway in the security association between the endpoints (unless
the gateway is on the transport level), and additional complexities the gateway is on the transport level), and additional complexities
in form of the decrypt-encrypt cycles needed for each forwarded in form of the decrypt-encrypt cycles needed for each forwarded
packet. SRTP, due to its keying structure, also requires that each packet. SRTP, due to its keying structure, also requires that each
RTP session needs different master keys, as use of the same key in RTP session needs different master keys, as use of the same key in
two RTP sessions can for some ciphers result in two-time pads that two RTP sessions can for some ciphers result in two-time pads that
completely breaks the 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 or than one media source per endpoint and media type (typically audio or
video). Even in conferencing applications, especially voice-only, video). Even in conferencing applications, especially voice-only,
the conference focus or bridge has provided a single stream with a the conference focus or bridge has provided a single stream to each
mix of the other participants to each participant. It is also common participant containing a mix of the other participants. It is also
to have individual RTP sessions between each endpoint and the RTP common to have individual RTP sessions between each endpoint and the
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 Endpoints that aren't updated to handle multiple streams following
updated to handle multiple streams following these recommendations, a these recommendations can have issues with participating in RTP
particular application can have issues with multiple SSRCs within a sessions containing multiple SSRCs within a single session, such as:
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 RTP stream of each type of devices have to make sure that only one RTP stream of each media
gets delivered to the endpoint if it's expecting only one, and that type gets delivered to the endpoint if it's expecting only one, and
the multiplexing format is what the device expects. It is highly that the multiplexing format is what the device expects. It is
unlikely that RTP translator-based interworking can be made to highly 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 RTP multiplexing choice has impact on network level mechanisms The RTP implementer needs to consider that the RTP multiplexing
that need to be considered by the implementer. choice also impacts network level mechanisms.
4.2.1. Quality of Service 4.2.1. Quality of Service
When it comes to Quality of Service mechanisms, they are either flow Quality of Service mechanisms are either flow based or packet marking
based or packet marking based. RSVP [RFC2205] is an example of a based. RSVP [RFC2205] is an example of a flow based mechanism, while
flow based mechanism, while Diff-Serv [RFC2474] is an example of a Diff-Serv [RFC2474] is an example of a packet marking based one.
packet marking based one. For a packet marking based scheme, the
method of multiplexing will not affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between For a flow based scheme, additional SSRC will receive the same QoS as
the multiplexing methods. Additional SSRC will result in all RTP all other RTP streams being part of the same 5-tuple (protocol,
streams being part of the same 5-tuple (protocol, source address, source address, destination address, source port, destination port),
destination address, source port, destination port) which is the most which is the most common selector for flow based QoS.
common selector for flow based QoS.
It must also be noted that packet marking based QoS mechanisms can For a packet marking based scheme, the method of multiplexing will
have limitations. A general observation is that different not affect the possibility to use QoS. Different Differentiated
Differentiated Services Code Points (DSCP) can be assigned to Services Code Points (DSCP) can be assigned to different packets
different packets within a flow as well as within an RTP stream. within a flow as well as within an RTP stream. However, care must be
However, care must be taken when considering which forwarding taken when considering which forwarding behaviours that are applied
behaviours that are applied on path due to these DSCPs. In some on path due to these DSCPs. In some cases the forwarding behaviour
cases the forwarding behaviour can result in packet reordering. For can result in packet reordering. For more discussion of this see
more discussion of this see [RFC7657]. [RFC7657].
The method for assigning marking to packets can impact what number of The method for assigning marking to packets can impact what number of
RTP sessions to choose. If this marking is done using a network RTP sessions to choose. If this marking is done using a network
ingress function, it can have issues discriminating the different RTP ingress function, it can have issues discriminating the different RTP
streams. The network API on the endpoint also needs to be capable of streams. The network API on the endpoint also needs to be capable of
setting the marking on a per-packet basis to reach the full setting the marking on a per-packet basis 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 networks 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:
End-Point Port Consumption: A given IP address only has 65536 End-Point Port Consumption: A given IP address only has 65536
available local ports per transport protocol for all consumers of available local ports per transport protocol for all consumers of
ports that exist on the machine. This is normally never an issue ports that exist on the machine. This is normally never an issue
for an end-user machine. It can become an issue for servers that for an end-user machine. It can become an issue for servers that
handle large number of simultaneous streams. However, if the handle large number of simultaneous streams. However, if the
application uses ICE to authenticate STUN requests, a server can application uses ICE to authenticate STUN requests, a server can
serve multiple endpoints from the same local port, and use the serve multiple endpoints from the same local port, and use the
whole 5-tuple (source and destination address, source and whole 5-tuple (source and destination address, source and
destination port, protocol) as identifier of flows after having destination port, protocol) as identifier of flows after having
securely bound them to the remote endpoint address using the STUN securely bound them to the remote endpoint address using the STUN
request. In theory the minimum number of media server ports request. In theory, the minimum number of media server ports
needed are the maximum number of simultaneous RTP Sessions a needed are the maximum number of simultaneous RTP sessions a
single endpoint can use. In practice, implementation will single endpoint can use. In practice, implementation will
probably benefit from using more server ports to simplify probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks. implementation or avoid performance bottlenecks.
NAT State: If an endpoint sits behind a NAT, each flow it generates NAT State: If an endpoint sits behind a NAT, each flow it generates
to an external address will result in a state that has to be kept to an external address will result in a state that has to be kept
in the NAT. That state is a limited resource. In home or Small in the NAT. That state is a limited resource. In home or Small
Office/Home Office (SOHO) NATs, memory or processing are usually Office/Home Office (SOHO) NATs, memory or processing are usually
the most limited resources. For large scale NATs serving many the most limited resources. For large scale NATs serving many
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, as a comparison, a real-time video conference session
conference session with audio and video is likely using less than with audio and video likely uses less than 10 UDP flows, compared
10 UDP flows, compared to certain web applications that can use to certain web applications that can use 100+ TCP flows to various
100+ TCP flows to various servers from a single browser instance. servers from a single browser instance.
NAT Traversal Extra Delay: 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 additional NAT/FW traversal time after finding
first valid candidate pair following the specified ICE procedures the first valid candidate pair following the specified ICE
are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing procedures is 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the
timer. That assumes a message in one direction, and then an pacing timer. That assumes a message in one direction,
immediate triggered check back. The reason it isn't more, is that immediately followed by a check back. The reason it isn't more,
ICE first finds one candidate pair that works prior to attempting is that ICE first finds one candidate pair that works prior to
to establish multiple flows. Thus, there is no extra time until attempting to establish multiple flows. Thus, there is no extra
one has found a working candidate pair. Based on that working time until one has found a working candidate pair. Based on that
pair the needed extra time is to in parallel establish the, in working pair, the extra time is needed to in parallel establish
most cases 2-3, additional flows. However, packet loss causes the, in most cases 2-3, additional flows. However, packet loss
extra delays, at least 100 ms, which is the minimal retransmission causes extra delays, at least 100 ms, which is the minimal
timer for ICE. retransmission timer 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 risk that deeply
deeply inspecting firewalls will have similar legacy issues with inspecting firewalls will have similar legacy issues with multiple
multiple SSRCs as some stack implementations. SSRCs as some RTP stack implementations.
Using additional RTP streams in the same RTP session and transport Using additional RTP streams in the same RTP session and transport
flow does not introduce any additional NAT traversal complexities per flow does not introduce any additional NAT traversal complexities per
RTP stream. This can be compared with normally one or two additional RTP stream. This can be compared with normally one or two additional
transport flows per RTP session when using multiple RTP sessions. transport flows per RTP session when using multiple RTP sessions.
Additional lower layer transport flows will be needed, unless an Additional lower layer transport flows will be needed, unless an
explicit de-multiplexing layer is added between RTP and the transport explicit de-multiplexing layer is added between RTP and the transport
protocol. At time of writing no such mechanism was defined. protocol. At time of writing no such mechanism was defined.
4.2.3. Multicast 4.2.3. Multicast
Multicast groups provides a powerful tool for a number of real-time Multicast groups provides a powerful tool for a number of real-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. There are also the RTP/RTCP extension to receivers, like in IPTV. There is also the RTP/RTCP extension to
better support Source Specific Multicast (SSM) [RFC5760]. Another better support Source Specific Multicast (SSM) [RFC5760]. Many-to-
application is the Many to Many communication, which RTP [RFC3550] many communication, which RTP [RFC3550] was originally built to
was originally built to support, but the multicast semantics do support, has several limitations in common with multicast.
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 with
actual receiver properties causes degradation for all participants to the intent to suit all receivers would have to adapt to the most
what is supported by the receiver with the worst conditions among the limited receiver experiencing the worst conditions among the group
group participants. For broadcast type of applications this is not participants, which imposes degradation for all participants. For
acceptable. Instead, various receiver-based solutions are employed broadcast-type applications with a large number of receivers, this is
to ensure that the receivers achieve best possible performance. By not acceptable. Instead, various receiver-based solutions are
using scalable encoding and placing each scalability layer in a employed to ensure that the receivers achieve best possible
different multicast group, the receiver can control the amount of performance. By using scalable encoding and placing each scalability
traffic it receives. To have each scalability layer on a different layer in a different multicast group, the receiver can control the
multicast group, one RTP session per multicast group is used. amount of traffic it receives. To have each scalability layer on a
different 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, when using broadcast applications it appears easiest and most Thus, when using broadcast applications it appears easiest and most
straightforward to use multiple RTP sessions for sending different straightforward to use multiple RTP sessions for sending different
media flows used for adapting to network conditions. It is also media flows used for adapting to network conditions. It is also
common that streams that improve transport robustness are sent in common that streams improving transport robustness are sent in their
their own multicast group to allow for interworking with legacy or to own multicast group to allow for interworking with legacy or to
support different levels of protection. support different levels of protection.
For many to many applications there are different needs. Here, the Many-to-many applications have different needs and the most
most appropriate choice will depend on how the actual application is appropriate multiplexing choice will depend on how the actual
realized. With sender side congestion control there might not exist application is realized. Multicast applications that are capable of
any benefit with using multiple RTP sessions. using sender side congestion control can avoid the use of multiple
multicast sessions and RTP sessions that result from use of receiver
side congestion control.
The properties of a broadcast application using RTP multicast: The properties of a broadcast application using RTP multicast:
1. Uses a group of RTP sessions, not one. Each endpoint will need 1. Uses a group of RTP sessions, not just one. Each endpoint will
to be a member of a number of RTP sessions in order to perform need to be a member of a number of RTP sessions in order to
well. perform well.
2. Within each RTP session, the number of RTP receivers is likely to 2. Within each RTP session, the number of RTP receivers is likely to
be much larger than the number of RTP senders. be much larger than the number of RTP senders.
3. The 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. The applications need signalling or RTP/RTCP functions to 4. The applications need signalling or RTP/RTCP functions to
identify the relationships between SSRCs in different RTP identify the relationships between SSRCs in different RTP
sessions when needs beyond CNAME exist. sessions when needs beyond CNAME exist.
Both broadcast and many to many multicast applications do share a Both broadcast and many-to-many multicast applications share a
signalling requirement; all of the participants will need to have the signalling requirement; all of the participants need the same RTP and
same RTP and payload type configuration. Otherwise, A could for payload type configuration. Otherwise, A could for example be using
example be using payload type 97 as the video codec H.264 while B payload type 97 as the video codec H.264 while B thinks it is MPEG-2.
thinks it is MPEG-2. It is to be noted that SDP offer/answer SDP offer/answer [RFC3264] is not appropriate for ensuring this
[RFC3264] is not appropriate for ensuring this property in broadcast/ property in broadcast/multicast context. The signalling aspects of
multicast context. The signalling aspects of broadcast/multicast are broadcast/multicast are not explored further in this memo.
not explored further in this memo.
Security solutions for this type of group communications are also Security solutions for this type of group communication are also
challenging. First, the key-management and the security protocol challenging. First, the key-management and the security protocol
need to support group communication. Second, source authentication need 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
can be a necessary differentiation in some applications. As SRTP's and can be a necessary differentiation in some applications. As
crypto suites are (so far) built around symmetric keys, the receiver SRTP's crypto suites are (so far) built around symmetric keys, the
will need to have the same key as the sender. This results in that receiver will need to have the same key as the sender. This results
no one in a multi-party session can be certain that a received packet in that no one in a multi-party session can be certain that a
really was sent by the claimed sender and not by another party having received packet really was sent by the claimed sender and not by
access to the key. At least unless TESLA source authentication another party having access to the key. The single SRTP algorithm
[RFC4383], which adds delay to achieve source authentication. In not having this propery is the TESLA source authentication [RFC4383].
most cases symmetric ciphers provide sufficient security properties, However, TESLA adds delay to achieve source authentication. In most
but there are a few cases where this does create issues. cases, symmetric ciphers provide sufficient security properties but
create issues in a few cases.
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 RTP wants to ensure that the party that left can no longer access the RTP
streams. This requires that everyone re-keys without disclosing the streams. This requires that everyone re-keys without disclosing the
keys to the excluded party. new 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 stream access differentiation between different receivers. Take for
simulcast version that only premium users are allowed to access. The example a scalable layer or a high quality simulcast version that
mechanism preventing a receiver from getting the high quality stream only premium users are allowed to access. The mechanism preventing a
can be based on the stream being encrypted with a key that user can't receiver from getting the high quality stream can be based on the
access without paying premium, having the key-management limit access stream being encrypted with a key that user can't access without
to the key. paying premium, using the key-management to limit access 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 sets 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 sessions 4.3.2. Key Management for Multi-party Sessions
Performing key-management for multi-party sessions can be a The capabilities of the key-management combined with the RTP
challenge. This section considers some of the issues. multiplexing choices affects the resulting security properties,
control over the secured media, and who have access to it.
Multi-party sessions, such as transport translator based sessions and Multi-party sessions contain at least one RTP stream from each active
multicast sessions, can neither use Security Description [RFC4568] participant. Depending on the multi-party topology [RFC7667], each
nor DTLS-SRTP [RFC5764] without an extension as each endpoint participant can both send and receive multiple RTP streams.
provides its set of keys. In centralised conferences, the signalling
counterpart is a conference server and the media plane unicast Transport translator-based sessions and multicast sessions, can
counterpart (to which DTLS messages would be sent) is the transport neither use Security Description [RFC4568] nor DTLS-SRTP [RFC5764]
translator. Thus, an extension like Encrypted Key Transport without an extension as each endpoint provides its set of keys. In
[I-D.ietf-perc-srtp-ekt-diet] or a MIKEY [RFC3830] based solution centralised conferences, the signalling counterpart is a conference
that allows for keying all session participants with the same master server, and the transport translator is the media plane unicast
key is needed. counterpart (to which DTLS messages would be sent). Thus, an
extension like Encrypted Key Transport [I-D.ietf-perc-srtp-ekt-diet]
or a MIKEY [RFC3830] based solution that allows for keying all
session participants with the same master key is needed.
Privacy Enchanced RTP Conferencing (PERC) also enables a different
trust model with semi-trusted media switching RTP middleboxes
[I-D.ietf-perc-private-media-framework].
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
from the choice of multiplexing and topology. This becomes from the choice of multiplexing and topology. This becomes
especially evident in RTP topologies having any type of middlebox especially evident in RTP topologies having any type of middlebox
that processes or modifies RTP/RTCP packets. Where there is very that processes or modifies RTP/RTCP packets. While there is very
small overhead for an RTP translator or mixer to rewrite an SSRC small overhead for an RTP translator or mixer to rewrite an SSRC
value in the RTP packet of an unencrypted session, the cost is higher value in the RTP packet of an unencrypted session, the cost is higher
when using cryptographic security functions. 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 SSRC, the determined by the SSRC field value. If one changes SSRC, the
encryption and authentication must use another key. Thus, changing encryption and authentication must use another key. Thus, changing
the SSRC value implies a decryption using the old SSRC and its the SSRC value implies a decryption using the old SSRC and its
security context, followed by an encryption using the new one. security context, followed by an encryption using the new one.
5. RTP Multiplexing Design Choices 5. RTP Multiplexing Design Choices
This section discusses how some RTP multiplexing design choices can This section discusses how some RTP multiplexing design choices can
be used in applications to achieve certain goals, and a summary of be used in applications to achieve certain goals, and a summary of
the implications of such choices. For each design there is the implications of such choices. For each design there is
discussion of benefits and downsides. discussion of benefits and downsides.
5.1. Multiple Media Types in one Session 5.1. Multiple Media Types in One Session
This design uses a single RTP session for multiple different media This design uses a single RTP session for multiple different media
types, like audio and video, and possibly also transport robustness types, like audio and video, and possibly also transport robustness
mechanisms like FEC or Retransmission. An endpoint can have zero, mechanisms like FEC or retransmission. An endpoint can send zero,
one or more media sources per media type, resulting in a number of one or more media sources per media type, resulting in a number of
RTP streams of various media types and both source and redundancy RTP streams of various media types for both source and redundancy
type. streams.
The Pros: The Advantages:
1. Single RTP session which implies: 1. Only a single RTP session is used, which implies:
* Minimal NAT/FW state. * Minimal need to keep 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. Can handle dynamic allocations of RTP streams well on an RTP * Minimal overhead for security association establishment.
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
3. Minimal overhead for security association establishment. 2. Dynamic allocation of RTP streams can be handled almost entirely
at RTP level. How localized this can be kept to RTP level
depends on the application's needs for explicit indication of the
stream usage and how timely that can be signalled.
The Cons: The Disadvantages:
a. Less suitable for interworking with other applications that uses a. It is less suitable for interworking with other applications that
individual RTP sessions per media type or multiple sessions for a use individual RTP sessions per media type or multiple sessions
single media type, due to the potential need of SSRC translation. for a single media type, due to the risk of SSRC collision and
thus potential need for SSRC translation.
b. Negotiation of bandwidth for the different media types is b. Negotiation of individual bandwidths for the different media
currently only possible using RID [I-D.ietf-mmusic-rid] in SDP. types is currently only possible in SDP when using RID
[I-D.ietf-mmusic-rid].
c. Not suitable for Split Component Terminal (see Section 3.10 of c. It is not suitable for Split Component Terminal (see Section 3.10
[RFC7667]). of [RFC7667]).
d. Flow-based QoS cannot provide separate treatment of RTP streams d. Flow-based QoS cannot be used to provide separate treatment of
compared to others in the single RTP session. RTP streams compared to others in the single RTP session.
e. If there is significant asymmetry between the RTP 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 RTP stream that does usage to avoid wasting RTCP reporting on the RTP stream that does
not need that frequent reporting. not need that frequent reporting.
f. Not suitable for applications where some receivers like to f. It is not suitable for applications where some receivers like to
receive only a subset of the RTP streams, especially if multicast receive only a subset of the RTP streams, especially if multicast
or transport translator is being used. or transport translator is being used.
g. Additional concern with legacy implementations that do not g. There is some additional concern with legacy implementations that
support the RTP specification fully when it comes to handling do not support the RTP specification fully when it comes to
multiple SSRC per endpoint, as also multiple simultaneous media handling multiple SSRC per endpoint, as multiple simultaneous
types need to be handled. media types are sent as separate SSRC in the same RTP session.
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.
5.2. Multiple SSRCs of the Same Media Type 5.2. Multiple SSRCs of the Same Media Type
In this design, 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 RTP 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 [RFC5761].
The Pros: The Advantages
1. Works well with Split Component Terminal (see Section 3.10 of 1. It works well with Split Component Terminal (see Section 3.10 of
[RFC7667]) where the split is per media type. [RFC7667]) where the split is per media type.
2. Enables Flow-based QoS with different prioritisation between 2. It enables flow-based QoS with different prioritisation between
media types. media types.
3. For applications with dynamic usage of RTP streams, i.e. 3. For applications with dynamic usage of RTP streams, i.e.
frequently added and removed, having much of the state associated frequently added and removed, having much of the state associated
with the RTP session rather than per individual SSRC can avoid with the RTP session rather than per individual SSRC can avoid
the need for in-session signalling of meta-information about each the need for in-session signalling of meta-information about each
SSRC. SSRC.
4. Low overhead for security association establishment. 4. There is low overhead for security association establishment.
The Cons: The Disadvantages
a. Slightly higher number of RTP sessions needed compared to a. There are a slightly higher number of RTP sessions needed
Multiple Media Types in one Session Section 5.1. This implies: compared to Multiple Media Types in one Session Section 5.1.
This implies:
* More NAT/FW state * More NAT/FW state is needed.
* Increased NAT/FW Traversal Cost in both processing and delay. * There is increased NAT/FW-traversal cost in both processing
and delay.
b. Some potential for concern with legacy implementations that don't b. There is some potential for concern with legacy implementations
support the RTP specification fully when it comes to handling that don't support the RTP specification fully when it comes to
multiple SSRC per endpoint. handling multiple SSRC per endpoint.
c. Not possible to control security association for sets of RTP c. It is not possible to control security association for sets of
streams within the same media type with today's key- management RTP streams within the same media type with today's key-
mechanisms, unless these are split into different RTP sessions. management mechanisms, unless these are split into different RTP
sessions (Section 5.3).
For RTP applications where all RTP 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 for the negotiation signaling of
signalling of the properties for the individual media type, and also properties per individual media type, and also enables flow based QoS
enables flow based QoS prioritisation between media types. It prioritisation between media types. It handles multi-party sessions
handles multi-party session well, independently of multicast or well, independently of multicast or centralised transport
centralised transport distribution, as additional sources can distribution, as additional sources can dynamically enter and leave
dynamically enter and leave the session. the session.
5.3. Multiple Sessions for one Media type 5.3. Multiple Sessions for One Media Type
This design goes one step further than above (Section 5.2) by using This design goes one step further than above (Section 5.2) by using
multiple RTP sessions also for a single media type. The main reason multiple RTP sessions also for a single media type. The main reason
for going in this direction is that the RTP application needs for going in this direction is that the RTP application needs
separation of the RTP streams due to their usage. Some typical separation of the RTP streams due to their usage, such as e.g.
reasons for going to this design are scalability over multicast, scalability over multicast, simulcast, need for extended QoS
simulcast, need for extended QoS prioritisation of RTP streams due to prioritisation, or the need for fine-grained signalling using RTP
their usage in the application, or the need for fine- grained session-focused signalling tools.
signalling using today's tools.
The Pros: The Advantages:
1. More suitable for multicast usage where receivers can 1. This is 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. The application can indicate its usage of the RTP streams on RTP 2. The application can indicate its usage of the RTP streams on RTP
session level, in case multiple different usages exist. session level, in case multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media 3. There is less need for SSRC-specific explicit signalling for each
stream and thus reduced need for explicit and timely signalling. media stream and thus reduced need for explicit and timely
signalling when RTP streams are added or removed.
4. Enables detailed QoS prioritisation for flow-based mechanisms. 4. It enables detailed QoS prioritisation for flow-based mechanisms.
5. Works well with Split Component Terminal (see Section 3.10 of 5. It works well with Split Component Terminal (see Section 3.10 of
[RFC7667]). [RFC7667]).
6. 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 Disadvantages:
a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.
b. Increased amount of session configuration state. a. There is an increased amount of session configuration state
compared to Multiple SSRCs of the Same Media Type, due to the
increased amount of RTP sessions.
c. For RTP streams that are part of scalability, simulcast or b. For RTP streams that are part of scalability, simulcast or
transport robustness, a method to bind sources across multiple transport robustness, a method to bind sources across multiple
RTP sessions is needed. RTP sessions is needed.
d. Some potential for concern with legacy implementations that does c. There is some potential for concern with legacy implementations
not support the RTP specification fully when it comes to handling that don't support the RTP specification fully when it comes to
multiple SSRC per endpoint. handling multiple SSRC per endpoint.
e. Higher overhead for security association establishment due to the d. There is higher overhead for security association establishment,
increased number of RTP sessions. due to the increased number of RTP sessions.
f. If the applications need finer control than on RTP session level e. If the applications need more fine-grained control than per RTP
over which participants that are included in different sets of session over which participants that are included in different
security associations, most of today's key-management will have sets of security associations, most of today's key-management
difficulties establishing such a session. will have difficulties establishing such a session.
For more complex RTP applications that have several different usages For more complex RTP applications that have several different usages
for RTP streams of the same media type, or uses scalability or for RTP streams of the same media type, 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. Single SSRC per Endpoint 5.4. Single SSRC per Endpoint
In this design each endpoint in a point-to-point session has only a In this design each endpoint in a point-to-point session has only a
single SSRC, thus the RTP session contains only two SSRCs, one local single SSRC, thus the RTP session contains only two SSRCs, one local
and one remote. This session can be used both unidirectional, i.e. and one remote. This session can be used both unidirectional, i.e.
only a single RTP stream or bi-directional, i.e. both endpoints have only a single RTP stream, or bi-directional, i.e. both endpoints have
one RTP stream each. If the application needs additional media flows one RTP stream each. If the application needs additional media flows
between the endpoints, they will have to establish additional RTP between the endpoints, it will have to establish additional RTP
sessions. sessions.
The Pros: The Advantages:
1. This design 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 RTP stream, especially the exact formats and bitrates for each RTP stream, especially
using today's tools in SDP. using today's tools in SDP.
3. It is possible to control security association per RTP stream 3. It is possible to control security association per RTP stream
with current key-management, since each RTP stream is directly with current key-management, since each RTP stream is directly
related to an RTP session, and the most used keying mechanisms related to an RTP session, and the most used keying mechanisms
operates on a per-session basis. operates on a per-session basis.
The Cons: The Disadvantages:
a. The number of RTP sessions grows directly in proportion with the
number of RTP streams, which has the implications:
* Linear growth of the amount of NAT/FW state with number of RTP a. There is a linear growth of the amount of NAT/FW state with
streams. number of RTP streams.
* Increased delay and resource consumption from NAT/FW b. There is increased delay and resource consumption from NAT/FW-
traversal. traversal.
* Likely larger signalling message and signalling processing c. There are likely larger signalling message and signalling
requirement due to the amount of session related information. processing requirements due to the increased amount of session-
related information.
* Higher potential for a single RTP stream to fail during d. There is higher potential for a single RTP stream to fail during
transport between the endpoints. transport between the endpoints, due to the need for separate
NAT/FW- traversal for every RTP stream since there is only one
stream per session.
b. When the number of RTP sessions grows, the amount of explicit e. The amount of explicit state for relating RTP streams grows,
state for relating RTP streams also grows, depending on how the depending on how the application relates RTP streams.
application needs to relate RTP streams.
c. The port consumption might become a problem for centralised f. The port consumption might become a problem for centralised
services, where the central node's port or 5-tuple filter services, where the central node's port or 5-tuple filter
consumption grows rapidly with the number of sessions. consumption grows rapidly with the number of sessions.
d. For applications where the RTP stream usage is highly dynamic, g. For applications where the RTP stream usage is highly dynamic,
i.e. entering and leaving, the amount of signalling can grow i.e. entering and leaving, the amount of signalling can become
high. Issues can also arise from the timely establishment of high. Issues can also arise from the need for timely
additional RTP sessions. establishment of additional RTP sessions.
e. If, against the recommendation, the same SSRC value is reused in h. If, against the recommendation, the same SSRC value is reused in
multiple RTP sessions rather than being randomly chosen, multiple RTP sessions rather than being randomly chosen,
interworking with applications that use a different multiplexing interworking with applications that use a different multiplexing
structure will require SSRC translation. structure will require SSRC translation.
RTP applications that need to interwork with legacy RTP applications RTP applications with a strong need to interwork with legacy RTP
can potentially benefit from this structure. However, a large number applications 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 non-mixed multi-party
RTP stream from each participant, although having same usage in the sessions, as any given RTP stream from each participant, although
application, needs its own RTP session. In addition, the dynamic having same usage in the application, needs its own RTP session. In
behaviour that can arise in multi-party applications can tax the addition, the dynamic behaviour that can arise in multi-party
signalling system and make timely media establishment more difficult. applications can tax the signalling system and make timely media
establishment more difficult.
5.5. Summary 5.5. Summary
There are some clear similarities between these designs. Both the Both the "Single SSRC per Endpoint" and the "Multiple Media Types in
"Single SSRC per Endpoint" and the "Multiple Media Types in one One Session" are cases that 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 SSRC level binding. From another perspective, the
second needs SSRC level binding. From another perspective, the two two solutions are the two extreme points when it comes to number of
solutions are the two extreme points when it comes to number of RTP RTP sessions needed.
sessions needed.
The two other designs "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 two examples that "Multiple Sessions for One Media Type", are two examples that
primarily allows for some implicit mapping of the role or usage of primarily allows for some implicit mapping of the role or usage of
the RTP streams based on which RTP session they appear in. It thus the RTP streams based on which RTP session they appear in. It thus
potentially allows for less signalling and in particular reduces the potentially allows for less signalling and in particular reduces the
need for real-time signalling in dynamic sessions. They also need for real-time signalling in sessions with dynamically changing
represent points in between the first two designs when it comes to number of RTP streams. They also represent points in-between the
amount of RTP sessions established, i.e. representing an attempt to first two designs when it comes to amount of RTP sessions
balance the amount of RTP sessions with the functionality the established, i.e. representing an attempt to balance the amount of
communication session provides both on network level and on RTP sessions with the functionality the communication session
signalling level. provides both on network level and on signalling level.
6. Guidelines 6. Guidelines
This section contains a number of multi-stream guidelines for This section contains a number of multi-stream guidelines for
implementers or specification writers. implementers or specification writers.
Do not require use of the same SSRC value across RTP sessions: Do not require use of the same SSRC value across RTP sessions:
As discussed in Section 3.4.3 there exist drawbacks in using the As discussed in Section 3.4.3 there exist drawbacks in using the
same SSRC in multiple RTP sessions as a mechanism to bind related same SSRC in multiple RTP sessions as a mechanism to bind related
RTP streams together. It is instead recommended to use a RTP streams together. It is instead recommended to use a
skipping to change at page 32, line 16 skipping to change at page 32, line 4
suggested to send them in the same RTP session. For example a suggested to send them in the same RTP session. For example a
telepresence room where there are three cameras, and each camera telepresence room where there are three cameras, and each camera
captures 2 persons sitting at the table, sending each camera as captures 2 persons sitting at the table, sending each camera as
its own RTP stream within a single RTP session is suggested. its own RTP stream within a single RTP session is suggested.
Use additional RTP sessions for streams with different requirements: Use additional RTP sessions for streams with different requirements:
When RTP 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 a 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 Support Multiple RTP Streams 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 use 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 must support both. Extensions that are not as generally generic must support both. Extensions that are not as generally
applicable will have to consider if interoperability is better applicable will have to consider if interoperability is better
served by defining a single solution or providing both options. served by defining a single solution or providing both 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 must include support for both multiple RTP mechanisms, they must include support for both multiple RTP
streams in the same RTP sessions and multiple RTP sessions, such streams in the same RTP session and multiple RTP sessions, such
that application developers can choose freely from the set of that application developers can choose freely from the set of
mechanisms without concerning themselves with which of the mechanisms without concerning themselves with which of the
multiplexing choices a particular solution supports. multiplexing choices a particular solution supports.
7. IANA Considerations 7. 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.
8. Security Considerations 8. Security Considerations
The security considerations of the RTP specification [RFC3550] and The security considerations of the RTP specification [RFC3550], any
any applicable RTP profile [RFC3551],[RFC4585],[RFC3711], the applicable RTP profile [RFC3551],[RFC4585],[RFC3711], and the
extensions for sending multiple media types in a single RTP session extensions for sending multiple media types in a single RTP session
[I-D.ietf-avtcore-multi-media-rtp-session], RID [I-D.ietf-avtcore-multi-media-rtp-session], RID
[I-D.ietf-mmusic-rid], BUNDLE [I-D.ietf-mmusic-rid], BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation], [RFC5760], [RFC5761], apply [I-D.ietf-mmusic-sdp-bundle-negotiation], [RFC5760], [RFC5761], apply
if selected and thus needs to be considered in the evaluation. if selected and thus need to be considered in the evaluation.
There is discussion of the security implications of choosing multiple There is discussion of the security implications of choosing multiple
SSRC vs multiple RTP sessions in Section 4.3. SSRC vs multiple RTP sessions in Section 4.3.
9. Contributors 9. Contributors
Hui Zheng (Marvin) from Huawei contributed to WG draft versions -04 Hui Zheng (Marvin) contributed to WG draft versions -04 and -05 of
and -05 of the document. the document.
10. References 10. Acknowledgments
10.1. Normative References The Authors like to acknowledge and thank Cullen Jennings, Dale R
Worley, Huang Yihong (Rachel), and Vijay Gurbani for review and
comments.
11. References
11.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-13 (work in
progress), December 2015.
[I-D.ietf-mmusic-rid]
Roach, A., "RTP Payload Format Restrictions", draft-ietf-
mmusic-rid-15 (work in progress), May 2018.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-54 (work in progress), December 2018.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-14 (work in progress), March 2019.
[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-10 (work in progress),
July 2019.
[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>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
DOI 10.17487/RFC3830, August 2004,
<https://www.rfc-editor.org/info/rfc3830>.
[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>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<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
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[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 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
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-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-13 (work in
progress), December 2015.
[I-D.ietf-avtext-rid] [I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", draft-ietf-avtext- Identifier Source Description (SDES)", draft-ietf-avtext-
rid-09 (work in progress), October 2016. rid-09 (work in progress), October 2016.
[I-D.ietf-mmusic-rid] [I-D.ietf-perc-private-media-framework]
Roach, A., "RTP Payload Format Restrictions", draft-ietf- Jones, P., Benham, D., and C. Groves, "A Solution
mmusic-rid-15 (work in progress), May 2018. Framework for Private Media in Privacy Enhanced RTP
Conferencing (PERC)", draft-ietf-perc-private-media-
[I-D.ietf-mmusic-sdp-bundle-negotiation] framework-12 (work in progress), June 2019.
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-53 (work in progress), September 2018.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-13 (work in progress), June 2018.
[I-D.ietf-perc-srtp-ekt-diet] [JINGLE] Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
Jennings, C., Mattsson, J., McGrew, D., Wing, D., and F. S., and J. Hildebrand, "XEP-0166: Jingle", XMPP.org
Andreasen, "Encrypted Key Transport for DTLS and Secure https://xmpp.org/extensions/xep-0166.html, September 2018.
RTP", draft-ietf-perc-srtp-ekt-diet-09 (work in progress),
October 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 35, line 20 skipping to change at page 36, line 20
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002, DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>. <https://www.rfc-editor.org/info/rfc3264>.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://www.rfc-editor.org/info/rfc3389>. September 2002, <https://www.rfc-editor.org/info/rfc3389>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
DOI 10.17487/RFC3830, August 2004,
<https://www.rfc-editor.org/info/rfc3830>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, 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 [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383, Real-time Transport Protocol (SRTP)", RFC 4383,
DOI 10.17487/RFC4383, February 2006, DOI 10.17487/RFC4383, February 2006,
<https://www.rfc-editor.org/info/rfc4383>. <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>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
Media Attributes in the Session Description Protocol "Session Traversal Utilities for NAT (STUN)", RFC 5389,
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, DOI 10.17487/RFC5389, October 2008,
<https://www.rfc-editor.org/info/rfc5576>. <https://www.rfc-editor.org/info/rfc5389>.
[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
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, 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,
skipping to change at page 37, line 20 skipping to change at page 37, line 36
[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>.
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657, (Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015, DOI 10.17487/RFC7657, November 2015,
<https://www.rfc-editor.org/info/rfc7657>. <https://www.rfc-editor.org/info/rfc7657>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC7826] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., [RFC7826] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, Ed., "Real-Time Streaming Protocol and M. Stiemerling, Ed., "Real-Time Streaming Protocol
Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December
2016, <https://www.rfc-editor.org/info/rfc7826>. 2016, <https://www.rfc-editor.org/info/rfc7826>.
[RFC7983] Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
Updates for Secure Real-time Transport Protocol (SRTP)
Extension for Datagram Transport Layer Security (DTLS)",
RFC 7983, DOI 10.17487/RFC7983, September 2016,
<https://www.rfc-editor.org/info/rfc7983>.
[RFC8088] Westerlund, M., "How to Write an RTP Payload Format", [RFC8088] Westerlund, M., "How to Write an RTP Payload Format",
RFC 8088, DOI 10.17487/RFC8088, May 2017, RFC 8088, DOI 10.17487/RFC8088, May 2017,
<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>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445, Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018, DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>. <https://www.rfc-editor.org/info/rfc8445>.
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 is unsuitable for most things related to as a multiplexing point is unsuitable for most issues related to
multiple RTP streams. If one attempts to use Payload type multiple RTP streams. Attempting to use Payload type multiplexing
multiplexing beyond its defined usage, that has well known negative beyond its defined usage has well known negative effects on RTP
effects on RTP. To use payload type as the single discriminator for discussed below. To use payload type as the single discriminator for
multiple streams implies that all the different RTP streams are being multiple streams implies that all the different RTP streams are being
sent with the same SSRC, thus using the same timestamp and sequence sent with the same SSRC, thus using the same timestamp and sequence
number space. This has many effects: number space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed 1. Putting constraints on RTP timestamp rate for the multiplexed
media. For example, RTP 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 RTP timestamp rate. When streams are forced to use the same RTP timestamp rate. When
this is not 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 RTP packets, like parts of a video frame. These multiple RTP packets, like parts of a video frame. These
payload 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
skipping to change at page 38, line 49 skipping to change at page 39, line 21
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 RTP 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 RTP payload type 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.
skipping to change at page 39, line 38 skipping to change at page 40, line 8
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 RTP 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
important for anyone building complete applications, so it is extremely important for anyone building complete applications, so it
deserving of discussion. is deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with We document salient issues here that need to be addressed by the WGs
signalling; they cannot be addressed by tweaking, extending or that use some form of signaling to establish RTP sessions. These
profiling RTP. issues cannot simply be addressed by tweaking, extending, or
profiling RTP, but require a dedicated and indepth look at the
signaling primitives that set up the RTP sessions.
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. RTSP also dependent on the signalling protocols carrying the SDP. RTSP
[RFC7826] and SAP [RFC2974] both use SDP in a declarative fashion, [RFC7826] and SAP [RFC2974] both use SDP in a declarative fashion,
while SIP [RFC3261] uses SDP with the additional definition of Offer/ while SIP [RFC3261] uses SDP with the additional definition of Offer/
Answer [RFC3264]. The impact on signalling and especially SDP needs Answer [RFC3264]. The impact on signalling and especially SDP needs
to be considered as it can greatly affect how to deploy a certain to be considered as it can greatly affect how to deploy a certain
multiplexing point choice. multiplexing point choice.
B.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 on RTP
RTP sessions, or at least in the case of SDP the media description. sessions, or at least in the case of SDP the media description.
There are a number of things that are signalled on media description There are a number of things that are signalled on media description
level but those are not necessarily strictly bound to an RTP session level but those are not necessarily strictly bound to an RTP session
and could be of interest to signal specifically for a particular RTP and could be of interest to signal specifically for a particular RTP
stream (SSRC) within the session. The following properties have been stream (SSRC) within the session. The following properties have been
identified as being potentially useful to signal not only on RTP identified as being potentially useful to signal not only on RTP
session level: session level:
o Bitrate/Bandwidth exist today only at aggregate or as a common o Bitrate/Bandwidth exist today only at aggregate or as a common
"any RTP 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 type (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 RTP 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
skipping to change at page 41, line 10 skipping to change at page 41, line 27
type for identifying the actual payload format and is bound to a type for identifying the actual payload format and is bound to a
particular payload type using the rtpmap attribute. This binding has particular payload type using the rtpmap attribute. This binding has
to be loosened in order to use SDP to describe RTP sessions to be loosened in order to use SDP to describe RTP sessions
containing multiple MIME top level types. containing multiple MIME top level types.
[I-D.ietf-mmusic-sdp-bundle-negotiation] describes how to let [I-D.ietf-mmusic-sdp-bundle-negotiation] describes how to let
multiple SDP media descriptions use a single underlying transport in multiple SDP media descriptions use a single underlying transport in
SDP, which allows to define one RTP session with media types having SDP, which allows to define one RTP session with media types having
different MIME top level types. different MIME top level types.
B.3. Signalling RTP stream Usage B.3. Signalling RTP Stream Usage
RTP streams being transported in RTP has some particular usage in an RTP streams being transported in RTP has some particular usage in an
RTP application. This usage of the RTP 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 RTP 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 RTP 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 RTP streams affected by their RTP- used will have to identify the RTP streams affected by their RTP-
skipping to change at page 41, line 44 skipping to change at page 42, line 15
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 RTP 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 Torshamnsgatan 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
Gronlandsgatan 31 Gronlandsgatan 31
SE-164 60 Kista SE-164 60 Kista
Sweden Sweden
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
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