< draft-westerlund-avtcore-multiplex-architecture-00.txt   draft-westerlund-avtcore-multiplex-architecture-01.txt >
Network Working Group M. Westerlund Network Working Group M. Westerlund
Internet-Draft B. Burman Internet-Draft B. Burman
Intended status: BCP Ericsson Intended status: Informational Ericsson
Expires: April 26, 2012 C. Perkins Expires: September 13, 2012 C. Perkins
University of Glasgow University of Glasgow
October 24, 2011 March 12, 2012
RTP Multiplexing Architecture RTP Multiplexing Architecture
draft-westerlund-avtcore-multiplex-architecture-00 draft-westerlund-avtcore-multiplex-architecture-01
Abstract Abstract
RTP has always been a protocol that supports multiple participants Real-time Transport Protocol is a flexible protocol possible to use
each sending their own media streams in an RTP session. Thus relying in a wide range of applications and network and system topologies.
on the three main multiplexing points in RTP; RTP session, SSRC and This flexibility and the implications of different choices should be
Payload Type for their various needs. However, most usages of RTP understood by any application developer using RTP. To facilitate
have been less complex often with a single SSRC in each direction, that understanding, this document contains an in-depth discussion of
with a single RTP session per media type. But the more complex the usage of RTP's multiplexing points; the RTP session, the
usages start to be more common and thus guidance on how to use RTP in Synchronization Source Identifier (SSRC), and the payload type. The
various complex cases are needed. This document analyzes a number of focus is put on the first two, trying to give guidance and source
cases and discusses the usage of the various multiplexing points and material for an analysis on the most suitable choices for the
the need for functionality when defining RTP/RTCP extensions that application being designed.
utilize multiple RTP streams and multiple RTP sessions.
Status of this Memo Status of this Memo
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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This Internet-Draft will expire on April 26, 2012. This Internet-Draft will expire on September 13, 2012.
Copyright Notice Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Requirements Language . . . . . . . . . . . . . . . . . . 5 2.1. Requirements Language . . . . . . . . . . . . . . . . . . 5
2.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5 2.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
3. RTP Multiplex Points . . . . . . . . . . . . . . . . . . . . . 6 3. RTP Multiplex Points . . . . . . . . . . . . . . . . . . . . . 6
3.1. Session . . . . . . . . . . . . . . . . . . . . . . . . . 6 3.1. Session . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.2. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 3.2. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.3. CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 8 3.3. CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.4. Payload Type . . . . . . . . . . . . . . . . . . . . . . . 8 3.4. Payload Type . . . . . . . . . . . . . . . . . . . . . . . 9
4. Multiple Streams Alternatives . . . . . . . . . . . . . . . . 9 4. Multiple Streams Alternatives . . . . . . . . . . . . . . . . 10
5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 10 5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 11
5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 11 5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 12
5.1.1. RTCP Reporting . . . . . . . . . . . . . . . . . . . . 11 5.1.1. RTCP Reporting . . . . . . . . . . . . . . . . . . . . 12
5.1.2. Compound RTCP Packets . . . . . . . . . . . . . . . . 12 5.1.2. Compound RTCP Packets . . . . . . . . . . . . . . . . 13
5.2. Point to Multipoint Using Multicast . . . . . . . . . . . 12 5.2. Point to Multipoint Using Multicast . . . . . . . . . . . 13
5.3. Point to Multipoint Using an RTP Translator . . . . . . . 14 5.3. Point to Multipoint Using an RTP Translator . . . . . . . 15
5.4. Point to Multipoint Using an RTP Mixer . . . . . . . . . . 15 5.4. Point to Multipoint Using an RTP Mixer . . . . . . . . . . 16
5.5. Point to Multipoint using Multiple Unicast flows . . . . . 16 5.5. Point to Multipoint using Multiple Unicast flows . . . . . 17
5.6. Decomposited End-Point . . . . . . . . . . . . . . . . . . 16 5.6. De-composite End-Point . . . . . . . . . . . . . . . . . . 18
6. Dismissing Payload Type Multiplexing . . . . . . . . . . . . . 18 6. Multiple Streams Discussion . . . . . . . . . . . . . . . . . 19
7. Multiple Streams Discussion . . . . . . . . . . . . . . . . . 20 6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 19
7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 20 6.2. RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 19
7.2. RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 20 6.2.1. The RTP Specification . . . . . . . . . . . . . . . . 19
7.2.1. The RTP Specification . . . . . . . . . . . . . . . . 20 6.2.2. Handling Varying sets of Senders . . . . . . . . . . . 22
7.2.2. Multiple SSRC Legacy Considerations . . . . . . . . . 22 6.2.3. Cross Session RTCP Requests . . . . . . . . . . . . . 22
7.2.3. RTP Specification Clarifications Needed . . . . . . . 23 6.2.4. Binding Related Sources . . . . . . . . . . . . . . . 22
7.2.4. Handling Varying sets of Senders . . . . . . . . . . . 23 6.2.5. Forward Error Correction . . . . . . . . . . . . . . . 24
7.2.5. Cross Session RTCP requests . . . . . . . . . . . . . 23 6.2.6. Transport Translator Sessions . . . . . . . . . . . . 25
7.2.6. Binding Related Sources . . . . . . . . . . . . . . . 23 6.3. Interworking . . . . . . . . . . . . . . . . . . . . . . . 25
7.2.7. Forward Error Correction . . . . . . . . . . . . . . . 25 6.3.1. Interworking Applications . . . . . . . . . . . . . . 26
7.2.8. Transport Translator Sessions . . . . . . . . . . . . 26 6.3.2. Multiple SSRC Legacy Considerations . . . . . . . . . 27
7.2.9. Multiple Media Types in one RTP session . . . . . . . 26 6.4. Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28
7.3. Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28 6.4.1. Session Oriented Properties . . . . . . . . . . . . . 28
7.3.1. Session Oriented Properties . . . . . . . . . . . . . 28 6.4.2. SDP Prevents Multiple Media Types . . . . . . . . . . 29
7.3.2. SDP Prevents Multiple Media Types . . . . . . . . . . 29 6.4.3. Media Stream Usage . . . . . . . . . . . . . . . . . . 29
7.4. Network Apsects . . . . . . . . . . . . . . . . . . . . . 29 6.5. Network Aspects . . . . . . . . . . . . . . . . . . . . . 30
7.4.1. Quality of Service . . . . . . . . . . . . . . . . . . 29 6.5.1. Quality of Service . . . . . . . . . . . . . . . . . . 30
7.4.2. NAT and Firewall Traversal . . . . . . . . . . . . . . 29 6.5.2. NAT and Firewall Traversal . . . . . . . . . . . . . . 31
7.4.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 31 6.5.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 32
7.4.4. Multiplexing multiple RTP Session on a Single 6.5.4. Multiplexing multiple RTP Session on a Single
Transport . . . . . . . . . . . . . . . . . . . . . . 32 Transport . . . . . . . . . . . . . . . . . . . . . . 33
7.5. Security Aspects . . . . . . . . . . . . . . . . . . . . . 32 6.6. Security Aspects . . . . . . . . . . . . . . . . . . . . . 33
7.5.1. Security Context Scope . . . . . . . . . . . . . . . . 32 6.6.1. Security Context Scope . . . . . . . . . . . . . . . . 33
7.5.2. Key-Management for Multi-party session . . . . . . . . 33 6.6.2. Key-Management for Multi-party session . . . . . . . . 34
8. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 33 6.6.3. Complexity Implications . . . . . . . . . . . . . . . 34
9. RTP Specification Clarifications . . . . . . . . . . . . . . . 35 6.7. Multiple Media Types in one RTP session . . . . . . . . . 35
9.1. RTCP Reporting from all SSRCs . . . . . . . . . . . . . . 35 7. Arch-Types . . . . . . . . . . . . . . . . . . . . . . . . . . 37
9.2. RTCP Self-reporting . . . . . . . . . . . . . . . . . . . 35 7.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 37
9.3. Combined RTCP Packets . . . . . . . . . . . . . . . . . . 35 7.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 39
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35 7.3. Multiple Sessions for one Media type . . . . . . . . . . . 40
11. Security Considerations . . . . . . . . . . . . . . . . . . . 36 7.4. Multiple Media Types in one Session . . . . . . . . . . . 41
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36 7.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 43
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36 8. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 43
13.1. Normative References . . . . . . . . . . . . . . . . . . . 36 9. Proposal for Future Work . . . . . . . . . . . . . . . . . . . 44
13.2. Informative References . . . . . . . . . . . . . . . . . . 36 10. RTP Specification Clarifications . . . . . . . . . . . . . . . 45
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 39 10.1. RTCP Reporting from all SSRCs . . . . . . . . . . . . . . 45
10.2. RTCP Self-reporting . . . . . . . . . . . . . . . . . . . 45
10.3. Combined RTCP Packets . . . . . . . . . . . . . . . . . . 45
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 46
12. Security Considerations . . . . . . . . . . . . . . . . . . . 46
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 46
14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 46
14.1. Normative References . . . . . . . . . . . . . . . . . . . 46
14.2. Informative References . . . . . . . . . . . . . . . . . . 46
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 49
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 51
1. Introduction 1. Introduction
This document focuses at issues of non-basic usage of RTP [RFC3550] Real-time Transport Protocol (RTP) [RFC3550] is a commonly used
where multiple media sources of the same media type are sent over protocol for real-time media transport. It is a protocol that
RTP. Separation of different media types is another issue that will provides great flexibility and can support a large set of different
be discussed in this document. The intended uses include for example applications. RTP has several multiplexing points designed for
multiple sources from the same end-point, multiple streams from a different purposes. These enable support of multiple media streams
single media source, multiple end-points each having a source, or an and switching between different encoding or packetization of the
application that needs multiple representations (encodings) of a media. By using multiple RTP sessions, sets of media streams can be
particular source. It will be shown that these uses are inter- structured for efficient processing or identification. Thus the
related and need a common discussion to ensure consistency. In question for any RTP application designer is how to best use the RTP
general, usage of the RTP session and media streams will be discussed session, the SSRC and the payload type to meet the application's
in detail. needs.
RTP is already designed for multiple participants in a communication
session. This is not restricted to multicast, as many believe, but
also provides functionality over unicast, using either multiple
transport flows below RTP or a network node that re-distributes the
RTP packets. The node can for example be a transport translator
(relay) that forwards the packets unchanged, a translator performing
media translation in addition to forwarding, or an RTP mixer that
creates new conceptual sources from the received streams. In
addition, multiple streams may occur when a single end-point have
multiple media sources of the same media type, like multiple cameras
or microphones that need to be sent simultaneously.
Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per end-point and media type (typically audio
and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each end-point and the
RTP mixer.
SSRC is the RTP media stream identifier that helps to uniquely The purpose of this document is to provide clear information about
identify media sources in RTP sessions. Even though available SSRC the possibilities of RTP when it comes to multiplexing. The RTP
space can theoretically handle more than 4 billion simultaneous application designer should understand the implications that come
sources, the perceived need for handling multiple SSRCs in from a particular choice of RTP multiplexing points. The document
implementations has been small. This has resulted in an installed will recommend against some usages as being unsuitable, in general or
legacy base that isn't fully RTP specification compliant and will for particular purposes.
have different issues if they receive multiple SSRCs of media, either
simultaneously or in sequence. These issues will manifest themselves
in various ways, either by software crashes or simply in limited
functionality, like only decoding and playing back the first or
latest received SSRC and discarding media related to any other SSRCs.
There have also arisen various cases where multiple SSRCs are used to RTP was from the beginning designed for multiple participants in a
represent different aspects of what is in fact a single underlying communication session. This is not restricted to multicast, as some
media source. A very basic case is RTP retransmission [RFC4588] may believe, but also provides functionality over unicast, using
which have one SSRC for the original stream, and a second SSRC either either multiple transport flows below RTP or a network node that re-
in the same session or in a different session to represent the distributes the RTP packets. The re-distributing node can for
retransmitted packets to ensure that the monitoring functions still example be a transport translator (relay) that forwards the packets
function. Another use case is scalable encoding, such as the RTP unchanged, a translator performing media translation in addition to
payload format for Scalable Video Coding (SVC) [RFC6190], which has forwarding, or an RTP mixer that creates new conceptual sources from
an operation mode named Multiple Session Transmission (MST) that uses the received streams. In addition, multiple streams may occur when a
one SSRC in each RTP session to send one or more scalability layers. single end-point have multiple media sources, like multiple cameras
A third example is simulcast where a single media source is encoded or microphones that need to be sent simultaneously.
in different versions and transmitted to an RTP mixer that picks
which version to actually distribute to a given receiver part of the
RTP session.
This situation has created a need for a document that discusses the This document has been written due to increased interest in more
existing possibilities in the RTP protocol and how these can and advanced usage of RTP, resulting in questions regarding the most
should be used in applications. A new set of applications needing appropriate RTP usage. The limitations in some implementations, RTP/
more advanced functionalities from RTP is also emerging on the RTCP extensions, and signalling has also been exposed. It is
market, such as telepresence and advanced video conferencing. Thus expected that some limitations will be addressed by updates or new
furthering the need for a more common understanding of how multiple extensions resolving the shortcomings. The authors also hope that
streams are handled in RTP to ensure media plane interoperability. clarification on the usefulness of some functionalities in RTP will
result in more complete implementations in the future.
The document starts with some definitions and then goes into the The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired existing RTP functionalities around multiplexing. Both the desired
behavior and the implications of a particular behavior depend on behavior and the implications of a particular behavior 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 behavior is followed by a discussion of some choices in multiplexing behavior
and their impacts. Finally, some recommendations and examples are and their impacts. Some arch-types of RTP usage are discussed.
provided.
Finally, some recommendations and examples are provided.
This document is currently an individual contribution, but it is the
intention of the authors that this should become a WG document that
objectively describes and provides suitable recommendations for which
there is WG consensus. Currently this document only represents the
views of the authors. The authors gladly accept any feedback on the
document and will be happy to discuss suitable recommendations.
2. Definitions 2. Definitions
2.1. Requirements Language 2.1. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119]. document are to be interpreted as described in RFC 2119 [RFC2119].
2.2. Terminology 2.2. Terminology
The following terms and abbreviations are used in this document: The following terms and abbreviations are used in this document:
End-point: A single entity sending or receiving RTP packets. It may End-point: A single entity sending or receiving RTP packets. It may
be decomposed into several functional blocks, but as long as it be decomposed into several functional blocks, but as long as it
behaves a single RTP stack entity it is classified as a single behaves a single RTP stack entity it is classified as a single
end-point. end-point.
Media Stream: A sequence of RTP packets using a single SSRC that Media Stream: A sequence of RTP packets using a single SSRC that
together carry part or all of the content of a specific Media Type together carries part or all of the content of a specific Media
from a specific sender source within a given RTP session. Type from a specific sender source within a given RTP session.
Media Source: The originator or source of a particular Media Stream.
It can either be a single media capturing device such as a video
camera, a microphone, or a specific output of a media production
function, such as an audio mixer, or some video editing function.
Media Aggregate: All Media Streams related to a particular Source. Media Aggregate: All Media Streams related to a particular Source.
Media Type: Audio, video, text or data whose form and meaning are Media Type: Audio, video, text or data whose form and meaning are
defined by a specific real-time application. defined by a specific real-time application.
Source: The source of a particular media stream. Either a single Multiplex: The operation of taking multiple entities as input,
media capturing device such as a video camera, or a microphone, or aggregating them onto some common resource while keeping the
a specific output of a media production function, such as an audio individual entities addressable such that they can later be fully
mixer, or some video editing function. and unambiguously separated (de-multiplexed) again.
RTP Session: As defined by [RFC3550], the end-points belonging to
the same RTP Session are those that share a single SSRC space.
That is, those end-points can see an SSRC identifier transmitted
by any one of the other end-points. An end-point can receive an
SSRC either as SSRC or as CSRC in RTP and RTCP packets. Thus, the
RTP Session scope is decided by the end-points' network
interconnection topology, in combination with RTP and RTCP
forwarding strategies deployed by end-points and any
interconnecting middle nodes.
Source: See Media Source.
3. RTP Multiplex Points 3. RTP Multiplex Points
This section describes the existing RTP tools that enable This section describes the existing RTP tools that enable
multiplexing of different media streams and RTP functionalities. multiplexing of different media streams.
3.1. Session 3.1. Session
The RTP Session is the highest semantic level in RTP and contains all The RTP Session is the highest semantic level in RTP and contains all
of the RTP functionality. of the RTP functionality.
RTP in itself does not contain any Session identifier, but relies on Identifier: RTP in itself does not contain any Session identifier,
the underlying transport. For example, when running RTP on top of but relies either on the underlying transport or on the used
UDP, an RTP endpoint can identify and delimit an RTP Session from signalling protocol, depending on in which context the identifier
other RTP Sessions through the UDP source and destination transport is used (e.g. transport or signalling). Due to this, a single RTP
address, consisting of network address and port number(s). Most Session may have multiple associated identifiers belonging to
commonly only the destination address, i.e. all traffic received on a different contexts.
particular port, is defined as belonging to a specific RTP Session.
It is worth noting that in practice a more narrow definition of the
transport flows that are related to a give RTP session is possible.
An RTP session can for example be defined as one or more 5-tuples
(Transport Protocol, Source Address, Source Port, Destination
Address, Destination Port). Any set of identifiers of RTP and RTCP
packet flows are sufficient to determine if the flow belongs to a
particular session or not.
Commonly, RTP and RTCP use separate ports and the destination Position: Depending on underlying transport and signalling
transport address is in fact an address pair, but in the case of RTP/ protocol. For example, when running RTP on top of UDP, an RTP
RTCP multiplex [RFC5761] there is only a single port. endpoint can identify and delimit an RTP Session from other RTP
Sessions through the UDP source and destination transport
address, consisting of network address and port number(s).
Commonly, RTP and RTCP use separate ports and the destination
transport address is in fact an address pair, but in the case
of RTP/RTCP multiplex [RFC5761] there is only a single port.
Another example is SDP signalling [RFC4566], where the grouping
framework [RFC5888] uses an identifier per "m="-line. If there
is a one-to-one mapping between "m="-line and RTP Session, that
grouping framework identifier can identify a single RTP
Session.
Usage: Identify separate RTP Sessions.
Uniqueness: Globally unique within the general communication
context for the specific end-point.
Inter-relation: Depending on the underlying transport and
signalling protocol.
Special Restrictions: None.
A source that changes its source transport address during a session A source that changes its source transport address during a session
must also choose a new SSRC identifier to avoid being interpreted as must also choose a new SSRC identifier to avoid being interpreted as
a looped source. a looped source.
The set of participants considered part of the same RTP Session is The set of participants considered part of the same RTP Session is
defined by[RFC3550] as those that share a single SSRC space. That defined by[RFC3550] as those that share a single SSRC space. That
is, those participants that can see an SSRC identifier transmitted by is, those participants that can see an SSRC identifier transmitted by
any one of the other participants. A participant can receive an SSRC any one of the other participants. A participant can receive an SSRC
either as SSRC or CSRC in RTP and RTCP packets. Thus, the RTP either as SSRC or CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the participants' network interconnection Session scope is decided by the participants' network interconnection
topology, in combination with RTP and RTCP forwarding strategies topology, in combination with RTP and RTCP forwarding strategies
deployed by end-points and any interconnecting middle nodes. deployed by end-points and any interconnecting middle nodes.
3.2. SSRC 3.2. SSRC
The Synchronization Source (SSRC) identifier is used to identify An RTP Session serves one or more Media Sources, each sending a Media
individual sources within an RTP Session. The SSRC number is Stream.
globally unique within an RTP Session and all RTP implementations
must be prepared to use procedures for SSRC collision handling, which Identifier: Synchronization Source (SSRC), 32-bit unsigned number.
results in an SSRC number change. The SSRC number is randomly
chosen, carried in every RTP packet header and is not dependent on Position: In every RTP and RTCP packet header. May be present in
network address. SSRC is also used as identifier to refer to RTCP payload. May be present in SDP signalling.
separate media streams in RTCP.
Usage: Identify individual Media Sources within an RTP Session.
Refer to individual Media Sources in RTCP messages and SDP
signalling.
Uniqueness: Randomly chosen, globally unique within an RTP
Session and not dependent on network address.
Inter-relation: SSRC belonging to the same synchronization
context (originating from the same end-point), within or
between RTP Sessions, are indicated through use of identical
SDES CNAME items in RTCP compound packets with those SSRC as
originating source. SDP signalling can provide explicit SSRC
grouping [RFC5576]. When CNAME is inappropriate or
insufficient, there exist a few other methods to relate
different SSRC. One such case is session-based RTP
retransmission [RFC4588]. In some cases, the same SSRC
Identifier value is used to relate streams in two different RTP
Sessions, such as in Multi-Session Transmission of scalable
video [RFC6190].
Special Restrictions: All RTP implementations must be prepared to
use procedures for SSRC collision handling, which results in an
SSRC number change. A Media Source that changes its RTP Session
identifier (e.g. source transport address) during a session must
also choose a new SSRC identifier to avoid being interpreted as
looped source. Note that RTP sequence number and RTP timestamp
are scoped by SSRC and thus independent between different SSRCs.
A media source having an SSRC identifier can be of different types: A media source having an SSRC identifier can be of different types:
Real: Connected to a "physical" media source, for example a camera Real: Connected to a "physical" media source, for example a camera
or microphone. or microphone.
Conceptual: A source with some attributed property generated by some Conceptual: A source with some attributed property generated by some
network node, for example a filtering function in an RTP mixer network node, for example a filtering function in an RTP mixer
that provides the most active speaker based on some criteria, or a that provides the most active speaker based on some criteria, or a
mix representing a set of other sources. mix representing a set of other sources.
skipping to change at page 7, line 47 skipping to change at page 8, line 37
anyway need a sender SSRC for use as source in RTCP reports. anyway need a sender SSRC for use as source in RTCP reports.
Note that a "multimedia source" that generates more than one media Note that a "multimedia source" that generates more than one media
type, e.g. a conference participant sending both audio and video, type, e.g. a conference participant sending both audio and video,
need not (and commonly should not) use the same SSRC value across RTP need not (and commonly should not) use the same SSRC value across RTP
sessions. RTCP Compound packets containing the CNAME SDES item is sessions. RTCP Compound packets containing the CNAME SDES item is
the designated method to bind an SSRC to a CNAME, effectively cross- the designated method to bind an SSRC to a CNAME, effectively cross-
correlating SSRCs within and between RTP Sessions as coming from the correlating SSRCs within and between RTP Sessions as coming from the
same end-point. The main property attributed to SSRCs associated same end-point. The main property attributed to SSRCs associated
with the same CNAME is that they are from a particular with the same CNAME is that they are from a particular
synchronization context and may be synchronized at playback. There synchronization context and may be synchronized at playback.
exist a few other methods to relate different SSRC where use of CNAME
is inappropriate, such as session-based RTP retransmission [RFC4588].
Note also that RTP sequence number and RTP timestamp are scoped by Note also that RTP sequence number and RTP timestamp are scoped by
SSRC and thus independent between different SSRCs. SSRC and thus independent between different SSRCs.
An RTP receiver receiving a previously unseen SSRC value must An RTP receiver receiving a previously unseen SSRC value must
interpret it as a new source. It may in fact be a previously interpret it as a new source. It may 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 should have conflict. However, the originator of the previous SSRC should have
ended the conflicting source by sending an RTCP BYE for it prior to ended the conflicting source by sending an RTCP BYE for it prior to
starting to send with the new SSRC, so the new SSRC is anyway starting to send with the new SSRC, so the new SSRC is anyway
skipping to change at page 8, line 15 skipping to change at page 9, line 4
An RTP receiver receiving a previously unseen SSRC value must An RTP receiver receiving a previously unseen SSRC value must
interpret it as a new source. It may in fact be a previously interpret it as a new source. It may 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 should have conflict. However, the originator of the previous SSRC should have
ended the conflicting source by sending an RTCP BYE for it prior to ended the conflicting source by sending an RTCP BYE for it prior to
starting to send with the new SSRC, so the new SSRC is anyway starting to send with the new SSRC, so the new SSRC is anyway
effectively a new source. effectively a new source.
Some RTP extension mechanisms already require the RTP stacks to Some RTP extension mechanisms already require the RTP stacks to
handle additional SSRCs, like SSRC multiplexed RTP retransmission handle additional SSRCs, like SSRC multiplexed RTP retransmission
[RFC4588]. However, that still only requires handling a single media [RFC4588]. However, that still only requires handling a single media
decoding chain per pair of SSRCs. decoding chain per pair of SSRCs.
3.3. CSRC 3.3. CSRC
The Contributing Source (CSRC) can arguably be seen as a sub-part of The Contributing Source (CSRC) can arguably be seen as a sub-part of
a specific SSRC and thus a multiplexing point. It is optionally a specific SSRC and thus a multiplexing point. It is optionally
included in the RTP header, shares the SSRC number space and included in the RTP header, shares the SSRC number space and
specifies which set of SSRCs that has contributed to the RTP payload. specifies which set of SSRCs that has contributed to the RTP payload.
However, even though each RTP packet and SSRC can be tagged with the However, even though each RTP packet and SSRC can be tagged with the
contained CSRCs, the media representation of an individual CSRC is in contained CSRCs, the media representation of an individual CSRC is in
general not possible to extract from the RTP payload since it is general not possible to extract from the RTP payload since it is
typically the result of a media mixing (merge) operation (by an RTP typically the result of a media mixing (merge) operation (by an RTP
mixer) on the individual media streams corresponding to the CSRC mixer) on the individual media streams corresponding to the CSRC
identifiers. Due to these restrictions, CSRC will not be considered identifiers. Due to these restrictions, CSRC will not be considered
a fully qualified multiplex point and will be disregarded in the rest a fully qualified multiplex point and will be disregarded in the rest
of this document. of this document.
3.4. Payload Type 3.4. Payload Type
The Payload Type number is also carried in every RTP packet header Each Media Stream can be represented in various encoding formats.
and identifies what format the RTP payload has. The term "format"
here includes whatever can be described by out-of-band signaling
means for dynamic payload types, as well as the statically allocated
payload types in [RFC3551]. In SDP the term "format" includes media
type, RTP timestamp sampling rate, codec, codec configuration,
payload format configurations, and various robustness mechanisms such
as redundant encodings [RFC2198].
The meaning of a Payload Type definition (the number) is re-used Identifier: Payload Type number.
between all media streams within an RTP session, when the definition
is either static or signaled through SDP. There however do exist
cases where each end-point have different sets of payload types due
to SDP offer/answer.
Although Payload Type definitions are commonly local to an RTP Position: In every RTP header and in SDP signalling.
Session, there are some uses where Payload Type numbers need be
unique across RTP Sessions. This is for example the case in Media
Decoding Dependency [RFC5583] where Payload Types are used to
describe media dependency across RTP Sessions.
Given that multiple Payload Types are defined in an RTP Session, a Usage: Identify a specific Media Stream encoding format. The
media sender is free to change the Payload Type on a per packet format definition may be taken from [RFC3551] for statically
basis. One example of designed per-packet change of Payload Type is allocated Payload Types, but should be explicitly defined in
a speech codec that makes use of generic Comfort Noise [RFC3389]. signalling, such as SDP, both for static and dynamic Payload
Types. The term "format" here includes whatever can be
described by out-of-band signaling means. In SDP, the term
"format" includes media type, RTP timestamp sampling rate,
codec, codec configuration, payload format configurations, and
various robustness mechanisms such as redundant encodings
[RFC2198].
The RTP Payload Type in RTP is designed such that only a single Uniqueness: Scoped by sending end-point within an RTP Session.
Payload Type is valid at any time instant in the SSRC's timestamp To avoid any potential for ambiguity, it is desirable that
time line, effectively time-multiplexing different Payload Types if payload types are unique across all sending end-points within
any switch occurs. Even when this constraint is met, having an RTP session, but this is often not true in practice. All
different rates on the RTP timestamp clock for the RTP Payload Types SSRC in an RTP session sent from an single end-point share the
in use in the same RTP Session have issues such as loss of same Payload Types definitions. The RTP Payload Type is
synchronization. Payload Type clock rate switching requires some designed such that only a single Payload Type is valid at any
special consideration that is described in the multiple clock rates time instant in the SSRC's RTP timestamp time line, effectively
specification [I-D.ietf-avtext-multiple-clock-rates]. time-multiplexing different Payload Types if any change occurs.
Used Payload Type may change on a per-packet basis for an SSRC,
for example a speech codec making use of generic Comfort Noise
[RFC3389].
Inter-relation: There are some uses where Payload Type numbers
need be unique across RTP Sessions. This is for example the
case in Media Decoding Dependency [RFC5583] where Payload Types
are used to describe media dependency across RTP Sessions.
Another example is session-based RTP retransmission [RFC4588].
Special Restrictions: Using different RTP timestamp clock rates for
the RTP Payload Types in use in the same RTP Session have issues
such as loss of synchronization. Payload Type clock rate
switching requires some special consideration that is described in
the multiple clock rates specification
[I-D.ietf-avtext-multiple-clock-rates].
If there is a true need to send multiple Payload Types for the same If there is a true need to send multiple Payload Types for the same
SSRC that are valid for the same RTP Timestamps, then redundant SSRC that are valid for the same RTP Timestamps, then redundant
encodings [RFC2198] can be used. Several additional constraints than encodings [RFC2198] can be used. Several additional constraints than
the ones mentioned above need to be met to enable this use, one of the ones mentioned above need to be met to enable this use, one of
which are that the combined payload sizes of the different Payload which is that the combined payload sizes of the different Payload
Types must not exceed the transport MTU. Types must not exceed the transport MTU.
Other aspects of RTP payload format use are described in RTP Payload Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto]. HowTo [I-D.ietf-payload-rtp-howto].
4. Multiple Streams Alternatives 4. Multiple Streams Alternatives
This section reviews the alternatives to enable multi-stream This section reviews the alternatives to enable multi-stream
handling. Let's start with describing mechanisms that could enable handling. Let's start with describing mechanisms that could enable
multiple media streams, independent of the purpose for having multiple media streams, independent of the purpose for having
skipping to change at page 9, line 50 skipping to change at page 10, line 48
within a RTP Session. within a RTP Session.
Session Multiplexing: Using additional RTP Sessions to handle Session Multiplexing: Using additional RTP Sessions to handle
additional Media Streams additional Media Streams
Payload Type Multiplexing: Using different RTP payload types for Payload Type Multiplexing: Using different RTP payload types for
different additional streams. different additional streams.
Independent of the reason to use additional media streams, achieving Independent of the reason to use additional media streams, achieving
it using payload type multiplexing is not a good choice as can be it using payload type multiplexing is not a good choice as can be
seen in the below section (Section 6). The RTP payload type alone is seen in the Appendix A. The RTP payload type alone is not suitable
not suitable for cases where additional media streams are required. for cases where additional media streams are required. Streams need
Streams need their own SSRCs, so that they get their own sequence their own SSRCs, so that they get their own sequence number space.
number space. The SSRC itself is also important so that the media The SSRC itself is also important so that the media stream can be
stream can be referenced and reported on. referenced and reported on.
This leaves us with two choices, either using SSRC multiplexing to This leaves us with two main choices, either using SSRC multiplexing
have multiple SSRCs from one end-point in one RTP session, or create to have multiple SSRCs from one end-point in one RTP session, or
additional RTP sessions to hold that additional SSRC. As the below create an additional RTP session to hold that additional SSRC. As
discussion will show, in reality we cannot choose a single one of the the below discussion will show, in reality we cannot choose a single
two solutions. To utilize RTP well and as efficiently as possible, one of the two solutions. To utilize RTP well and as efficiently as
both are needed. The real issue is finding the right guidance on possible, both are needed. The real issue is finding the right
when to create RTP sessions and when additional SSRCs in an RTP guidance on when to create RTP sessions and when additional SSRCs in
session is the right choice. an RTP session is the right choice.
In the below discussion, please keep in mind that the reasons for In the below discussion, please keep in mind that the reasons for
having multiple media streams vary and include but are not limited to having multiple media streams vary and include but are not limited to
the following: the following:
o Multiple Media Sources of the same media type o Multiple Media Sources
o Retransmission streams o Retransmission streams
o FEC stream o FEC stream
o Alternative Encoding o Alternative Encodings
o Scalability layer o Scalability layers
Thus the choice made due to one reason may not be the choice suitable Thus the choice made due to one reason may not be the choice suitable
for another reason. In the above list, the different items have for another reason. In the above list, the different items have
different levels of maturity in the discussion on how to solve them. different levels of maturity in the discussion on how to solve them.
The clearest understanding is associated with multiple media sources The clearest understanding is associated with multiple media sources
of the same media type. However, all warrant discussion and of the same media type. However, all warrant discussion and
clarification on how to deal with them. clarification on how to deal with them.
5. RTP Topologies and Issues 5. RTP Topologies and Issues
skipping to change at page 11, line 7 skipping to change at page 12, line 7
attempts to highlight the important behaviors concerning RTP attempts to highlight the important behaviors concerning RTP
multiplexing and multi-stream handling. It lists any identified multiplexing and multi-stream handling. It lists any identified
issues regarding RTP and RTCP handling, and introduces additional issues regarding RTP and RTCP handling, and introduces additional
topologies that are supported by RTP beyond those included in RTP topologies that are supported by RTP beyond those included in RTP
Topologies [RFC5117]. The RTP Topologies that do not follow the RTP Topologies [RFC5117]. The RTP Topologies that do not follow the RTP
specification or do not attempt to utilize the facilities of RTP are specification or do not attempt to utilize the facilities of RTP are
ignored in this document. ignored in this document.
5.1. Point to Point 5.1. Point to Point
This is the most basic use case with an RTP session containing of two This is the most basic use case with an RTP session containing two
end-points. Each end-point has one or more SSRCs. end-points. Each end-point has one or more SSRCs.
+---+ +---+ +---+ +---+
| A |<------->| B | | A |<------->| B |
+---+ +---+ +---+ +---+
Point to Point Figure 1: Point to Point
5.1.1. RTCP Reporting 5.1.1. RTCP Reporting
In cases when an end-point uses multiple SSRCs, we have found two In cases when an end-point uses multiple SSRCs, we have found two
closely related issues. The first is if every SSRC shall report on closely related issues. The first is if every SSRC shall report on
all other SSRC, even the ones originating from the same end-point. all other SSRC, even the ones originating from the same end-point.
The reason for this would be ensure that no monitoring function The reason for this would be to ensure that no monitoring function
should suspect a breakage in the RTP session. should suspect a breakage in the RTP session.
The second issue around RTCP reporting arise when an end-point The second issue around RTCP reporting arise when an end-point
receives one or more media streams, and when the receiving end-point receives one or more media streams, and when the receiving end-point
itself sends multiple SSRC in the same RTP session. As transport itself sends multiple SSRC in the same RTP session. As transport
statistics are gathered per end-point and shared between the nodes, statistics are gathered per end-point and shared between the nodes,
all the end-point's SSRC will report based on the same received data, all the end-point's SSRC will report based on the same received data,
the only difference will be which SSRCs sends the report. This could the only difference will be which SSRCs sends the report. This could
be considered unnecessary overhead, but for consistency it might be be considered unnecessary overhead, but for consistency it might be
simplest to always have all sending SSRCs send RTCP reports on all simplest to always have all sending SSRCs send RTCP reports on all
media streams the end-point receives. media streams the end-point receives.
The current RTP text is silent about sending RTCP Receiver Reports The current RTP text is silent about sending RTCP Receiver Reports
for an endpoint's own sources, but does not preclude either sending for an endpoint's own sources, but does not preclude either sending
or omitting them. The uncertainty in the expected behavior in those or omitting them. The uncertainty in the expected behavior in those
cases have likely caused variations in the implementation strategy. cases has likely caused variations in the implementation strategy.
This could cause an interoperability issue where it is not possible This could cause an interoperability issue where it is not possible
to determine if the lack of reports are a true transport issue, or to determine if the lack of reports is a true transport issue, or
simply a result of implementation. simply a result of implementation.
Although this issue is valid already for the simple point to point Although this issue is valid already for the simple point to point
case, it needs to be considered in all topologies. From the case, it needs to be considered in all topologies. From the
perspective of an end-point, any solution needs to take into account perspective of an end-point, any solution needs to take into account
what a particular end-point can determine without explicit what a particular end-point can determine without explicit
information of the topology. For example, a Transport Translator information of the topology. For example, a Transport Translator
(Relay) topology will look quite similar as point to point on an RTP (Relay) topology will look quite similar to point to point on a
level but is different. The main difference between a point to point transport level but is different on RTP level. Assume a first
with two SSRC being sent from the remote end-point and a Transport scenario with two SSRC being sent from an end-point to a Transport
Translator with two single SSRC remote clients are that the RTT may Translator, and a second scenario with two single SSRC remote end-
vary between the SSRCs (but it is not guaranteed), and that the SSRCs points sending to the same Transport Translator. The main
may have different CNAMEs. differences between those two scenarios are that in the second
scenario, the RTT may vary between the SSRCs (but it is not
guaranteed), and the SSRCs may also have different CNAMEs.
5.1.2. Compound RTCP Packets 5.1.2. Compound RTCP Packets
When an end-point has multiple SSRCs and it needs to send RTCP When an end-point has multiple SSRCs and it needs to send RTCP
packets on behalf of these SSRCs, the question arises if and how RTCP packets on behalf of these SSRCs, the question arises if and how RTCP
packets with different source SSRCs can be sent in the same compound packets with different source SSRCs can be sent in the same compound
packet. If it is allowed, then some consideration of the packet. If it is allowed, then some consideration of the
transmission scheduling is needed. transmission scheduling is needed.
5.2. Point to Multipoint Using Multicast 5.2. Point to Multipoint Using Multicast
This section discusses the Point to Multi-point using Multicast to This section discusses the Point to Multi-point using Multicast to
interconnect the session participants. This needs to consider both interconnect the session participants. This needs to consider both
Any Source Multicast (ASM) and Source-Specific Multicast (SSM). Any Source Multicast (ASM) and Source-Specific Multicast (SSM).
There are large commercial deployments of multicast for applications
like IPTV.
+-----+ +-----+
+---+ / \ +---+ +---+ / \ +---+
| A |----/ \---| B | | A |----/ \---| B |
+---+ / Multi- \ +---+ +---+ / Multi- \ +---+
+ Cast + + Cast +
+---+ \ Network / +---+ +---+ \ Network / +---+
| C |----\ /---| D | | C |----\ /---| D |
+---+ \ / +---+ +---+ \ / +---+
+-----+ +-----+
Point to Multipoint Using Any Source Multicast Figure 2: Point to Multipoint Using Any Source Multicast
In Any Source Multicast, any of the participants can send to all the In Any Source Multicast, any of the participants can send to all the
other participants, simply by sending a packet to the multicast other participants, simply by sending a packet to the multicast
group. That is not possible in Source Specific Multicast [RFC4607] group. That is not possible in Source Specific Multicast [RFC4607]
where only a single source (Distribution Source) can send to the where only a single source (Distribution Source) can send to the
multicast group, creating a topology that looks like the one below: multicast group, creating a topology that looks like the one below:
Source-specific +--------+ +-----+
+--------+ +-----+ Multicast |Media | | | Source-specific
|Media | | | +----------------> R(1) |Sender 1|<----->| D S | Multicast
|Sender 1|<----->| D S | | | +--------+ | I O | +--+----------------> R(1)
+--------+ | I O | +--+ |
| S U | | | | | S U | | | |
+--------+ | T R | | +-----------> R(2) | +--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ +---- : | | |Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | | |Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | | +--------+ | B | | | | | |
: | U | +--+--> R(n) | | | : | U | +--+--> R(n) | | |
: | T +-| | | | | : | T +-| | | | |
| I | |<---------+ | | | : | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | | +--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ | |Media | | N |T|<--------------------+ |
|Sender M|<----->| | |<-------------------------+ |Sender M|<----->| | |<-------------------------+
+--------+ +-----+ Unicast +--------+ +-----+ RTCP Unicast
FT = Feedback Target FT = Feedback Target
Transport from the Feedback Target to the Distribution Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not Source is via unicast or multicast RTCP if they are not
co-located. co-located.
Point to Multipoint using Source Specific Multicast Figure 3: Point to Multipoint using Source Specific Multicast
In this topology a number of Media Senders (1 to M) are allowed to In this topology a number of Media Senders (1 to M) are allowed to
send media to the SSM group, sends media to the distribution source send media to the SSM group, sends media to the distribution source
which then forwards the media streams to the multicast group. The which then forwards the media streams to the multicast group. The
media streams reach the Receivers (R(1) to R(n)). The Receiver's media streams reach the Receivers (R(1) to R(n)). The Receiver's
RTCP cannot be sent to the multicast group. To support RTCP, an RTP RTCP cannot be sent to the multicast group. To support RTCP, an RTP
extension for SSM [RFC5760] was defined that use unicast transmission extension for SSM [RFC5760] was defined to use unicast transmission
to send RTCP from the receivers to one or more Feedback Targets (FT). to send RTCP from the receivers to one or more Feedback Targets (FT).
As multicast is a one to many distribution system this must be taken As multicast is a one to many distribution system, this must be taken
into consideration. For example, the only practical method for into consideration. For example, the only practical method for
adapting the bit-rate sent towards a given receiver is to use a set adapting the bit-rate sent towards a given receiver for large groups
of multicast groups, where each multicast group represents a is to use a set of multicast groups, where each multicast group
particular bit-rate. The media encoding is either scalable, where represents a particular bit-rate. Otherwise the whole group gets
multiple layers can be combined, or simulcast where a single version media adapted to the participant with the worst conditions. The
is selected. By either selecting or combing multicast groups, the media encoding is either scalable, where multiple layers can be
receiver can control the bit-rate sent on the path to itself. It is combined, or simulcast where a single version is selected. By either
also common that transport robustification is sent in its own selecting or combing multicast groups, the receiver can control the
multicast group to allow for interworking with legacy or to support bit-rate sent on the path to itself. It is also common that streams
different levels of protection. that improve transport robustness is sent in its own multicast group
to allow for interworking with legacy or to support different levels
of protection.
The result of this is three common behaviors for RTP multicast: The result of this is three common behaviors for RTP multicast:
1. Use of multiple RTP sessions for the same media type. 1. Use of multiple RTP sessions for the same media type.
2. The need for identifying RTP sessions that are related in one of 2. The need for identifying RTP sessions that are related in one of
several ways. several possible ways.
3. The need for binding related SSRCs in different RTP sessions 3. The need for binding related SSRCs in different RTP sessions
together. together.
This indicates that Multicast is an important consideration when This indicates that Multicast is an important consideration when
working with the RTP multiplexing and multi stream architecture working with the RTP multiplexing and multi stream architecture
questions. It is also important to note that so far there is no questions. It is also important to note that so far there is no
special mode for basic behavior between multicast and unicast usages special mode for basic behavior between multicast and unicast usages
of RTP. Yes, there are extensions targeted to deal with multicast of RTP. Yes, there are extensions targeted to deal with multicast
specific cases but the general applicability does need to be specific cases, but the general applicability does need to be
considered. considered.
5.3. Point to Multipoint Using an RTP Translator 5.3. Point to Multipoint Using an RTP Translator
Transport Translators (Relays) are a very important consideration for Transport Translators (Relays) are a very important consideration for
this document as they result in an RTP session situation that is very this document as they result in an RTP session situation that is very
similar to how an ASM group RTP session would behave. similar to how an ASM group RTP session would behave.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Translator | | Translator |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Transport Translator (Relay) Figure 4: Transport Translator (Relay)
One of the most important aspects with the simple relay is that it is One of the most important aspects with the simple relay is that it is
both easy to implement and require minimal amount of resources as both easy to implement and require minimal amount of resources as
only transport headers are rewritten, no RTP modifications nor media only transport headers are rewritten, no RTP modifications nor media
transcoding occur. Thus it is most likely the cheapest and most transcoding occur. Thus it is most likely the cheapest and most
generally deployable method for multi-point sessions. The most generally deployable method for multi-point sessions. The most
obvious downside of this basic relaying is that the translator has no obvious downside of this basic relaying is that the translator has no
control over how many streams needs to be delivered to a receiver. control over how many streams needs to be delivered to a receiver.
Nor can it simply select to deliver only certain streams, at least Nor can it simply select to deliver only certain streams, as it
not without new RTCP extensions to coherently handle the fact that creates session inconsistencies. If some middlebox temporarily stops
some middlebox temporarily stops a stream, preventing some receivers a stream, this prevents some receivers from reporting on it. From
from reporting on it. This consistency problem in RTCP reporting the senders perspective it will look like a transport failure.
needs to be handled. Applications having needs to stop or switch streams in the central
node should consider using an RTP mixer to avoid this issue.
The Transport Translator does not need to have an SSRC of itself, nor The Transport Translator does not need to have an SSRC of itself, nor
need it send any RTCP reports on the flows that passes it, but it may need it send any RTCP reports on the flows that pass it, but it may
choose to do that. choose to do that.
Use of a transport translator results in that any of the end-points Use of a transport translator results in that any of the end-points
will receive multiple SSRCs over a single unicast transport flow from will receive multiple SSRCs over a single unicast transport flow from
the translator. That is independent of the other end-points having the translator. That is independent of the other end-points having
only a single or several SSRCs. End-points that have multiple SSRCs only a single or several SSRCs. End-points that have multiple SSRCs
put further requirements on how SSRCs can be related or bound within put further requirements on how SSRCs can be related or bound within
and across RTP sessions and how they can be identified on an and across RTP sessions and how they can be identified on an
application level. application level. The transport translator has a signalling
requirement that also exist in any source multicast; all of the
participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It should
be noted that SDP offer/answer [RFC3264] has issues with ensuring
this property.
A Media Translator can perform a large variety of media functions A Media Translator can perform a large variety of media functions
affecting the media stream passing the translator, coming from one affecting the media stream passing the translator, coming from one
source and destined to a particular end-point. The media stream can source and destined to a particular end-point. The translator can
be transcoded to a different bit-rate, change to another encoder, transcode to a different bit-rate, transcode to use another encoder,
change the packetization of the media stream, add FEC streams, or change the packetization of the media stream, add FEC streams, or
terminate RTP retransmissions. The latter behaviors require the terminate RTP retransmissions. The latter behaviors require the
translator to use SSRCs that only exist in a particular sub-domain of translator to use SSRCs that only exist in a particular sub-domain of
the RTP session, and it may also create additional sessions when the the RTP session, and it may also create additional sessions when the
mechanism applied on one side so requires. mechanism applied on one side so requires.
5.4. Point to Multipoint Using an RTP Mixer 5.4. Point to Multipoint Using an RTP Mixer
The most commonly used topology in centralized conferencing is based The most commonly used topology in centralized conferencing is based
on the RTP Mixer. The main reason for this is that it provides a on the RTP Mixer. The main reason for this is that it provides a
skipping to change at page 15, line 44 skipping to change at page 16, line 51
underlying sources based on some mixer policy or control signalling. underlying sources based on some mixer policy or control signalling.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
RTP Mixer Figure 5: RTP Mixer
In the case where the mixer does stream selection, an application may In the case where the mixer does stream selection, an application may
in fact desire multiple simultaneous streams but only as many as the in fact desire multiple simultaneous streams but only as many as the
mixer can handle. As long as the mixer and an end-point can agree on mixer can handle. As long as the mixer and an end-point can agree on
the maximum number of streams and how the streams that are delivered the maximum number of streams and how the streams that are delivered
are selected, this provides very good functionality. As these are selected, this provides very good functionality. As these
streams are forwarded using the mixer's SSRCs, there are no streams are forwarded using the mixer's SSRCs, there are no
inconsistencies within the session. inconsistencies within the session.
5.5. Point to Multipoint using Multiple Unicast flows 5.5. Point to Multipoint using Multiple Unicast flows
skipping to change at page 16, line 25 skipping to change at page 17, line 33
| A |<---->| B | | A |<---->| B |
+---+ +---+ +---+ +---+
^ ^ ^ ^
\ / \ /
\ / \ /
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Point to Multi-Point using Multiple Unicast Transprots Figure 6: Point to Multi-Point using Multiple Unicast Transports
This doesn't create any additional requirements beyond the need to This doesn't create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session. have multiple transport flows associated with a single RTP session.
Note that an end-point may use a single local port to receive all Note that an end-point may use a single local port to receive all
these transport flows, or it might have separate local reception these transport flows, or it might have separate local reception
ports for each of the end-points. ports for each of the end-points.
5.6. Decomposited End-Point There exists an alternative structure for establishing the above
communication scenario (Figure 6) which uses independent RTP sessions
between each pair of peers, i.e. three different RTP sessions.
Unless independently adapted the same RTP media stream could be sent
in both of the RTP sessions an end-point has. The difference exists
in the behaviors around RTCP, for example common RTCP bandwidth for
one joint session, rather than three independent pools, and the
awareness based on RTCP reports between the peers of how that third
leg is doing.
5.6. De-composite End-Point
There is some possibility that an RTP end-point implementation in There is some possibility that an RTP end-point implementation in
fact reside on multiple devices, each with their own network address. fact reside on multiple devices, each with their own network address.
A very basic use case for this would be to separate audio and video A very basic use case for this would be to separate audio and video
processing for a particular end-point, like a conference room, into processing for a particular end-point, like a conference room, into
one device handling the audio and another handling the video being one device handling the audio and another handling the video, being
interconnected by some control functions allowing them to behave as a interconnected by some control functions allowing them to behave as a
single end-point. single end-point.
+---------------------+ +---------------------+
| End-point A | | End-point A |
| Local Area Network | | Local Area Network |
| +------------+ | | +------------+ |
| +->| Audio |<+----\ | +->| Audio |<+----\
| | +------------+ | \ +------+ | | +------------+ | \ +------+
| | +------------+ | +-->| | | | +------------+ | +-->| |
| +->| Video |<+--------->| B | | +->| Video |<+--------->| B |
| | +------------+ | +-->| | | | +------------+ | +-->| |
| | +------------+ | / +------+ | | +------------+ | / +------+
| +->| Control |<+----/ | +->| Control |<+----/
| +------------+ | | +------------+ |
+---------------------+ +---------------------+
Decomposited End-Point Figure 7: De-composite End-Point
In the above usage, let us assume that the RTP sessions are different In the above usage, let us assume that the RTP sessions are different
for audio and video. The audio and video parts will use a common for audio and video. The audio and video parts will use a common
CNAME and also have a common clock to ensure that synchronization and CNAME and also have a common clock to ensure that synchronization and
clock drift handling works despite the decomposition. However, if clock drift handling works despite the decomposition.
the audio and video were in a single RTP session then this use case
becomes problematic. This as all transport flow receivers will need However, if the audio and video were in a single RTP session then
to receive all the other media streams that are part of the session. this use case becomes problematic. This as all transport flow
Thus the audio component will receive also all the video media receivers will need to receive all the other media streams that are
streams, while the video component will receive all the audio ones, part of the session. Thus the audio component will receive also all
thus doubling the site's bandwidth requirements from all other the video media streams, while the video component will receive all
session participants. With a joint RTP session it also becomes the audio ones, doubling the site's bandwidth requirements from all
other session participants. With a joint RTP session it also becomes
evident that a given end-point, as interpreted from a CNAME evident that a given end-point, as interpreted from a CNAME
perspective, has two sets of transport flows for receiving the perspective, has two sets of transport flows for receiving the
streams and the decomposition isn't hidden. streams and the decomposition is not hidden.
The requirements that can derived from the above usage is that the The requirements that can derived from the above usage is that the
transport flows for each RTP session might be under common control transport flows for each RTP session might be under common control
but still go to what looks like different end-points based on but still go to what looks like different end-points based on
addresses and ports. A conclusion can also be reached that addresses and ports. A conclusion can also be reached that
decomposition without using separate RTP sessions has downsides and decomposition without using separate RTP sessions has downsides and
potential for RTP/RTCP issues. potential for RTP/RTCP issues.
There exist another use case which might be considered as a There exist another use case which might be considered as a de-
decomposited end-point. However, as will be shown this should be composite end-point. However, as will be shown this should be
considered a translator instead. An example of this is when an end- considered a translator instead. An example of this is when an end-
point A sends a media flow to B. On the path there is a device C that point A sends a media flow to B. On the path there is a device C that
on A's behalf does something with the media streams, for example adds on A's behalf does something with the media streams, for example adds
an RTP session with FEC information for A's media streams. C will in an RTP session with FEC information for A's media streams. C will in
this case need to bind the new FEC streams to A's media stream by this case need to bind the new FEC streams to A's media stream by
using the same CNAME as A. using the same CNAME as A.
+------+ +------+ +------+ +------+ +------+ +------+
| | | | | | | | | | | |
| A |------->| C |-------->| B | | A |------->| C |-------->| B |
| | | |---FEC-->| | | | | |---FEC-->| |
+------+ +------+ +------+ +------+ +------+ +------+
When Decomposition is a Translator Figure 8: When De-composition is a Translator
This type of functionality where C does something with the media This type of functionality where C does something with the media
stream on behalf of A is clearly covered under the media translator stream on behalf of A is clearly covered under the media translator
definition (Section 5.3). definition (Section 5.3).
6. Dismissing Payload Type Multiplexing 6. Multiple Streams Discussion
Before starting a discussion on when to use what alternative, we will
first document a number of reasons why using the payload type as a
multiplexing point for anything related to multiple streams is
unsuitable and will not be considered further.
If one attempts to use Payload type multiplexing beyond it's defined
usage, that has well known negative effects on RTP. To use Payload
type as the single discriminator for multiple streams implies that
all the different media streams are being sent with the same SSRC,
thus using the same timestamp and sequence number space. This has
many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not
possible, Payload Type multiplexing cannot be used.
2. Many RTP payload formats may fragment a media object over
multiple packets, like parts of a video frame. These payload
formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all
fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in
sequence.
3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/
T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before
being used with Payload Type multiplexing.
4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which
stream a packet loss relates to.
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
sequence number, not by Payload Type. If only some of the
Payload Type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets must be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported,
so sending feedback for a specific media stream is difficult
without extending existing RTCP reporting.
7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing.
8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number
of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this
is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing.
11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.
12. A legacy end-point that doesn't understand the indication that
different RTP payload types are different media streams may be
slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types.
7. Multiple Streams Discussion
7.1. Introduction 6.1. Introduction
Using multiple media streams is a well supported feature of RTP. Using multiple media streams is a well supported feature of RTP.
However, what can be unclear for most implementors or people writing However, it can be unclear for most implementers or people writing
RTP/RTCP extensions attempting to apply multiple streams, is when it RTP/RTCP applications or extensions attempting to apply multiple
is most appropriate to add an additional SSRC in an existing RTP streams when it is most appropriate to add an additional SSRC in an
session and when it is better to use multiple RTP sessions. This existing RTP session and when it is better to use multiple RTP
section tries to discuss the various considerations needed. The next sessions. This section tries to discuss the various considerations
section then concludes with some guidelines. needed. The next section then concludes with some guidelines.
7.2. RTP/RTCP Aspects 6.2. RTP/RTCP Aspects
This section discusses RTP and RTCP aspects worth considering when This section discusses RTP and RTCP aspects worth considering when
selecting between SSRC multiplexing and Session multiplexing. selecting between SSRC multiplexing and Session multiplexing.
7.2.1. The RTP Specification 6.2.1. The RTP Specification
RFC 3550 contains some recommendations and a bullet list with 5 RFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Let's review arguments for different aspects of RTP multiplexing. Let's review
Section 5.2 of [RFC3550], reproduced below: Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points "For efficient protocol processing, the number of multiplexing points
should be minimized, as described in the integrated layer processing should be minimized, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference is different for each RTP session. For example, in a teleconference
skipping to change at page 21, line 37 skipping to change at page 21, line 4
RTP session would avoid the first three problems but not the last RTP session would avoid the first three problems but not the last
two. two.
On the other hand, multiplexing multiple related sources of the same On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It may also be appropriate treatment is applicable for all of them. It may also be appropriate
to multiplex streams of the same medium using different SSRC values to multiplex streams of the same medium using different SSRC values
in other scenarios where the last two problems do not apply." in other scenarios where the last two problems do not apply."
Let's consider one argument at a time. The first is an argument for Let's consider one argument at a time. The first is an argument for
using different SSRC for each individual media stream, which still is using different SSRC for each individual media stream, which still is
very applicable. very applicable.
The second argument is advocating against using payload type The second argument is advocating against using payload type
multiplexing, which still stands as can been seen by the extensive multiplexing, which still stands as can been seen by the extensive
list of issues found in Section 6. list of issues found in Appendix A.
The third argument is yet another argument against payload type The third argument is yet another argument against payload type
multiplexing. multiplexing.
The fourth is an argument against multiplexing media streams that The fourth is an argument against multiplexing media streams that
require different handling into the same session. This is to require different handling into the same session. This is to
simplify the processing at any receiver of the media stream. If all simplify the processing at any receiver of the media stream. If all
media streams that exist in an RTP session is of one media type and media streams that exist in an RTP session are of one media type and
one particular purpose, there is no need for deeper inspection of the one particular purpose, there is no need for deeper inspection of the
packets before processing them in both end-points and RTP aware packets before processing them in both end-points and RTP aware
middle nodes. middle nodes.
The fifth argument discusses network aspects that we will discuss The fifth argument discusses network aspects that we will discuss
more below in Section 7.4. It also goes into aspects of more below in Section 6.5. It also goes into aspects of
implementation, like decomposed end-points where different processes implementation, like decomposed end-points where different processes
or inter-connected devices handle different aspects of the whole or inter-connected devices handle different aspects of the whole
multi-media session. multi-media session.
A summary of RFC 3550's view on multiplexing is to use unique SSRCs A summary of RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its' own media/packet stream, and secondly use for anything that is its' own media/packet stream, and secondly use
different RTP sessions for media streams that don't share media type different RTP sessions for media streams that don't share media type
and purpose, to maximize flexibility when it comes to processing and and purpose, to maximize flexibility when it comes to processing and
handling of the media streams. handling of the media streams.
This mostly agrees with the discussion and recommendations in this This mostly agrees with the discussion and recommendations in this
document. However, there has been an evolution of RTP since that document. However, there has been an evolution of RTP since that
text was written which needs to be reflected in the discussion. text was written which needs to be reflected in the discussion.
Additional clarifications for specific cases are also needed. Additional clarifications for specific cases are also needed.
7.2.2. Multiple SSRC Legacy Considerations 6.2.1.1. Different Media Types Recommendations
When establishing RTP sessions that may contain end-points that
aren't updated to handle multiple streams following these
recommendations, a particular application can have issues with
multiple SSRCs within a single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
RTP Session multiplexing could potentially avoid these issues if The above quote from RTP [RFC3550] includes a strong recommendation:
there is only a single SSRC at each end-point, and in topologies
which appears like point to point as seen the end-point. However,
forcing the usage of session multiplexing due to this reason would be
a great mistake, as it is likely that a significant set of
applications will need a combination of SSRC multiplexing of several
media sources and session multiplexing for other aspects such as
encoding alternatives, robustification or simply to support legacy.
However, this issue does need consideration when deploying multiple
media streams within an RTP session where legacy end-points may
occur.
7.2.3. RTP Specification Clarifications Needed "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."
The RTP specification contains a few things that are potential It has been identified in "Why RTP Sessions Should Be Content
interoperability issues when using multiple SSRCs within a session. Neutral" [I-D.alvestrand-rtp-sess-neutral] that the above statement
These issues are described and discussed in Section 9. These should is poorly supported by any of the motivations provided in the RTP
not be considered strong arguments against using SSRC multiplexing specification. This document has a more detailed analysis of
when otherwise appropriate, and there are some issues we expect to be potential issues in having multiple media types in the same RTP
solved in the near future. session in Section 6.7. An important influence for underlying
thinking for the RTP design and likely this statement can be found in
the academic paper by David Clark and David Tennenhouse
"Architectural considerations for a new generation of protocols"
[ALF].
7.2.4. Handling Varying sets of Senders 6.2.2. Handling Varying sets of Senders
Another potential issue that needs to be considered is where a A potential issue that some application designers may need to
limited set of simultaneously active sources varies within a larger consider is the case where the set of simultaneously active sources
set of session members. As each media decoding chain may contain varies within a larger set of session members. As each media
state, it is important that this type of usage ensures that a decoding chain may contain state, it is important that this type of
receiver can flush a decoding state for an inactive source and if usage ensures that a receiver can flush a decoding state for an
that source becomes active again, it does not assume that this inactive source and if that source becomes active again, it does not
previous state exists. assume that this previous state exists.
This behavior might in certain applications be possible to limit to a This behavior will cause similar issues independent of SSRC or
particular RTP Session and instead use multiple RTP sessions. But in Session multiplexing. It might be possible in certain applications
some cases it is likely unavoidable and the most appropriate thing is to limit the changes to a subset of communication session
to SSRC multiplex. participants by have the sub-set use particular RTP Sessions.
7.2.5. Cross Session RTCP requests 6.2.3. Cross Session RTCP Requests
There currently exist no functionality to make truly synchronized and There currently exists no functionality to make truly synchronized
atomic RTCP requests across multiple RTP Sessions. Instead separate and atomic RTCP messages with some type of request semantics across
RTCP messages will have to be sent in each session. This gives SSRC multiple RTP Sessions. Instead, separate RTCP messages will have to
multiplexed streams a slight advantage as RTCP requests for different be sent in each session. This gives SSRC multiplexed streams a
streams in the same session can be sent in a compound RTCP packet. slight advantage as RTCP messages for different streams in the same
Thus providing an atomic operation if different modifications of session can be sent in a compound RTCP packet. Thus providing an
different streams are requested at the same time. atomic operation if different modifications of different streams are
requested at the same time.
In Session multiplexed cases, the RTCP timing rules in the sessions In Session multiplexed cases, the RTCP timing rules in the sessions
and the transport aspects, such as packet loss and jitter, prevents a and the transport aspects, such as packet loss and jitter, prevents a
receiver from relying on atomic operations, instead more robust and receiver from relying on atomic operations, forcing it to use more
forgiving mechanisms need to be used. robust and forgiving mechanisms.
7.2.6. Binding Related Sources 6.2.4. Binding Related Sources
A common problem in a number of various RTP extensions has been how A common problem in a number of various RTP extensions has been how
to bind together related sources. This issue is common independent to bind related sources together. This issue is common to SSRC
of SSRC multiplexing and Session Multiplexing, and any solution and multiplexing and Session Multiplexing, and any solution and
recommendation to the problem should work equally well for both to recommendation related to the problem should work equally well with
avoid creating barriers between using session multiplexing and SSRC both methods to avoid creating barriers between using session
multiplexing. multiplexing and SSRC multiplexing.
The current solutions don't have these properties. There exist one The current solutions do not have these properties. There exists one
solution for grouping RTP session together in SDP [RFC5888] to know solution for grouping RTP session together in SDP [RFC5888] to know
which RTP session contains for example the FEC data for the source which RTP session contains for example the FEC data for the source
data in another session. However, this mechanism does not work on data in another session. However, this mechanism does not work on
individual media flows and is thus not directly applicable to the individual media flows and is thus not directly applicable to the
problem. The other solution is also SDP based and can group SSRCs problem. The other solution is also SDP based and can group SSRCs
within a single RTP session [RFC5576]. Thus this mechanism can bind within a single RTP session [RFC5576]. Thus this mechanism can bind
media streams in SSRC multiplexed cases. Both solutions have the media streams in SSRC multiplexed cases. Both solutions have the
shortcoming of being restricted to SDP based signalling and also do shortcoming of being restricted to SDP based signalling and also do
not work in cases where the session's dynamic properties are such not work in cases where the session's dynamic properties are such
that it is difficult or resource consuming to keep the list of that it is difficult or resource consuming to keep the list of
skipping to change at page 24, line 48 skipping to change at page 23, line 46
This will prevent also the media streams not having an actual This will prevent also the media streams not having an actual
collision from being usable during the re-synchronization and also collision from being usable during the re-synchronization and also
increases the time until synchronization is finalized. In addition, increases the time until synchronization is finalized. In addition,
it requires exception handling in the SSRC generation. it requires exception handling in the SSRC generation.
The above collision issue does not occur in case of having only one The above collision issue does not occur in case of having only one
SSRC space across all sessions and all participants will be part of SSRC space across all sessions and all participants will be part of
at least one session, like the base layer in layered encoding. In at least one session, like the base layer in layered encoding. In
that case the only downside is the special behavior that needs to be that case the only downside is the special behavior that needs to be
well defined by anyone using this. But, having an exception behavior well defined by anyone using this. But, having an exception behavior
where the SSRC space is common across all session an that doesn't fit where the SSRC space is common across all session is an issue as this
all the RTP extensions or payload formats present in the sessions is behavior does not fit all the RTP extensions or payload formats. It
a issue. It is possible to create a situation where the different is possible to create a situation where the different mechanisms
mechanisms can't be combined due to the non standard SSRC allocation cannot be combined due to the non standard SSRC allocation behavior.
behavior.
Existing mechanisms with known issues: Existing mechanisms with known issues:
RTP Retransmission (RFC4588): Has two modes, one for SSRC RTP Retransmission (RFC4588): Has two modes, one for SSRC
multiplexing and one for Session multiplexing. The session multiplexing and one for Session multiplexing. The session
multiplexing requires the same CNAME and mandates that the same multiplexing requires the same CNAME and mandates that the same
SSRC is used in both sessions. Using the same SSRC does work but SSRC is used in both sessions. Using the same SSRC does work but
will potentially have issues in certain cases. In SSRC will potentially have issues in certain cases. In SSRC
multiplexed mode the CNAME is used, and when the first multiplexed mode the CNAME is used to bind media and
retransmission request is sent, one must not have another retransmission streams together. However, if multiple media
retransmission request outstanding for an SSRC which don't have a streams are sent from the same end-point in the same session this
the binding between the original SSRC and the retransmission does not provide non-ambiguous binding. Therefore when the first
stream's SSRC. This works but creates some limitations that can retransmission request for a media stream is sent, one must not
be avoided by a more explicit mechanism. The SDP based ssrc-group have another retransmission request outstanding for an SSRC which
mechanism is sufficient in this case as long as the application don't have a binding between the original SSRC and the
can rely on the signalling based solution. retransmission stream's SSRC. This works but creates some
limitations that can be avoided by a more explicit mechanism. The
SDP based ssrc-group mechanism is sufficient in this case as long
as the application can rely on the signalling based solution.
Scalable Video Coding (RFC6190): As an example of scalable coding, Scalable Video Coding (RFC6190): As an example of scalable coding,
SVC [RFC6190] has various modes. The Multi Session Transmission SVC [RFC6190] has various modes. The Multi Session Transmission
(MST) uses Session multiplexing to separate scalability layers. (MST) uses Session multiplexing to separate scalability layers.
However, this specification has failed to explicit how these However, this specification has failed to be explicit on how these
layers are bound together in cases where CNAME isn't sufficient. layers are bound together in cases where CNAME is not sufficient.
CNAME is no longer sufficient when more than one media source CNAME is no longer sufficient when more than one media source
occur within a session that have the same CNAME, for example due occur within a session that has the same CNAME, for example due to
to multiple video cameras capturing the same lecture hall. This multiple video cameras capturing the same lecture hall. This
likely implies that a single SSRC space as recommend by Section likely implies that a single SSRC space as recommend by Section
8.3 of RTP [RFC3550] is to be used. 8.3 of RTP [RFC3550] is to be used.
Forward Error Correction: If some type of FEC or redundancy stream Forward Error Correction: If some type of FEC or redundancy stream
is being sent, it needs it's own SSRC, with the exception of is being sent, it needs its own SSRC, with the exception of
constructions like redundancy encoding [RFC2198]. Thus in case of constructions like redundancy encoding [RFC2198]. Thus in case of
transmitting the FEC in the same session as the source data, the transmitting the FEC in the same session as the source data, the
inter SSRC relation within a session is needed. In case of inter SSRC relation within a session is needed. In case of
sending the redundant data in a separate session from the source, sending the redundant data in a separate session from the source,
the SSRC in each session needs to be related. This occurs for the SSRC in each session needs to be related. This occurs for
example in RFC5109 when using session separation of original and example in RFC5109 when using session separation of original and
FEC data. SSRC multiplexing is not supported, only using FEC data. SSRC multiplexing is not supported, only using
redundant encoding is supported. redundant encoding is supported.
This issue appears to need action to harmonize and avoid future This issue appears to need action to harmonize and avoid future
shortcomings in extension specifications. A proposed solution for shortcomings in extension specifications. A proposed solution for
handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname]. handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].
7.2.7. Forward Error Correction 6.2.5. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes There exist a number of Forward Error Correction (FEC) based schemes
for how to reduce the packet loss of the original streams. Most of for how to reduce the packet loss of the original streams. Most of
the FEC schemes will protect a single source flow. The protection is the FEC schemes will protect a single source flow. The protection is
achieved by transmitting a certain amount of redundant information achieved by transmitting a certain amount of redundant information
that is encoded such that it can repair one or more packet loss over that is encoded such that it can repair one or more packet loss over
the set of packets they protect. This sequence of redundant the set of packets they protect. This sequence of redundant
information also needs to be transmitted as its own media stream, or information also needs to be transmitted as its own media stream, or
in some cases instead of the original media stream. Thus many of in some cases instead of the original media stream. Thus many of
these schemes creates a need for binding the related flows as these schemes create a need for binding the related flows as
discussed above. They also create additional flows that need to be discussed above. They also create additional flows that need to be
transported. Looking at the history of these schemes, there is both transported. Looking at the history of these schemes, there is both
SSRC multiplexed and Session multiplexed solutions and some schemes SSRC multiplexed and Session multiplexed solutions and some schemes
that support both. that support both.
Using a Session multiplexed solution provides good support for legacy Using a Session multiplexed solution provides good support for legacy
when deploying FEC or changing the scheme used so that some set of when deploying FEC or changing the scheme used, in the sense that it
receivers may not be able to utilize the FEC information. By placing supports the case where some set of receivers may not be able to
it in a separate RTP session, it can easily be ignored. utilize the FEC information. By placing it in a separate RTP
session, it can easily be ignored.
In usages involving multicast, having the FEC information on its own In usages involving multicast, having the FEC information on its own
multicast group and RTP session allows for flexibility, for example multicast group and RTP session allows for flexibility, for example
when using Rapid Acquisition of Multicast Groups (RAMS) [RFC6285]. when using Rapid Acquisition of Multicast Groups (RAMS) [RFC6285].
During the RAMS burst where data is received over unicast and where During the RAMS burst where data is received over unicast and where
it is possible to combine with unicast based retransmission it is possible to combine with unicast based retransmission
[RFC4588], there is no need to burst the FEC data related to the [RFC4588], there is no need to burst the FEC data related to the
burst of the source media streams needed to catch up with the burst of the source media streams needed to catch up with the
multicast group. This saves bandwidth to the receiver during the multicast group. This saves bandwidth to the receiver during the
burst, enabling quicker catch up. When the receiver has catched up burst, enabling quicker catch up. When the receiver has caught up
and joins the multicast group(s) for the source, it can at the same and joins the multicast group(s) for the source, it can at the same
time join the multicast group with the FEC information. Having the time join the multicast group with the FEC information. Having the
source stream and the FEC in separate groups allow for easy source stream and the FEC in separate groups allow for easy
separation in the Burst/Retransmission Source (BRS) without having to separation in the Burst/Retransmission Source (BRS) without having to
individually classify packets. individually classify packets.
7.2.8. Transport Translator Sessions 6.2.6. Transport Translator Sessions
A basic Transport Translator relays any incoming RTP and RTCP packets A basic Transport Translator relays any incoming RTP and RTCP packets
to the other participants. The main difference between SSRC to the other participants. The main difference between SSRC
multiplexing and Session multiplexing resulting from this use case is multiplexing and Session multiplexing resulting from this use case is
that for SSRC multiplexing it is not possible for a particular that for SSRC multiplexing it is not possible for a particular
session participant to decide to receive a subset of media streams. session participant to decide to receive a subset of media streams.
When using separate RTP sessions for the different sets of media When using separate RTP sessions for the different sets of media
streams, a single participant can choose to leave one of the sessions streams, a single participant can choose to leave one of the sessions
but not the other. but not the other.
7.2.9. Multiple Media Types in one RTP session 6.3. Interworking
Having different media types, like audio and video, in the same RTP There are several different kinds of interworking, and this section
sessions is not forbidden, only recommended against as can be seen in discusses two related ones. The interworking between different
Section 7.2.1. When using multiple media types, there are a number applications and the implications of potentially different choices of
of considerations: usage of RTP's multiplexing points. The second topic relates to what
limitations may have to be considered working with some legacy
applications.
Payload Type gives Media Type: This solution is dependent on getting 6.3.1. Interworking Applications
the media type from the Payload Type. Thus overloading this de-
multiplexing point in a receiver for two purposes. First for the
main media type and determining the processing chain, then later
for the exact configuration of the encoder and packetization.
Payload Type field limiations: The total number of Payload Types It is not uncommon that applications or services of similar usage,
available to use in an RTP session is fairly limited, especially especially the ones intended for interactive communication, ends up
if Multiplexing RTP Data and Control Packets on a Single Port in a situation where one want to interconnect two or more of these
[RFC5761] is used. For certain applications negotiating a large applications. From an RTP perspective this could be problem free if
set of codes and configuration may become an issue. all the applications have made the same multiplexing choices, have
the same capabilities in number of simultaneous media streams
combined with the same set of RTP/RTCP extensions being supported.
Unfortunately this may not always be true.
Don't switch media types for an SSRC: The primary reasons to avoid In these cases one ends up in a situation where one might use a
switching from sending for example audio to sending video using gateway to interconnect applications. This gateway then needs to
the same SSRC is the implications on a receiver. When this change the multiplexing structure or adhere to limitations in each
happens, the processing chain in the receiver will have to switch application. If one's goal is to make minimal amount of work in such
from one media type to another. As the different media type's a gateway, there are some multiplexing choices that one should avoid.
entire processing chains are different and are connected to The lowest amount of work represents solutions where one can take an
different outputs it is difficult to reuse the decoding chain, SSRC from one RTP session in one application and forward it into
which a normal codec change likely can. Instead the entire another RTP session. For example if one has one application that has
processing chain has to be torn down and replaced. In addition, multiple SSRCs for one media type in one session and another
there is likely a clock rate switching problem, possibly resulting application that instead has chosen to use multiple RTP sessions with
in synchronization loss at the point of switching media type if only a single SSRC per end-point in each of these sessions. Then
some packet loss occurs. mapping an SSRC from the side with one session into an RTP session is
possible. However mapping SSRC from different RTP sessions into a
single RTP session has the potential of creating SSRC collisions,
especially if an end-point has not generated independent random SSRC
values in each RTP session. This issue is even more likely in a case
where one side uses a single RTP session with multiple media types
and the other uses different RTP session for different media or
robustness mechanism such as retransmission [RFC4588]. Then it is
more likely or maybe even required to use the same SSRC in the
different RTP sessions.
RTCP Bit-rate Issues: If the media types are significantly different In cases where the used structure is incompatible, the gateway will
in bit-rate, the RTCP bandwidth rates assigned to each source in a need to make SSRC translation. Thus this incurs overhead and some
session can result in interesting effects, like that the RTCP bit- potential loss of functionality. First of all, if one translates the
rate share for an audio stream is larger than the actual audio SSRC in an RTP header then one will be forced to decrypt and re-
bit-rate. In itself this doesn't cause any conflicts, only encrypt if one uses SRTP and thus also needs to be part of the
potentially unnecessary overhead. It is possible to avoid this security association. Secondly, changing the SSRC also means that
using AVPF or SAVPF and setting trr-int parameter, which can bring one needs to translate all RTCP messages. This can be more complex,
down unnecessary regular reporting while still allowing for rapid but important so that the gateway does not end up having to terminate
feedback. the end-to-end RTCP chain. In that case the gateway will need to be
able to take the role of a true end-point in each session, which may
include functions such as bit-rate adaptation and correctly respond
to whatever RTCP extensions are being used, and then translate them
or locally respond to them. Thirdly, an SSRC translation may require
that one changes RTP payloads; for example, an RTP retransmission
packet contains an original sequence number that must match the
sequence number used in for the corresponding packet with the new
SSRC. And for FEC packets this is even worse, as the original SSRC
is included as part of the data for which FEC redundant data is
calculated. A fourth issue is the potential for these gateways to
block evolution of the applications by blocking unknown RTP and RTCP
extensions that the regular application has been extended with.
Decomposited end-points: Decomposited nodes that rely on the regular If one uses security functions, like SRTP, they can as seen above
network to separate audio and video to different devices do not incur both additional risk due to the gateway needing to be in
work well with this session setup. If they are forced to work, security association between the end-points, unless the gateway is on
all media receiver parts of a decomposited end-point will receive the transport level, and additional complexities in form of the
all media, thus doubling the bit-rate consumption for the end- decrypt-encrypt cycles needed for each forwarded packet. SRTP, due
point. to its keying structure, also makes it hard to move a flow from one
RTP session to another as each RTP session will have one or more
different master keys and these must not be the same in multiple RTP
sessions as that can result in two-time pads that completely breaks
the confidentiality of the packets.
RTP Mixers and Translators: An RTP mixer or Media Translator will An additional issue around interworking is that for multi-party
also have to support this particular session setup, where it applications it can be impossible to judge which different RTP
before could rely on the RTP session to determine what processing multiplexing behaviors that will be used by end-points that attempt
options should be applied to the incoming packets. to join a session. Thus if one attempts to use a multiplexing choice
that has poor interworking, one may have to switch at a later stage
when someone wants to participate in a multi-party session using an
RTP application supporting only another behavior. It is likely
difficult to implement the switch without some media disruption.
As can be seen, there is nothing in here that prevents using a single To summarize, certain types of applications are likely to be inter-
RTP session for multiple media types, however it does create a number worked. Sets of applications of similar type should strive to use
of limitations and special case implementation requirements. So the same multiplexing structure to avoid the need to make an RTP
anyone considering to use this setup should carefully review if the session level gateway. This as it incurs complexity costs, can force
reasons for using a single RTP session is sufficient to motivate this the gateway to be part of security associations, force SSRC
special case. translation and even payload translation which is also a potential
hinder to application evolution.
7.3. Signalling Aspects 6.3.2. Multiple SSRC Legacy Considerations
Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per end-point and media type (typically audio
and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each end-point and the
RTP mixer.
When establishing RTP sessions that may contain end-points that
aren't updated to handle multiple streams following these
recommendations, a particular application can have issues with
multiple SSRCs within a single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
RTP Session multiplexing could potentially avoid these issues if
there is only a single SSRC at each end-point, and in topologies
which appears like point to point as seen the end-point. However,
forcing the usage of session multiplexing due to this reason would be
a great mistake, as it is likely that a significant set of
applications will need a combination of SSRC multiplexing of several
media sources and session multiplexing for other aspects such as
encoding alternatives, adding robustness or simply to support legacy.
However, this issue does need consideration when deploying multiple
media streams within an RTP session where legacy end-points may
occur.
6.4. Signalling Aspects
There exist various signalling solutions for establishing RTP There exist various signalling solutions for establishing RTP
sessions. Many are SDP [RFC4566] based, however SDP functionality is sessions. Many are SDP [RFC4566] based, however SDP functionality is
also dependent on the signalling protocols carrying the SDP. Where also dependent on the signalling protocols carrying the SDP. Where
RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative
fashion, SIP [RFC3261] uses SDP with the additional definition of fashion, while SIP [RFC3261] uses SDP with the additional definition
Offer/Answer [RFC3264]. The impact on signalling and especially SDP of Offer/Answer [RFC3264]. The impact on signalling and especially
needs to be considered as it can greatly affect how to deploy a SDP needs to be considered as it can greatly affect how to deploy a
certain multiplexing point choice. certain multiplexing point choice.
7.3.1. Session Oriented Properties 6.4.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused around One aspect of the existing signalling is that it is focused around
sessions, or at least in the case of SDP the media description. sessions, or at least in the case of SDP the media description.
There are a number of things that are signalled on a session level/ There are a number of things that are signalled on a session level/
media description but that are not necessarily strictly bound to an media description but those are not necessarily strictly bound to an
RTP session and could be of interest to signal specifically for a RTP session and could be of interest to signal specifically for a
particular media stream within the session. The following properties particular media stream (SSRC) within the session. The following
have been identified as being potentially useful to signal not only properties have been identified as being potentially useful to signal
on RTP session level: not only on RTP session level:
o Bitrate/Bandwidth exist today only at aggregate or a common any o Bitrate/Bandwidth exist today only at aggregate or a common any
media stream limit media stream limit
o Which SSRC that will use which RTP Payload Types o Which SSRC that will use which RTP Payload Types
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 SSRC multiplexed limitations. However, if the aim is to deploy an SSRC multiplexed
solution that contains several sets of media streams with different solution that contains several sets of media streams with different
properties (encoding/packetization parameter, bit-rate, etc), putting properties (encoding/packetization parameter, bit-rate, etc), putting
each set in a different RTP session would directly enable negotiation each set in a different RTP session would directly enable negotiation
of the parameters for each set. If insisting on SSRC multiplexing, a of the parameters for each set. If insisting on SSRC multiplexing
number of signalling extensions are needed to clarify that there are only, a number of signalling extensions are needed to clarify that
multiple sets of media streams with different properties and that there are multiple sets of media streams with different properties
they shall in fact be kept different, since a single set will not and that they shall in fact be kept different, since a single set
satisfy the applications requirements. will not satisfy the application's requirements.
This does in fact create a strong driver to use RTP session This does in fact create a strong driver to use RTP session
multiplexing for any case where different sets of media streams with multiplexing for any case where different sets of media streams with
different requirements exist. different requirements exist.
7.3.2. SDP Prevents Multiple Media Types 6.4.2. SDP Prevents Multiple Media Types
SDP encoded in its structure a prevention against using multiple SDP encoded in its structure prevention against using multiple media
media types in the same RTP session. A media description in SDP can types in the same RTP session. A media description in SDP can only
only have a single media type; audio, video, text, image, have a single media type; audio, video, text, image, application.
application. This media type is used as the top-level media type for This media type is used as the top-level media type for identifying
identifying the actual payload format bound to a particular payload the actual payload format bound to a particular payload type using
type using the rtpmap attribute. Thus a high fence against using the rtpmap attribute. Thus a high fence against using multiple media
multiple media types in the same session was created. types in the same session was created.
There is a proposal in the MMUSIC WG for how one could allow multiple There is an accepted WG item in the MMUSIC WG to define how multiple
media lines describe a single underlying transport media lines describe a single underlying transport
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus support either [I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus it becomes
one RTP session with multiple media types. There is also a solution possible in SDP to define one RTP session with multiple media types.
for multiplexing multiple RTP sessions onto the same transport
[I-D.westerlund-avtcore-single-transport-multiplexing].
7.4. Network Apsects 6.4.3. Media Stream Usage
Media streams being transported in RTP has some particular usage in
an RTP application. This usage of the media stream is in many
applications so far implicitly signalled. For example by having all
audio media streams arriving in the only audio RTP session they are
to be decoded, mixed and played out. However, in more advanced
applications that use multiple media streams there will be more than
a single usage or purpose among the set of media streams being sent
or received. RTP applications will need to signal this usage
somehow. Here the choice of SSRC multiplexing versus session
multiplexing will have significant impact. If one uses SSRC
multiplexing to its full extent one will have to explicitly indicate
for each SSRC what its' usage and purpose are using some signalling
between the application instances.
This SSRC usage signalling will have some impact on the application
and also on any central RTP nodes. It is important in the design to
consider the implications of the need for additional signalling
between the nodes. One consideration is if a receiver can utilize
the media stream at all before it has received the signalling message
describing the media stream and its usage. Another consideration is
that any RTP central node, like an RTP mixer or translator that
selects, mixes or processes streams, in most cases will need to
receive the same signalling to know how to treat media streams with
different usage in the right fashion.
Application designers should consider putting media streams of the
same usage and/or receiving the same treatment in middleboxes in the
same RTP sessions and use the RTP session as an explicit indication
of how to deal with media streams. By having session level
indication of usage and have different RTP sessions for different
usages, the need for stream specific signalling can be reduced.
Especially signalling of the type that is time critical and needs to
be provided prior to the media stream being available.
6.5. Network Aspects
The multiplexing choice has impact on network level mechanisms that The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor. need to be considered by the implementor.
7.4.1. Quality of Service 6.5.1. Quality of Service
When it comes to Quality of Service mechanisms, they are either flow When it comes to Quality of Service mechanisms, they are either flow
based or marking based. RSVP [RFC2205] is an example of a flow based based or marking based. RSVP [RFC2205] is an example of a flow based
mechanism, while Diff-Serv [RFC2474] is an example of a Marking based mechanism, while Diff-Serv [RFC2474] is an example of a Marking based
one. For a marking based scheme, the method of multiplexing will not one. For a marking based scheme, the method of multiplexing will not
affect the possibility to use QoS. affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between However, for a flow based scheme there is a clear difference between
the methods. SSRC multiplexing will result in all media streams the methods. SSRC multiplexing will result in all media streams
being part of the same 5-tuple (protocol, source address, destination being part of the same 5-tuple (protocol, source address, destination
address, source port, destination port) which is the most common address, source port, destination port) which is the most common
selector for flow based QoS. Thus, separation of the level of QoS selector for flow based QoS. Thus, separation of the level of QoS
between media streams is not possible. That is however possible for between media streams is not possible. That is however possible for
session based multiplexing, where each different version can be in a session based multiplexing, where each different version can be in a
different RTP session that can be sent over different 5-tuples. different RTP session that can be sent over different 5-tuples.
7.4.2. NAT and Firewall Traversal 6.5.2. NAT and Firewall Traversal
In today's network there exist a large number of middleboxes. The In today's network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW). Translators (NAT) and Firewalls (FW).
Below we analyze and comment on the impact of requiring more Below we analyze 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
skipping to change at page 30, line 46 skipping to change at page 32, line 6
NAT Traversal Excess Time: Making the NAT/FW traversal takes a NAT Traversal Excess Time: Making 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 can take following the case scenario for how much extra time it can take following the
specified ICE procedures are: 1.5*RTT + Ta*(Additional_Flows-1), specified ICE procedures are: 1.5*RTT + Ta*(Additional_Flows-1),
where Ta is the pacing timer, which ICE specifies to be no smaller where Ta is the pacing timer, which ICE specifies to be no smaller
than 20 ms. That assumes a message in one direction, and then an than 20 ms. That assumes a message in one direction, and then an
immediate triggered check back. This as ICE first finds one immediate triggered check back. This as ICE first finds one
candidate pair that works prior to establish multiple flows. candidate pair that works prior to establish multiple flows.
Thus, there are no extra time until one has found a working Thus, there is no extra time until one has found a working
candidate pair. Based on that working pair the extra time is to candidate pair. Based on that working pair the needed extra time
in parallel establish the, in most cases 2-3, additional flows. is to in parallel establish the, in most cases 2-3, additional
flows.
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
it should be fairly low as one flow from the same interfaces has it should be fairly low as one flow from the same interfaces has
just been successfully established . Thus only rare events such just been successfully established. Thus only rare events such as
as NAT resource overload, or selecting particular port numbers NAT resource overload, or selecting particular port numbers that
that are filtered etc, should be reasons for failure. are filtered etc, should be reasons for failure.
Deep Packet Inspection and Multiple Streams: Firewalls differ in how
deeply they inspect packets. There exist some potential that
deeply inspecting firewalls will have similar legacy issues with
multiple SSRCs as some stack implementations.
SSRC multiplexing keeps additional media streams within one RTP SSRC multiplexing keeps additional media streams within one RTP
Session and does not introduce any additional NAT traversal Session and does not introduce any additional NAT traversal
complexities per media stream. In contrast, the session multiplexing complexities per media stream. In contrast, the session multiplexing
is using one RTP session per media stream. Thus additional lower is using one RTP session per media stream. Thus additional lower
layer transport flows will be required, unless an explicit de- layer transport flows will be required, unless an explicit de-
multiplexing layer is added between RTP and the transport protocol. multiplexing layer is added between RTP and the transport protocol.
A proposal for how to multiplex multiple RTP sessions over the same A proposal for how to multiplex multiple RTP sessions over the same
single lower layer transport exist in single lower layer transport exist in
[I-D.westerlund-avtcore-single-transport-multiplexing]. [I-D.westerlund-avtcore-single-transport-multiplexing].
7.4.3. Multicast 6.5.3. Multicast
Multicast groups provides a powerful semantics for a number of real- Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like time applications, especially the ones that desire broadcast-like
behaviors with one end-point transmitting to a large number of behaviors with one end-point transmitting to a large number of
receivers, like in IPTV. But that same semantics do result in a receivers, like in IPTV. But that same semantics do result in a
certain number of limitations. certain number of limitations.
One limitation is that for any group, sender side adaptation to the One limitation is that for any group, sender side adaptation to the
actual receiver properties causes a degradation for all participants actual receiver properties causes degradation for all participants to
to what is supported by the receiver with the worst conditions among what is supported by the receiver with the worst conditions among the
the group participants. In most cases this is not acceptable. group participants. In most cases this is not acceptable. Instead
Instead various receiver based solutions are employed to ensure that various receiver based solutions are employed to ensure that the
the receivers achieve best possible performance. By using scalable receivers achieve best possible performance. By using scalable
encoding and placing each scalability layer in a different multicast encoding and placing each scalability layer in a different multicast
group, the receiver can control the amount of traffic it receives. group, the receiver can control the amount of traffic it receives.
To have each scalability layer on a different multicast group, one To have each scalability layer on a different multicast group, one
RTP session per multicast group is used. RTP session per multicast group is used.
If instead a single RTP session over multiple transports were to be If instead a single RTP session over multiple transports were to be
deployed, i.e. multicast groups with each layer as it's own SSRC, deployed, i.e. multicast groups with each layer as it's own SSRC,
then very different views of the RTP session would exist. That as then very different views of the RTP session would exist. That as
one receiver may see only a single layer (SSRC), while another may one receiver may see only a single layer (SSRC), while another may
see three SSRCs if it joined three multicast groups. This would see three SSRCs if it joined three multicast groups. This would
skipping to change at page 32, line 11 skipping to change at page 33, line 22
able to determine if a receiver isn't reporting on a particular SSRC able to determine if a receiver isn't reporting on a particular SSRC
due to that it is not a member of that multicast group, or because it due to that it is not a member of that multicast group, or because it
doesn't receive it as a result of a transport failure. doesn't receive it as a result of a transport failure.
Thus it appears easiest and most straightforward to use multiple RTP Thus it appears easiest and most straightforward to use multiple RTP
sessions. In addition, the transport flow considerations in sessions. In addition, the transport flow considerations in
multicast are a bit different from unicast. First of all there is no multicast are a bit different from unicast. First of all there is no
shortage of port space, as each multicast group has its own port shortage of port space, as each multicast group has its own port
space. space.
7.4.4. Multiplexing multiple RTP Session on a Single Transport 6.5.4. Multiplexing multiple RTP Session on a Single Transport
For applications that doesn't need flow based QoS and like to save For applications that doesn't need flow based QoS and like to save
ports and NAT/FW traversal costs, there is a proposal for how to ports and NAT/FW traversal costs and where usage of multiple media
achieve multiplexing of multiple RTP sessions over the same lower types in one RTP session is not suitable, there is a proposal for how
to achieve multiplexing of multiple RTP sessions over the same lower
layer transport layer transport
[I-D.westerlund-avtcore-single-transport-multiplexing]. Using such a [I-D.westerlund-avtcore-single-transport-multiplexing]. Using such a
solution would allow session multiplexing without most of the solution would allow session multiplexing without most of the
perceived downsides of additional RTP sessions creating a need for perceived downsides of additional RTP sessions creating a need for
additional transport flows. additional transport flows.
7.5. Security Aspects 6.6. Security Aspects
On the basic level there is no significant difference in security On the basic level there is no significant difference in security
when having one RTP session and having multiple. However, there are when having one RTP session and having multiple. However, there are
a few more detailed considerations that might need to be considered a few more detailed considerations that might need to be considered
in certain usages. in certain usages.
7.5.1. Security Context Scope 6.6.1. Security Context Scope
When using SRTP [RFC3711] the security context scope is important and When using SRTP [RFC3711] the security context scope is important and
can be a necessary differentiation in some applications. As SRTP's can be a necessary differentiation in some applications. As SRTP's
crypto suites (so far) is built around symmetric keys, the receiver crypto suites (so far) is built around symmetric keys, the receiver
will need to have the same key as the sender. This results in that will need to have the same key as the sender. This results in that
none in a multi-party session can be certain that a received packet no one in a multi-party session can be certain that a received packet
really was sent by the claimed sender or by another party having really was sent by the claimed sender or by another party having
access to the key. In most cases this is a sufficient security access to the key. In most cases this is a sufficient security
property, but there are a few cases where this does create property, but there are a few cases where this does create
situations. situations.
The first case is when someone leaves a multi-party session and one The first case is when someone leaves a multi-party session and one
wants to ensure that the party that left can no longer access the wants to ensure that the party that left can no longer access the
media streams. This requires that everyone re-keys without media streams. This requires that everyone re-keys without
disclosing the keys to the excluded party. disclosing the keys to the excluded party.
skipping to change at page 33, line 11 skipping to change at page 34, line 24
stream can be based on the stream being encrypted with a key that stream can be based on the stream being encrypted with a key that
user can't access without paying premium, having the key-management user can't access without paying premium, having the key-management
limit access to the key. limit access to the key.
In the latter case it is likely easiest from signalling, transport In the latter case it is likely easiest from signalling, transport
(if done over multicast) and security to use a different RTP session. (if done over multicast) and security to use a different RTP session.
That way the user(s) not intended to receive a particular stream can That way the user(s) not intended to receive a particular stream can
easily be excluded. There is no need to have SSRC specific keys, easily be excluded. There is no need to have SSRC specific keys,
which many of the key-management systems cannot handle. which many of the key-management systems cannot handle.
7.5.2. Key-Management for Multi-party session 6.6.2. Key-Management for Multi-party session
Performing key-management for Multi-party session can be a challenge. Performing key-management for Multi-party session can be a challenge.
This section considers some of the issues. This section considers some of the issues.
Transport translator based session cannot use Security Description Transport translator based session cannot use Security Description
[RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end- [RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end-
point provides it's set of keys. In centralized conference, the point provides its set of keys. In centralized conference, the
signalling counterpart is a conference server and the media plane signalling counterpart is a conference server and the media plane
unicast counterpart (to which DTLS messages would be sent) is the unicast counterpart (to which DTLS messages would be sent) is the
translator. Thus an extension like Encrypted Key Transport translator. Thus an extension like Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] are needed or a MIKEY [RFC3830] based [I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution
solution that allows for keying all session participants with the that allows for keying all session participants with the same master
same master key. key.
Keying of multicast transported SRTP face similar challenges as the Keying of multicast transported SRTP face similar challenges as the
transport translator case. transport translator case.
6.6.3. Complexity Implications
The usage of security functions can surface complexity implications
of the choice of multiplexing and topology. This becomes especially
evident in RTP topologies having any type of middlebox that processes
or modifies RTP/RTCP packets. Where there is very small overhead for
a not secured RTP translator or mixer to rewrite an SSRC value in the
RTP packet, the cost of doing it when using cryptographic security
functions is higher. For example if using SRTP [RFC3711], the actual
security context and exact crypto key are determined by the SSRC
field value. If one changes it, the encryption and authentication
tag must be performed using another key. Thus changing the SSRC
value implies a decryption using the old SSRC and its security
context followed by an encryption using the new one.
There exist many valid cases where a middlebox will be forced to
perform such cryptographic operations due to the intended purpose of
the middlebox, for example a media transcoding RTP translator cannot
avoid performing these operations as they will produce a different
payload compared to the input. However, there exist some cases where
another topology and/or multiplexing choice could avoid the
complexities.
6.7. Multiple Media Types in one RTP session
Having different media types, like audio and video, in the same RTP
sessions is not forbidden, only recommended against as earlier
discussed in Section 6.2.1.1. When using multiple media types, there
are a number of considerations:
Payload Type gives Media Type: This solution is dependent on getting
the media type from the Payload Type. Thus overloading this de-
multiplexing point in a receiver making it serve two purposes.
First to provide the main media type and determining the
processing chain, then later for the exact configuration of the
encoder and packetization.
Payload Type field limitations: The total number of Payload Types
available to use in an RTP session is fairly limited, especially
if Multiplexing RTP Data and Control Packets on a Single Port
[RFC5761] is used. For certain applications negotiating a large
set of codes and configuration this may become an issue.
An SSRC cannot use two clock rates simultaneously: The used RTP
clock rate for an SSRC is determined from the payload type. As
discussed in Appendix A it is not possible to simultaneously use
two different clock rates for the same SSRC. Even switching clock
rate once has potential issues if packet loss occurs at the same
time. Different media types commonly have different clock rates
preventing or creating issues to use two different media types for
the same SSRC.
Do not switch media types for an SSRC: The primary reasons to avoid
switching from sending for example audio to sending video using
the same SSRC is the implications on a receiver. When this
happens, the processing chain in the receiver will have to switch
from one media type to another. As the different media type's
entire processing chains are different and are connected to
different outputs it is difficult to reuse the decoding chain,
which a normal codec change likely can. Instead the entire
processing chain has to be torn down and replaced. In addition,
there is likely a clock rate switching problem, possibly resulting
in synchronization loss at the point of switching media type if
some packet loss occurs. So this is a behavior that shall be
avoided.
RTCP Bit-rate Issues: If the media types are significantly different
in bit-rate, the RTCP bandwidth rates assigned to each source in a
session can result in interesting effects, like that the RTCP bit-
rate share for an audio stream is larger than the actual audio
bit-rate. In itself this doesn't cause any conflicts, only
potentially unnecessary overhead. It is possible to avoid this
using AVPF or SAVPF and setting trr-int parameter, which can bring
down unnecessary regular reporting while still allowing for rapid
feedback.
De-composite end-points: De-composite nodes that rely on the regular
network to separate audio and video to different devices do not
work well with this session setup. If they are forced to work,
all media receiver parts of a de-composite end-point will receive
all media, thus doubling the bit-rate consumption for the end-
point.
Flow based QoS Separation: Flow based QoS mechanisms will see all
the media streams in the RTP session as part of a single flow.
Therefore there is no possibility to provide separated QoS
behavior for the different media types or flows.
RTP Mixers and Translators: An RTP mixer or Media Translator will
also have to support this particular session setup, where it
before could rely on the RTP session to determine what processing
options should be applied to the incoming packets.
Legacy Implementations: The use of multiple media types has the
potential for even larger issues with legacy implementations than
single media type SSRC multiplexing due to the occurrence of
multiple media types among the payload type configurations.
As can be seen, there is nothing in here that prevents using a single
RTP session for multiple media types, however it does create a number
of limitations and special case implementation requirements. So
anyone considering using this setup should carefully review if the
reasons for using a single RTP session are sufficient to motivate the
needed special handling.
7. Arch-Types
This section discusses some arch-types of how RTP multiplexing can be
used in applications to achieve certain goals and a summary of their
implications. For each arch-type there is discussion of benefits and
downsides.
7.1. Single SSRC per Session
In this arch-type each end-point in a point-to-point session has only
a single SSRC, thus the RTP session contains only two SSRCs, one
local and one remote. This session can be used both unidirectional,
i.e. only a single media stream or bi-directional, i.e. both end-
points have one media stream each. If the application needs
additional media flows between the end-points, they will have to
establish additional RTP sessions.
The Pros:
1. This arch-type has great legacy interoperability potential as it
will not tax any RTP stack implementations.
2. The signalling has good possibilities to negotiate and describe
the exact formats and bit-rates for each media stream, especially
using today's tools in SDP.
3. It does not matter if usage or purpose of the media stream is
signalled on media stream level or session level as there is no
difference.
4. It is possible to control security association per RTP session
with current key-management.
The Cons:
a. The number of required RTP sessions cannot really be higher,
which has the implications:
* Linear growth of the amount of NAT/FW state with number of
media streams.
* Increased delay and resource consumption from NAT/FW
traversal.
* Likely larger signalling message and signalling processing
requirement due to the amount of session related information.
* Higher potential for a single media stream to fail during
transport between the end-points.
b. When the number of RTP sessions grows, the amount of explicit
state for relating media stream also grows, linearly or possibly
exponentially, depending on how the application needs to relate
media streams.
c. The port consumption may become a problem for centralized
services, where the central node's port consumption grows rapidly
with the number of sessions.
d. For applications where the media streams are highly dynamic in
their usage, i.e. entering and leaving, the amount of signalling
can grow high. Issues arising from the timely establishment of
additional RTP sessions can also arise.
e. Cross session RTCP requests needs is likely to exist and may
cause issues.
f. If the same SSRC value is reused in multiple RTP sessions rather
than being randomly chosen, interworking with applications that
uses another multiplexing structure than this application will
have issues and require SSRC translation.
g. Cannot be used with Any Source Multicast (ASM) as one cannot
guarantee that only two end-points participate as packet senders.
Using SSM, it is possible to restrict to these requirements if no
RTCP feedback is used.
h. For most security mechanisms, each RTP session or transport flow
requires individual key-management and security association
establishment thus increasing the overhead.
i. Does not support multiparty session within a session. Instead
each multi-party participant will require an individual RTP
session to a given end-point, even if a central node is used.
RTP applications that need to inter-work with legacy RTP
applications, like VoIP and video conferencing, can potentially
benefit from this structure. However, a large number of media
descriptions in SDP can also run into issues with existing
implementations. For any application needing a larger number of
media flows, the overhead can become very significant. This
structure is also not suitable for multi-party sessions, as any given
media stream from each participant, although having same usage in the
application, must have its own RTP session. In addition, the dynamic
behavior that can arise in multi-party applications can tax the
signalling system and make timely media establishment more difficult.
7.2. Multiple SSRCs of the Same Media Type
In this arch-type, each RTP session serves only a single media type.
The RTP session can contain multiple media streams, either from a
single end-point or due to multiple end-points. This commonly
creates a low number of RTP sessions, typically only two one for
audio and one for video with a corresponding need for two listening
ports when using RTP and RTCP multiplexing.
The Pros:
1. Low number of RTP sessions needed compared to single SSRC case.
This implies:
* Reduced NAT/FW state
* Lower NAT/FW Traversal Cost in both processing and delay.
2. Allows for early de-multiplexing in the processing chain in RTP
applications where all media streams of the same type have the
same usage in the application.
3. Works well with media type de-composite end-points.
4. Enables Flow-based QoS with different prioritization between
media types.
5. For applications with dynamic usage of media streams, i.e. they
come and go frequently, having much of the state associated with
the RTP session rather than an individual SSRC can avoid the need
for in-session signalling of meta-information about each SSRC.
6. Low overhead for security association establishment.
The Cons:
a. May have some need for cross session RTCP requests for things
that affect both media types in an asynchronous way.
b. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per end-point.
c. Will not be able to control security association for sets of
media streams within the same media type with today's key-
management mechanisms, only between SDP media descriptions.
For RTP applications where all media streams of the same media type
share same usage, this structure provides efficiency gains in amount
of network state used and provides more faith sharing with other
media flows of the same type. At the same time, it is still
maintaining almost all functionalities when it comes to negotiation
in the signalling of the properties for the individual media type and
also enabling flow based QoS prioritization between media types. It
handles multi-party session well, independently of multicast or
centralized transport distribution, as additional sources can
dynamically enter and leave the session.
7.3. Multiple Sessions for one Media type
In this arch-type one goes one step further than in the above
(Section 7.2) by using multiple RTP sessions also for a single media
type. The main reason for going in this direction is that the RTP
application needs separation of the media streams due to their usage.
Some typical reasons for going to this arch-type are scalability over
multicast, simulcast, need for extended QoS prioritization of media
streams due to their usage in the application, or the need for fine
granular signalling using today's tools.
The Pros:
1. More suitable for Multicast usage where receivers can
individually select which RTP sessions they want to participate
in, assuming each RTP session has its own multicast group.
2. Detailed indication of the application's usage of the media
stream, where multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely signalling.
4. Enables detailed QoS prioritization for flow based mechanisms.
5. Works well with de-composite end-points.
6. Handles dynamic usage of media streams well.
7. For transport translator based multi-party sessions, this
structure allows for improved control of which type of media
streams an end-point receives.
8. The scope for who is included in a security association can be
structured around the different RTP sessions, thus enabling such
functionality with existing key-management.
The Cons:
a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.
b. Increased amount of session configuration state.
c. May need synchronized cross-session RTCP requests and require
some consideration due to this.
d. For media streams that are part of scalability, simulcast or
transport robustness it will be needed to bind sources, which
must support multiple RTP sessions.
e. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per end-point.
f. Higher overhead for security association establishment.
g. If the applications need finer control than on media type level
over which session participants that are included in different
sets of security associations, most of today's key-management
will have difficulties establishing such a session.
For more complex RTP applications that have several different usages
for media streams of the same media type and / or uses scalability or
simulcast, this solution can enable those functions at the cost of
increased overhead associated with the additional sessions. This
type of structure is suitable for more advanced applications as well
as multicast based applications requiring differentiation to
different participants.
7.4. Multiple Media Types in one Session
This arch-type is to use a single RTP session for multiple different
media types, like audio and video, and possibly also transport
robustness mechanisms like FEC or Retransmission. Each media stream
will use its own SSRC and a given SSRC value from a particular end-
point will never use the SSRC for more than a single media type.
The Pros:
1. Single RTP session which implies:
* Minimal NAT/FW state.
* Minimal NAT/FW Traversal Cost.
* Fate-sharing for all media flows.
2. Enables separation of the different media types based on the
payload types so media type specific end-point or central
processing can still be supported despite single session.
3. Can handle dynamic allocations of media streams well on an RTP
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
4. Minimal overhead for security association establishment.
The Cons:
a. Not suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a
single media type, due to high risk of forced SSRC translation.
b. Negotiation of bandwidth for the different media types is
currently not possible in SDP. This requires SDP extensions to
enable payload or source specific bandwidth. Likely to be a
problem due to media type asymmetry in required bandwidth.
c. Does enforce higher bandwidth and processing on de-composite end-
points.
d. Flow based QoS cannot provide separate treatment to some media
streams compared to other in the single RTP session.
e. If there is significant asymmetry between the media streams RTCP
reporting needs, there are some challenges in configuration and
usage to avoid wasting RTCP reporting on the media stream that
does not need that frequent reporting.
f. Not suitable for applications where some receivers like to
receive only a subset of the media streams, especially if
multicast or transport translator is being used.
g. Additional concern with legacy implementations that does not
support the RTP specification fully when it comes to handling
multiple SSRC per end-point, as also multiple simultaneous media
types needs to be handled.
h. If the applications need finer control over which session
participants that are included in different sets of security
associations, most key-management will have difficulties
establishing such a session.
The analysis in this document and considerations in Section 6.7
implies that this is suitable only in a set of restricted use cases.
The aspect in the above list that can be most difficult to judge long
term is likely the potential need for interworking with other
applications and services.
7.5. Summary
There are some clear relations between these arch-types. Both the
"single SSRC per RTP session" and the "multiple media types in one
session" are cases which require full explicit signalling of the
media stream relations. However, they operate on two different
levels where the first primarily enables session level binding, and
the second needs to do it all on SSRC level. From another
perspective, the two solutions are the two extreme points when it
comes to number of RTP sessions required.
The two other arch-types "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for one Media Type" are examples of two other
cases that first of all allows for some implicit mapping of the role
or usage of the media streams based on which RTP session they appear
in. It thus potentially allows for less signalling and in particular
reduced need for real-time signalling in dynamic sessions. They also
represent points in between the first two when it comes to amount of
RTP sessions established, i.e. representing an attempt to reduce the
amount of sessions as much as possible without compromising the
functionality the session provides both on network level and on
signalling level.
8. Guidelines 8. Guidelines
This section contains a number of recommendations for implementors or This section contains a number of recommendations for implementors or
specification writers when it comes to handling multi-stream. specification writers when it comes to handling multi-stream.
Don't Require the same SSRC across Sessions: As discussed in Do not Require the same SSRC across Sessions: As discussed in
Section 7.2.6 there exist drawbacks in using the same SSRC in Section 6.2.4 there exist drawbacks in using the same SSRC in
multiple RTP sessions as a mechanism to bind related media streams multiple RTP sessions as a mechanism to bind related media streams
together. Instead a mechanism to explicitly signal the relation together. It is instead recommended that a mechanism to
SHOULD be used, either in RTP/RTCP or in the used signalling explicitly signal the relation is used, either in RTP/RTCP or in
mechanism that establish the RTP session(s). the used signalling mechanism that establishes the RTP session(s).
Use SSRC multiplexing for additional Media Sources: In the cases an Use SSRC multiplexing for additional Media Sources: In the cases an
RTP end-point needs to transmit additional media source(s) of the RTP end-point needs to transmit additional media source(s) of the
same media type and purpose in the application it is RECOMMENDED same media type and purpose in the application, it is recommended
to send them as additional SSRCs in the same RTP session. For to send them as additional SSRCs in the same RTP session. For
example a telepresence room where there are three cameras, and example a tele-presence room where there are three cameras, and
each camera captures 2 persons sitting at the table, sending each each camera captures 2 persons sitting at the table, sending each
camera as its own SSRC within a single RTP session is recommended. camera as its own SSRC within a single RTP session is recommended.
Use additional RTP sessions for streams with different purposes: Use additional RTP sessions for streams with different purposes:
When media streams have different purpose or processing When media streams have different purpose or processing
requirements it is RECOMMENDED that the different types of streams requirements it is recommended that the different types of streams
are put in different RTP sessions. are put in different RTP sessions.
When using Session Multiplexing use grouping: When using Session When using Session Multiplexing use grouping: When using Session
Multiplexing solutions it is RECOMMENDED to be explicitly group Multiplexing solutions, it is recommended to be explicitly group
the involved RTP sessions using the signalling mechanism, for the involved RTP sessions using the signalling mechanism, for
example The Session Description Protocol (SDP) Grouping Framework. example The Session Description Protocol (SDP) Grouping Framework.
[RFC5888] [RFC5888], using some appropriate grouping semantics.
RTP/RTCP Extensions May Support SSRC and Session Multiplexing: When RTP/RTCP Extensions May Support SSRC and Session Multiplexing: When
defining an RTP or RTCP extension, the creator needs to consider defining an RTP or RTCP extension, the creator needs to consider
if this extension is applicable in both SSRC multiplexed and if this extension is applicable in both SSRC multiplexed and
Session multiplexed usages. If it is, then any generic extensions Session multiplexed usages. Any extension intended to be generic
are RECOMMENDED to support both. Applications that are not as is recommended to support both. Applications that are not as
generally applicable will have to consider if interoperability is generally applicable will have to consider if interoperability is
better served by defining a single solution or providing both better served by defining a single solution or providing both
options. options.
Transport Support Extensions: When defining new RTP/RTCP extensions Transport Support Extensions: When defining new RTP/RTCP extensions
intended for transport support, like the retransmission or FEC intended for transport support, like the retransmission or FEC
mechanisms, they are RECOMMENDED to include support for both SSRC mechanisms, they are recommended to include support for both SSRC
and Session multiplexing so that application developers can choose and Session multiplexing so that application developers can choose
freely from the set of mechanisms without concerning themselves freely from the set of mechanisms without concerning themselves
with if a particular solution only supports one of the with which of the multiplexing choices a particular solution
multiplexing choices. supports.
This discussion and guidelines points out that a small set of 9. Proposal for Future Work
The above discussion and guidelines indicates that a small set of
extension mechanisms could greatly improve the situation when it extension mechanisms could greatly improve the situation when it
comes to using multiple streams independently of Session multiplexing comes to using multiple streams independently of Session multiplexing
or SSRC multiplexing. These extensions are: or SSRC multiplexing. These extensions are:
Media Source Identification: A Media source identification that can Media Source Identification: A Media source identification that can
be used to bind together media streams that are related to the be used to bind together media streams that are related to the
same media source. A proposal same media source. A proposal
[I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES [I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
item SRCNAME that also can be used with the a=ssrc SDP attribute item SRCNAME that also can be used with the a=ssrc SDP attribute
to provide signalling layer binding information. to provide signalling layer binding information.
SSRC limiations within RTP sessions: By providing a signalling SSRC limitations within RTP sessions: By providing a signalling
solution that allows the signalling peers to explicitly express solution that allows the signalling peers to explicitly express
both support and limitations on how many simultaneous media both support and limitations on how many simultaneous media
streams an end-point can handle within a given RTP Session. That streams an end-point can handle within a given RTP Session. That
ensures that usage of SSRC multiplexing occurs when supported and ensures that usage of SSRC multiplexing occurs when supported and
without overloading an end-point. This extension is proposed in without overloading an end-point. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc]. [I-D.westerlund-avtcore-max-ssrc].
9. RTP Specification Clarifications 10. RTP Specification Clarifications
This section describes a number of clarifications to the RTP This section describes a number of clarifications to the RTP
specifications that are likely necessary for aligned behavior when specifications that are likely necessary for aligned behavior when
RTP sessions contains more SSRCs than one local and one remote. RTP sessions contain more SSRCs than one local and one remote.
9.1. RTCP Reporting from all SSRCs 10.1. RTCP Reporting from all SSRCs
When one have multiple SSRC in an RTP node, then all these SSRC must When one have multiple SSRC in an RTP node, all these SSRC must send
send RTCP SR or RR as long as the SSRC exist. It is not sufficient RTCP SR or RR as long as the SSRC exist. It is not sufficient that
that only one SSRC in the node sends report blocks on the incoming only one SSRC in the node sends report blocks on the incoming RTP
RTP streams. The reason for this is that a third party monitor may streams. The reason for this is that a third party monitor may not
not necessarily be able to determine that all these SSRC are in fact necessarily be able to determine that all these SSRC are in fact co-
co-located and originate from the same stack instance that gather located and originate from the same stack instance that gather report
report data. data.
9.2. RTCP Self-reporting 10.2. RTCP Self-reporting
For any RTP node that sends more than one SSRC, there exist the For any RTP node that sends more than one SSRC, there is the question
question if SSRC1 needs to report its reception of SSRC2 and vice if SSRC1 needs to report its reception of SSRC2 and vice versa. The
versa. The reason that they in fact need to report on all other reason that they in fact need to report on all other local streams as
local streams as being received is report consistency. A third party being received is report consistency. A third party monitor that
monitor that considers the full matrix of media streams and all known considers the full matrix of media streams and all known SSRC reports
SSRC reports on these media streams would detect a gap in the reports on these media streams would detect a gap in the reports which could
which could be a transport issue unless identified as in fact being be a transport issue unless identified as in fact being sources from
sources from same node. same node.
9.3. Combined RTCP Packets 10.3. Combined RTCP Packets
When a node contains multiple SSRCs, it is questionable if an RTCP When a node contains multiple SSRCs, it is questionable if an RTCP
compound packet can only contain RTCP packets from a single SSRC or compound packet can only contain RTCP packets from a single SSRC or
if multiple SSRCs can include their packets in a joint compound if multiple SSRCs can include their packets in a joint compound
packet. The high level question is a matter for any receiver packet. The high level question is a matter for any receiver
processing on what to expect. In addition to that question there is processing on what to expect. In addition to that question there is
the issue of how to use the RTCP timer rules in these cases, as the the issue of how to use the RTCP timer rules in these cases, as the
existing rules are focused on determining when a single SSRC can existing rules are focused on determining when a single SSRC can
send. send.
10. IANA Considerations 11. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
11. Security Considerations 12. Security Considerations
12. Acknowledgements There is discussion of the security implications of choosing SSRC vs
Session multiplexing in Section 6.6.
13. References 13. Acknowledgements
13.1. Normative References The authors would like to thanks Harald Alvestrand for providing
input into the discussion regarding multiple media types in a single
RTP session.
14. References
14.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
13.2. Informative References 14.2. 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), Computer Communications Review, Vol. 20(4),
September 1990. September 1990.
[I-D.alvestrand-rtp-sess-neutral]
Alvestrand, H., "Why RTP Sessions Should Be Content
Neutral", draft-alvestrand-rtp-sess-neutral-00 (work in
progress), December 2011.
[I-D.holmberg-mmusic-sdp-bundle-negotiation] [I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers", Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011. progress), October 2011.
[I-D.ietf-avt-srtp-ekt] [I-D.ietf-avt-srtp-ekt]
McGrew, D., Andreasen, F., Wing, D., and K. Fischer, Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
"Encrypted Key Transport for Secure RTP", Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
draft-ietf-avt-srtp-ekt-02 (work in progress), March 2011. (work in progress), October 2011.
[I-D.ietf-avtext-multiple-clock-rates] [I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M., "Support for multiple clock rates in Petit-Huguenin, M., "Support for multiple clock rates in
an RTP session", draft-ietf-avtext-multiple-clock-rates-01 an RTP session", draft-ietf-avtext-multiple-clock-rates-02
(work in progress), July 2011. (work in progress), January 2012.
[I-D.ietf-payload-rtp-howto] [I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format", Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-01 (work in progress), draft-ietf-payload-rtp-howto-01 (work in progress),
July 2011. July 2011.
[I-D.westerlund-avtcore-max-ssrc] [I-D.westerlund-avtcore-max-ssrc]
Westerlund, M., Burman, B., and F. Jansson, "Multiple Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling", Synchronization sources (SSRC) in RTP Session Signaling",
draft-westerlund-avtcore-max-ssrc (work in progress), draft-westerlund-avtcore-max-ssrc (work in progress),
skipping to change at page 39, line 23 skipping to change at page 49, line 45
Protocol (SDP) Grouping Framework", RFC 5888, June 2010. Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190, "RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011. May 2011.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax, [RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP "Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011. Sessions", RFC 6285, June 2011.
Appendix A. Dismissing Payload Type Multiplexing
This section documents a number of reasons why using the payload type
as a multiplexing point for most things related to multiple streams
is unsuitable. If one attempts to use Payload type multiplexing
beyond it's defined usage, that has well known negative effects on
RTP. To use Payload type as the single discriminator for multiple
streams implies that all the different media streams are being sent
with the same SSRC, thus using the same timestamp and sequence number
space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not
possible, Payload Type multiplexing cannot be used.
2. Many RTP payload formats may fragment a media object over
multiple packets, like parts of a video frame. These payload
formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all
fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in
sequence.
3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/
T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before
being used with Payload Type multiplexing.
4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which
stream a packet loss relates to.
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
sequence number, not by Payload Type. If only some of the
Payload Type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets must be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported,
so sending feedback for a specific media stream is difficult
without extending existing RTCP reporting.
7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing.
8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number
of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this
is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing.
11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.
12. A legacy end-point that doesn't understand the indication that
different RTP payload types are different media streams may be
slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types.
Authors' Addresses Authors' Addresses
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Farogatan 6
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
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