Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Informational Ericsson
Expires: January 17, 2013 C. Perkins
University of Glasgow
H. Alvestrand
Google
July 16, 2012
Guidelines for using the Multiplexing Features of RTP
draft-westerlund-avtcore-multiplex-architecture-02
Abstract
Real-time Transport Protocol (RTP) is a flexible protocol possible to
use in a wide range of applications and network and system
topologies. This flexibility and the implications of different
choices should be understood by any application developer using RTP.
To facilitate that understanding, this document contains an in-depth
discussion of the usage of RTP's multiplexing points; the RTP session
and the Synchronisation Source Identifier (SSRC). The document tries
to give guidance and source material for an analysis on the most
suitable choices for the application being designed.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 17, 2013.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
2.2. Subjects Out of Scope . . . . . . . . . . . . . . . . . . 7
3. RTP Concepts . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.1. Session . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.2. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
3.3. CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.4. Payload Type . . . . . . . . . . . . . . . . . . . . . . . 10
4. Multiple Streams Alternatives . . . . . . . . . . . . . . . . 12
5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 13
5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 13
5.1.1. Translators & Gateways . . . . . . . . . . . . . . . . 14
5.2. Point to Multipoint Using Multicast . . . . . . . . . . . 15
5.3. Point to Multipoint Using an RTP Transport Translator . . 17
5.4. Point to Multipoint Using an RTP Mixer . . . . . . . . . . 18
5.4.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 19
5.4.2. Media Switching . . . . . . . . . . . . . . . . . . . 22
5.4.3. RTP Source Projecting . . . . . . . . . . . . . . . . 24
5.5. Point to Multipoint using Multiple Unicast flows . . . . . 26
5.6. De-composite Endpoint . . . . . . . . . . . . . . . . . . 27
6. Multiple Streams Discussion . . . . . . . . . . . . . . . . . 28
6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 28
6.2. RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 28
6.2.1. The RTP Specification . . . . . . . . . . . . . . . . 29
6.2.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 31
6.2.3. Handling Varying sets of Senders . . . . . . . . . . . 32
6.2.4. Cross Session RTCP Requests . . . . . . . . . . . . . 32
6.2.5. Binding Related Sources . . . . . . . . . . . . . . . 33
6.2.6. Forward Error Correction . . . . . . . . . . . . . . . 35
6.2.7. Transport Translator Sessions . . . . . . . . . . . . 36
6.3. Interworking . . . . . . . . . . . . . . . . . . . . . . . 36
6.3.1. Types of Interworking . . . . . . . . . . . . . . . . 36
6.3.2. RTP Translator Interworking . . . . . . . . . . . . . 36
6.3.3. Gateway Interworking . . . . . . . . . . . . . . . . . 37
6.3.4. Multiple SSRC Legacy Considerations . . . . . . . . . 38
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6.4. Network Aspects . . . . . . . . . . . . . . . . . . . . . 38
6.4.1. Quality of Service . . . . . . . . . . . . . . . . . . 39
6.4.2. NAT and Firewall Traversal . . . . . . . . . . . . . . 39
6.4.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 41
6.4.4. Multiplexing multiple RTP Session on a Single
Transport . . . . . . . . . . . . . . . . . . . . . . 41
6.5. Security Aspects . . . . . . . . . . . . . . . . . . . . . 42
6.5.1. Security Context Scope . . . . . . . . . . . . . . . . 42
6.5.2. Key Management for Multi-party session . . . . . . . . 42
6.5.3. Complexity Implications . . . . . . . . . . . . . . . 43
7. Arch-Types . . . . . . . . . . . . . . . . . . . . . . . . . . 43
7.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 43
7.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 45
7.3. Multiple Sessions for one Media type . . . . . . . . . . . 46
7.4. Multiple Media Types in one Session . . . . . . . . . . . 48
7.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 49
8. Summary considerations and guidelines . . . . . . . . . . . . 50
8.1. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . 50
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 51
10. Security Considerations . . . . . . . . . . . . . . . . . . . 51
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 51
11.1. Normative References . . . . . . . . . . . . . . . . . . . 51
11.2. Informative References . . . . . . . . . . . . . . . . . . 51
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 55
Appendix B. Proposals for Future Work . . . . . . . . . . . . . . 57
Appendix C. RTP Specification Clarifications . . . . . . . . . . 57
C.1. RTCP Reporting from all SSRCs . . . . . . . . . . . . . . 58
C.2. RTCP Self-reporting . . . . . . . . . . . . . . . . . . . 58
C.3. Combined RTCP Packets . . . . . . . . . . . . . . . . . . 58
Appendix D. Signalling considerations . . . . . . . . . . . . . . 58
D.1. Signalling Aspects . . . . . . . . . . . . . . . . . . . . 59
D.1.1. Session Oriented Properties . . . . . . . . . . . . . 59
D.1.2. SDP Prevents Multiple Media Types . . . . . . . . . . 60
D.1.3. Signalling Media Stream Usage . . . . . . . . . . . . 60
Appendix E. Changes from -01 to -02 . . . . . . . . . . . . . . . 61
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61
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1. Introduction
Real-time Transport Protocol (RTP) [RFC3550] is a commonly used
protocol for real-time media transport. It is a protocol that
provides great flexibility and can support a large set of different
applications. RTP has several multiplexing points designed for
different purposes. These enable support of multiple media streams
and switching between different encoding or packetization of the
media. By using multiple RTP sessions, sets of media streams can be
structured for efficient processing or identification. Thus the
question for any RTP application designer is how to best use the RTP
session, the SSRC and the payload type to meet the application's
needs.
The purpose of this document is to provide clear information about
the possibilities of RTP when it comes to multiplexing. The RTP
application designer should understand the implications that come
from a particular usage of the RTP multiplexing points. The document
will recommend against some usages as being unsuitable, in general or
for particular purposes.
RTP was from the beginning designed for multiple participants in a
communication session. This is not restricted to multicast, as some
may believe, but also provides functionality over unicast, using
either multiple transport flows below RTP or a network node that re-
distributes the RTP packets. The re-distributing node can for
example be a transport translator (relay) that forwards the packets
unchanged, a translator performing media or protocol translation in
addition to forwarding, or an RTP mixer that creates new conceptual
sources from the received streams. In addition, multiple streams may
occur when a single endpoint have multiple media sources, like
multiple cameras or microphones that need to be sent simultaneously.
This document has been written due to increased interest in more
advanced usage of RTP, resulting in questions regarding the most
appropriate RTP usage. The limitations in some implementations, RTP/
RTCP extensions, and signalling has also been exposed. It is
expected that some limitations will be addressed by updates or new
extensions resolving the shortcomings. The authors also hope that
clarification on the usefulness of some functionalities in RTP will
result in more complete implementations in the future.
The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired
behaviour and the implications of a particular behaviour depend on
which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behaviour
and their impacts. Some arch-types of RTP usage are discussed.
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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
2.1. Terminology
The following terms and abbreviations are used in this document:
Endpoint: A single entity sending or receiving RTP packets. It may
be decomposed into several functional blocks, but as long as it
behaves a single RTP stack entity it is classified as a single
endpoint.
Multiparty: A communication situation including multiple end-points.
In this document it will be used to refer to situations where more
than two end-points communicate.
Media Source: The source of a stream of data of one Media Type, It
can either be a single media capturing device such as a video
camera, a microphone, or a specific output of a media production
function, such as an audio mixer, or some video editing function.
Sending data from a Media Source may cause multiple RTP sources to
send multiple Media Streams.
Media Stream: A sequence of RTP packets using a single SSRC that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Source: The originator or source of a particular Media Stream.
Identified using an SSRC in a particular RTP session. An RTP
source is the source of a single media stream, and is associated
with a single endpoint and a single Media Source. An RTP Source
is just called a Source in RFC 3550.
Media Sink: A recipient of a Media Stream. The endpoint sinking
media are Identified using one or more SSRCs. There may be more
than one Media Sink for one RTP source.
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CNAME: "Canonical name" - identifier associated with one or more RTP
sources from a single endpoint. Defined in the RTP specification
[RFC3550]. A CNAME identifies a synchronisation context. A CNAME
is associated with a single endpoint, although some RTP nodes will
use an end-points CNAME on that end-points behalf. An endpoint
may use multiple CNAMEs. A CNAME is intended to be globally
unique and stable for the full duration of a communication
session. [RFC6222] gives updated guidelines for choosing CNAMEs.
Media Type: Audio, video, text or data whose form and meaning are
defined by a specific real-time application.
Multiplex: The operation of taking multiple entities as input,
aggregating them onto some common resource while keeping the
individual entities addressable such that they can later be fully
and unambiguously separated (de-multiplexed) again.
RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can receive an SSRC
either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the endpoints' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by endpoints and any interconnecting middle nodes.
RTP Session Group: One or more RTP sessions that are used together
to perform some function. Examples are multiple RTP sessions used
to carry different layers of a layered encoding. In an RTP
Session Group, CNAMEs are assumed to be valid across all RTP
sessions, and designate synchronisation contexts that can cross
RTP sessions.
Source: Term that should not be used alone. An RTP Source, as
identified by its SSRC, is the source of a single Media Stream; a
Media Source can be the source of mutiple Media Streams.
SSRC: An RTP 32-bit unsigned integer used as identifier for a RTP
Source.
CSRC: Contributing Source, A SSRC identifier used in a context, like
the RTP headers CSRC list, where it is clear that the Media Source
is not the source of the media stream, instead only a contributor
to the Media Stream.
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Signalling: The process of configuring endpoints to participate in
one or more RTP sessions.
(tbd: The terms "SSRC multiplexing" and "session multiplexing" are
confusing, with unclear historical meanings; they need to be removed
from this document in the interests of clarity)
2.2. Subjects Out of Scope
This document is focused on issues that affect RTP. Thus, issues
that involve signalling protocols, such as whether SIP, Jingle or
some other protocol is in use for session configuration, the
particular syntaxes used to define RTP session properties, or the
constraints imposed by particular choices in the signalling
protocols, are mentioned only as examples in order to describe the
RTP issues more precisely.
This document assumes the applications will use RTCP. While there
are such applications that don't send RTCP, they do not conform to
the RTP specification, and thus should be regarded as reusing the RTP
packet format, not as implementing the RTP protocol.
3. RTP Concepts
This section describes the existing RTP tools that are particularly
important when discussing multiplexing of different media streams.
3.1. Session
The RTP Session is the highest semantic level in RTP and contains all
of the RTP functionality. RTP itself has no normative statements
about the relationship between different RTP sessions.
Identifier: RTP in itself does not contain any Session identifier,
but relies either on the underlying transport or on the used
signalling protocol, depending on in which context the identifier
is used (e.g. transport or signalling). Due to this, a single RTP
Session may have multiple associated identifiers belonging to
different contexts.
Position: Depending on underlying transport and signalling
protocol. For example, when running RTP on top of UDP, an RTP
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).
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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, but identity can only be detected by
the general communication context for the specific endpoint.
Inter-relation: Depending on the underlying transport and
signalling protocol.
Special Restrictions: None.
A RTP source in an RTP session that changes its source transport
address during a session must also choose a new SSRC identifier to
avoid being interpreted as a looped source.
The set of participants considered part of the same RTP Session is
defined by the RTP specification [RFC3550] as those that share a
single SSRC space. That is, those participants that can see an SSRC
identifier transmitted by 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 Session scope is decided by the
participants' network interconnection topology, in combination with
RTP and RTCP forwarding strategies deployed by endpoints and any
interconnecting middle nodes.
3.2. SSRC
An SSRC identifies a RTP source or a media sink. For end-points that
both source and sink media streams its SSRCs are used in both roles.
At any given time, a RTP source has one and only one SSRC - although
that may change over the lifetime of the RTP source or sink. An RTP
Session serves one or more RTP sources, each sending a Media Stream.
Identifier: Synchronisation Source (SSRC), 32-bit unsigned number.
Position: In every RTP and RTCP packet header. May be present in
RTCP payload. May be present in SDP signalling.
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Usage: Identify individual RTP sources and media sinks within an
RTP Session. Refer to individual RTP sources and media sinks
in RTCP messages and SDP signalling.
Uniqueness: Randomly chosen, intended to be globally unique
within an RTP Session and not dependent on network address.
SSRC value collisions may occur and must be handled as
specified in RTP [RFC3550].
Inter-relation: SSRC belonging to the same synchronisation
context (originating from the same endpoint), 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 RTP 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 RTP source having an SSRC identifier can be of different types:
Real: Connected to a "physical" media source, for example a camera
or microphone.
Conceptual: A source with some attributed property generated by some
network node, for example a filtering function in an RTP mixer
that provides the most active speaker based on some criteria, or a
mix representing a set of other sources.
Media Sink: A source that does not generate any RTP media stream in
itself (e.g. an endpoint only receiving in an RTP session), but
anyway need a sender SSRC for use as source in RTCP reports.
Note that a endpoint that generates more than one media type, e.g. a
conference participant sending both audio and video, need not (and
commonly should not) use the same SSRC value across RTP sessions.
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RTCP Compound packets containing the CNAME SDES item is the
designated method to bind an SSRC to a CNAME, effectively cross-
correlating SSRCs within and between RTP Sessions as coming from the
same endpoint. The main property attributed to SSRCs associated with
the same CNAME is that they are from a particular synchronisation
context and may be synchronised at playback.
Note also that RTP sequence number and RTP timestamp are scoped by
SSRC and thus independent between different SSRCs.
An RTP receiver receiving a previously unseen SSRC value must
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
conflict. However, the originator of the previous SSRC should have
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
effectively a new source.
3.3. CSRC
The Contributing Source (CSRC) is not a separate identifier, but an
usage of the SSRC identifier. It is optionally included in the RTP
header as list of up to 15 contributing RTP sources. CSRC shares the
SSRC number space and 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 contained CSRCs, the media
representation of an individual CSRC is in general not possible to
extract from the RTP payload since it is typically the result of a
media mixing (merge) operation (by an RTP mixer) on the individual
media streams corresponding to the CSRC identifiers. The exception
is the case when only a single CSRC is indicated as this represent
forwarding of a media stream, possibly modified. The RTP header
extension for Mixer-to-Client Audio Level Indication [RFC6465]
expands on the receivers information about a packet with CSRC list.
Due to these restrictions, CSRC will not be considered a fully
qualified multiplex point and will be disregarded in the rest of this
document.
3.4. Payload Type
Each Media Stream utilises one or more encoding formats, identified
by the Payload Type.
The Payload Type is not a multiplexing point. Appendix A gives some
of the many reasons why attempting to use it as a multiplexing point
will have bad results.
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Identifier: Payload Type number.
Position: In every RTP header and in SDP signalling.
Usage: Identify a specific Media Stream encoding format. The
format definition may be taken from [RFC3551] for statically
allocated Payload Types, but should be explicitly defined in
signalling, such as SDP, both for static and dynamic Payload
Types. The term "format" here includes whatever can be
described by out-of-band signalling 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].
Uniqueness: Scoped by sending endpoint within an RTP Session. To
avoid any potential for ambiguity, it is desirable that payload
types are unique across all sending endpoints within an RTP
session, but this is often not true in practice. All SSRC in
an RTP session sent from an single endpoint share the same
Payload Types definitions. The RTP Payload Type is designed
such that only a single Payload Type is valid at any time
instant in the SSRC's RTP timestamp time line, effectively
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 to 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 synchronisation. 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
SSRC that are valid for the same RTP Timestamps, then redundant
encodings [RFC2198] can be used. Several additional constraints than
the ones mentioned above need to be met to enable this use, one of
which is that the combined payload sizes of the different Payload
Types must not exceed the transport MTU.
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Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto].
4. Multiple Streams Alternatives
The reasons why an endpoint may choose to send multiple media streams
are widespread. In the below discussion, please keep in mind that
the reasons for having multiple media streams vary and include but
are not limited to the following:
o Multiple Media Sources
o Multiple Media Streams may be needed to represent one Media Source
(for instance when using layered encodings)
o A Retransmission stream may repeat the content of another Media
Stream
o An FEC stream may provide material that can be used to repair
another Media Stream
o Alternative Encodings, for instance different codecs for the same
audio stream
o Alternative formats, for instance multiple resolutions of the same
video stream
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
different levels of maturity in the discussion on how to solve them.
The clearest understanding is associated with multiple media sources
of the same media type. However, all warrant discussion and
clarification on how to deal with them.
This section reviews the alternatives to enable multi-stream
handling. Let's start with describing mechanisms that could enable
multiple media streams, independent of the purpose for having
multiple streams.
SSRC Multiplexing: Each additional Media Stream gets its own SSRC
within a RTP Session.
Session Multiplexing: Using additional RTP Sessions to handle
additional Media Streams
As the below discussion will show, in reality we cannot choose a
single one of the two solutions. To utilise RTP well and as
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efficiently as possible, both are needed. The real issue is finding
the right guidance on when to create RTP sessions and when additional
SSRCs in an RTP session is the right choice.
5. RTP Topologies and Issues
The impact of how RTP Multiplex is performed will in general vary
with how the RTP Session participants are interconnected; the RTP
Topology [RFC5117]. This section describes the topologies and
attempts to highlight the important behaviours concerning RTP
multiplexing and multi-stream handling. It lists any identified
issues regarding RTP and RTCP handling, and introduces additional
topologies that are supported by RTP beyond those included in RTP
Topologies [RFC5117]. The RTP Topologies that do not follow the RTP
specification or do not attempt to utilise the facilities of RTP are
ignored in this document.
5.1. Point to Point
This is the most basic use case with two endpoints directly
interconnected and no additional entities are involved in the
communication.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
A number of applications are using a single RTP session with one RTP
source per endpoint. This is likely the simplest case, as you
basically doesn't have to make any choices regarding multiplexing.
When you add an additional source to either endpoint you immediately
create the question do one send the media stream in the existing RTP
session or should I use an additional RTP session.
This raises a number of considerations that are discussed in detail
below (Section 6). But the range over such aspects as:
o Does my communication peer support RTP as defined with multiple
SSRCs?
o Do I need network differentiation in form of QoS?
o Can the application more easily process and handle the media
streams if they are in different RTP sessions?
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o etc.
The application designer will have to make choices here. The point
to point topology can contain one to many RTP sessions with one to
many RTP sources (SSRC) per session.
5.1.1. Translators & Gateways
A point to point communication can end up in a situation when the
peer it is communicating with is not compatible with the other peer
for various reasons. This is in many situation resolved by the
inclusion of a translator in-between the two peers.
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
Point to Point with Translator
The translator main purpose is to make the peer look to the other
peer like something it is compatible with. An RTP translator will
commonly not be distinguishable from the actual end-point. It is
intentional not identifiable on RTP level. Reasons a translator can
be required are:
o No common media codec for a media type thus requiring transcoding.
o Different usages of the RTP multiplexing points
o Usage of different media transport protocols
o Usage of different transport protocols
o Different security solutions
The RTP translator will rewrite RTP and RTCP as required to provide a
consistent view to each peer of the traffic the translator forwards
and the feedback being provided back to the RTP source.
In some case security policies or the need for monitoring the media
streams the direct communication are directed to a pass through a
specific middlebox, commonly called a gateway. This is often placed
on the border of administrative domain where the security policies
are in effect. Many gateways simple relay the RTP and RTCP traffic
between the domains, but some may do more by including above
mentioned translator functions or even go as far as terminating the
RTP session and do application level forwarding of the media traffic.
The later places requirements on the gateway to have full
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understanding of the application logic and especially be able to cope
with any congestion control or media adaptation.
A variant of translator behaviour worth pointing out is when an
endpoint A sends a media flow to B. On the path there is a device T
that on A's behalf does something with the media streams, for example
adds an RTP session with FEC information for A's media streams. T
will in this case need to bind the new FEC streams to A's media
stream by using the same CNAME as A.
+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
Figure 2: When De-composition is a Translator
This type of functionality where T does something with the media
stream on behalf of A is clearly covered under the media translator
definition (Section 5.3).
5.2. Point to Multipoint Using Multicast
This section discusses the Point to Multi-point using Multicast to
interconnect the session participants. This needs to consider both
Any Source Multicast (ASM) and Source-Specific Multicast (SSM).
There are large commercial deployments of multicast for applications
like IPTV.
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 3: Point to Multipoint Using Any Source Multicast
In Any Source Multicast, any of the participants can send to all the
other participants, simply by sending a packet to the multicast
group. That is not possible in Source Specific Multicast [RFC4607]
where only a single source (Distribution Source) can send to the
multicast group, creating a topology that looks like the one below:
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+--------+ +-----+
|Media | | | Source-specific
|Sender 1|<----->| D S | Multicast
+--------+ | I O | +--+----------------> R(1)
| S U | | | |
+--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | |
: | U | +--+--> R(n) | | |
: | T +-| | | | |
: | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ |
|Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast
FT = Feedback Target
Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not
co-located.
Figure 4: Point to Multipoint using Source Specific Multicast
In the SSM topology (Figure 4) a number of RTP sources (1 to M) are
allowed to send media to the SSM group. These send media to the
distribution source which then forwards the media streams to the
multicast group. The media streams reach the Receivers (R(1) to
R(n)). The Receivers' RTCP cannot be sent to the multicast group.
To support RTCP, an RTP extension for SSM [RFC5760] was defined to
use unicast transmission 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
into consideration. For example, the only practical method for
adapting the bit-rate sent towards a given receiver for large groups
is to use a set of multicast groups, where each multicast group
represents a particular bit-rate. Otherwise the whole group gets
media adapted to the participant with the worst conditions. The
media encoding is either scalable, where multiple layers can be
combined, or simulcast where a single version is selected. By either
selecting or combing multicast groups, the receiver can control the
bit-rate sent on the path to itself. It is also common that streams
that improve transport robustness are sent in their own multicast
group to allow for interworking with legacy or to support different
levels of protection.
The result of this is some common behaviours for RTP multicast:
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1. Multicast applications use a group of RTP sessions, not one.
Each endpoint will need to be a member of a number of RTP
sessions in order to perform well.
2. Within each RTP session, the number of media sinks is likely to
be much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement; all of
the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It should
be noted that SDP offer/answer [RFC3264] has issues with ensuring
this property. The signalling aspects of multicast are not explored
further in this memo.
Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol must support group communication. Source authentication
becomes more difficult and requires special solutions. For more
discussion on this please review Options for Securing RTP Sessions
[I-D.ietf-avtcore-rtp-security-options].
5.3. Point to Multipoint Using an RTP Transport Translator
This mode is described in section 3.3 of RFC 5117.
Transport Translators (Relays) result in an RTP session situation
that is very similar to how an ASM group RTP session would behave.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 5: Transport Translator (Relay)
An RTP translator forwards both RTP and RTCP packets from all
participants to all other participants.
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One of the most important aspects with the simple relay is that it is
only rewriting transport headers, no RTP modifications nor media
transcoding occur. The most obvious downside of this basic relaying
is that the translator has no control over how many streams need to
be delivered to a receiver. Nor can it simply select to deliver only
certain streams, as this creates session inconsistencies: If the
translator temporarily stops a stream, this prevents some receivers
from reporting on it. From the sender's perspective it will look
like a transport failure. Applications 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
does it need to send any RTCP reports on the flows that pass it.
This as the RTP source will receive feedback for the full path in the
RTCP being sent back. However the transport translator may choose to
send RTCP reports using its own SSRC, as if it itself contained a
media sink, in order to make information about the source-to-
translator link available to monitors.
Use of a transport translator results in that all the endpoints will
receive multiple SSRCs over a single unicast transport flow from the
translator. That is independent of whether the other endpoints have
only a single or several SSRCs.
The Transport Translator has the same signalling requirement as
multicast: All participants must have the same payload type
configuration. Also most of the ASM security issues also arise here.
Some alternative when it comes to solution do exist as there after
all exist a central node to communicate with. One that also can
enforce some security policies depending on the level of trust placed
in the node.
5.4. Point to Multipoint Using an RTP Mixer
An mixer (Figure 6) is a centralised node that selects or mixes
content in a conference to optimise the RTP session so that each
endpoint only needs connect to one entity, the mixer. The mixer can
also reduce the bit-rate needed from the mixer down to a conference
participants as the media sent from the mixer to the end-point can be
optimised in different ways. These optimisations include methods
like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is required
instead of three in the depicted scenario (Figure 6).
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6: RTP Mixer
Mixers has some downsides, the first is that the mixer must be a
trusted node as they either performs media operations or at least
repacketize the media. Both type of operations requires when using
SRTP that the mixer verifies integrity, decrypts the content, perform
its operation and form new RTP packets, encrypts and integrity
protect them. This applies to all types of mixers described below.
The second downside is that all these operations and optimisation of
the session requires processing. How much depends on the
implementation as will become evident below.
A mixer, unlike a pure transport translator, is always application
specific: the application logic for stream mixing or stream selection
has to be embedded within the mixer, and controlled using application
specific signalling. The implementation of an mixer can take several
different forms and we will discuss the main themes available that
doesn't break RTP.
Please note that a Mixer could also contain translator
functionalities, like a media transcoder to adjust the media bit-rate
or codec used on a particular RTP media stream.
5.4.1. Media Mixing
This type of mixer is one which clearly can be called RTP mixer is
likely the one that most thinks of when they hear the term mixer.
Its basic pattern of operation is that it will receive the different
participants RTP media stream. Select which that are to be included
in a media domain mix of the incoming RTP media streams. Then create
a single outgoing stream from this mix.
The most commonly deployed media mixer is probably the audio mixer,
used in voice conferencing, where the output consists of some mixture
of all the input streams; this needs minimal signalling to be
successful. Audio mixing is straight forward and commonly possible
to do for a number of participants. Lets assume that you want to mix
N number of streams from different participants. Then the mixer need
to perform N decodings. Then it needs to produce N or N+1 mixes, the
reasons that different mixes are needed are so that each contributing
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source get a mix which don't contain themselves, as this would result
in an echo. When N is lower than the number of all participants one
may produce a Mix of all N streams for the group that are currently
not included in the mix, thus N+1 mixes. These audio streams are
then encoded again, RTP packetised and sent out.
Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed
video streams can be done. In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of
other sources as smaller overlays is another. Also here one commonly
need to produce a number of different compositions so that the
contributing part doesn't need to see themselves. Then the mixer re-
encodes the created video stream, RTP packetise it and send it out
The problem with media mixing is that it both consume large amount of
media processing and encoding resources. The second is the quality
degradation created by decoding and re-encoding the RTP media stream.
Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition.
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+-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Audio-| |-Audio---+ | +---+ | | |
| | | AA1|--------->|---------+-+-|DEC|->| | |
| | | |<---------|MA1 <----+ | +---+ | | |
| | | | |(BA1+CA1)|\| +---+ | | |
| | +-------| |---------+ +-|ENC|<-| B+C | |
| +---------| |-----------+ +---+ | | |
+-----------+ | | | |
| | M | |
+-B---------+ | | E | |
| +-RTP2----| |-RTP2------+ | D | |
| | +-Audio-| |-Audio---+ | +---+ | I | |
| | | BA1|--------->|---------+-+-|DEC|->| A | |
| | | |<---------|MA2 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+C | |
+-----------+ |-----------+ +---+ | | |
| | M | |
+-C---------+ | | I | |
| +-RTP3----| |-RTP3------+ | X | |
| | +-Audio-| |-Audio---+ | +---+ | E | |
| | | CA1|--------->|---------+-+-|DEC|->| R | |
| | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ |
+----------------------------+
Figure 7: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 7. The mixer present one SSRC towards the peer
client, e.g. MA1 to Peer A, which is the media mix of the other
participants. As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure. It
has to do with the SSRC space and the identity information. When A
receives the MA1 stream which is a combination of BA1 and CA1 streams
in the other PeerConnections RTP could enable the mixer to include
CSRC information in the MA1 stream to identify the contributing
source BA1 and CA1.
The CSRC has in its turn utility in RTP extensions, like the in Mixer
to Client audio levels RTP header extension [RFC6465]. If the SSRC
from endpoint to mixer leg are used as CSRC in another PeerConnection
then RTP1, RTP2 and RTP3 becomes one joint session as they have a
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common SSRC space. At this stage the mixer also need to consider
which RTCP information it need to expose in the different legs. For
the above situation commonly nothing more than the Source Description
(SDES) information and RTCP BYE for CSRC need to be exposed. The
main goal would be to enable the correct binding against the
application logic and other information sources. This also enables
loop detection in the RTP session.
5.4.1.1. RTP Session Termination
There exist an possible implementation choice to have the RTP
sessions being separated between the different legs in the multi-
party communication session and only generate RTP media streams in
each without carrying on RTP/RTCP level any identity information
about the contributing sources. This removes both the functionality
that CSRC can provide and the possibility to use any extensions that
build on CSRC and the loop detection. It may appear a simplification
if SSRC collision would occur between two different end-points as
they can be avoided to be resolved and instead remapped between the
independent sessions if at all exposed. However, SSRC/CSRC remapping
requires that SSRC/CSRC are never used in the application level as
reference. This as they only have local importance, if they are used
on a multi-party session scope the result would be miss-referencing.
Session termination may appear to resolve some issues, it however
creates other issues that needs resolving, like loop detection,
identification of contributing sources and the need to handle mapped
identities and ensure that the right one is used towards the right
identities and never used directly between multiple end-points.
5.4.2. Media Switching
An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption
cycle as one rewrites RTP headers. This both reduces the amount of
computational resources needed in the mixer and increases the media
quality per transmitted bit. This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional
streams the mixer produces. These streams are created by selecting
media from one of the by the mixer received RTP media streams and
forward the media using the mixers own SSRCs. The mixer can then
switch between available sources if that is required by the concept
for the source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field must be set to the value of the Mixer's SSRC.
Secondly, the sequence number must be the next in the sequence of
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outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration may have
to be changed if the different endpoint mixer legs have not arrived
on the same numbering for a given configuration. This also requires
that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still
required.
Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants. Thus the
mixer has two SSRCs sending video to each peer.
+-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------->| S | |
| | | |<------------|MV1 <----+-+-BV1----| W | |
| | | |<------------|MV2 <----+-+-EV1----| I | |
| | +-------| |---------+ | | T | |
| +---------| |-----------+ | C | |
+-----------+ | | H | |
| | | |
+-B---------+ | | M | |
| +-RTP2----| |-RTP2------+ | A | |
| | +-Video-| |-Video---+ | | T | |
| | | BV1|------------>|---------+-+------->| R | |
| | | |<------------|MV3 <----+-+-AV1----| I | |
| | | |<------------|MV4 <----+-+-EV1----| X | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ | | |
+-----------+ | | | |
: : : :
: : : :
+-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | |
| | | CV1|------------>|---------+-+------->| | |
| | | |<------------|MV11 <---+-+-AV1----| | |
| | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+
Figure 8: Media Switching RTP Mixer
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The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participations.
To ensure that a media receiver can correctly decode the RTP media
stream after a switch, it becomes necessary to ensure for state
saving codecs that they start from default state at the point of
switching. Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on
earlier state. This can be done using Full Intra Request [RFC5104]
RTCP codec control message.
Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different end-point and mixer legs. The
same arguments and considerations as discussed in Section 5.4.1.1
applies here.
5.4.3. RTP Source Projecting
Another method for handling media in the RTP mixer is to project all
potential RTP sources (SSRCs) into a per end-point independent RTP
session. The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the
mixer in another RTP session receives media from that end-point.
This is similar to the media switching Mixer but have some important
differences in RTP details.
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+-A---------+ +-MIXER---------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------>| | |
| | | |<------------|BV1 <----+-+-------| S | |
| | | |<------------|CV1 <----+-+-------| W | |
| | | |<------------|DV1 <----+-+-------| I | |
| | | |<------------|EV1 <----+-+-------| T | |
| | | |<------------|FV1 <----+-+-------| C | |
| | +-------| |---------+ | | H | |
| +---------| |-----------+ | | |
+-----------+ | | M | |
| | A | |
+-B---------+ | | T | |
| +-RTP2----| |-RTP2------+ | R | |
| | +-Video-| |-Video---+ | | I | |
| | | BV1|------------>|---------+-+------>| X | |
| | | |<------------|AV1 <----+-+-------| | |
| | | |<------------|CV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|FV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ | | |
+-----------+ | | | |
: : : :
: : : :
+-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | |
| | | CV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+
Figure 9: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 9)
one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1. If this mixer intend to have the same behaviour as
in Section 5.4.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
will concurrently transmit media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end-
points each RTP media stream will require some rewriting when being
projected from one session into another. The main thing is that the
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sequence number will need to be consecutively incremented based on
the packet actually being transmitted in each RTP session. Thus the
RTP sequence number offset will change each time a source is turned
on in a RTP session.
As the RTP sessions are independent the SSRC numbers used can be
handled independently also thus working around any SSRC collisions by
having remapping tables between the RTP sessions. This will result
that each endpoint may have a different view of the application usage
of a particular SSRC. Thus the application must not use SSRC as
references to RTP media streams when communicating with other peers
directly.
The mixer will also be responsible to act on any RTCP codec control
requests coming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need
to implement conference related codec control functionalities to
provide a good experience. Full Intra Request to request from the
media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the
source to mixer link.
This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can at any
time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to
cope with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.
5.5. Point to Multipoint using Multiple Unicast flows
Based on the RTP session definition, it is clearly possible to have a
joint RTP session over multiple transport flows like the below three
endpoint joint session. In this case, A needs to send its' media
streams and RTCP packets to both B and C over their respective
transport flows. As long as all participants do the same, everyone
will have a joint view of the RTP session.
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+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 10: Point to Multi-Point using Multiple Unicast Transports
This doesn't create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session.
Note that an endpoint may use a single local port to receive all
these transport flows, or it might have separate local reception
ports for each of the endpoints.
There exists an alternative structure for establishing the above
communication scenario (Figure 10) 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 endpoint has. The
difference exists in the behaviours 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 Endpoint
The implementation of an application may desire to send a subset of
the application's data to each of multiple devices, each with their
own network address. A very basic use case for this would be to
separate audio and video processing for a particular endpoint, like a
conference room, into one device handling the audio and another
handling the video, being interconnected by some control functions
allowing them to behave as a single endpoint in all aspects except
for transport Figure 11.
Which decomposition that is possible is highly dependent on the RTP
session usage. It is not really feasible to decomposed one logical
end-point into two different transport node in one RTP session. From
a third party monitor of such an attempt the two entities would look
like two different end-points with a CNAME collision. This put a
requirement on that the only type of de-composited endpoint that RTP
really supports is one where the different parts have separate RTP
sessions to send and/or receive media streams intended for them.
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+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+----\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+--------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+----/
| +------------+ |
+---------------------+
Figure 11: De-composite End-Point
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
CNAME and also have a common clock to ensure that synchronisation and
clock drift handling works despite the decomposition. Also the RTCP
handling works correctly as long as only one part of the de-composite
is part of each RTP session. That way any differences in the path
between A's audio entity and B and A's video and B are related to
different SSRCs in different RTP sessions.
The requirements that can derived from the above usage is that the
transport flows for each RTP session might be under common control
but still go to what looks like different endpoints based on
addresses and ports. This geometry cannot be accomplished using one
RTP session, so in this case, multiple RTP sessions are needed.
6. Multiple Streams Discussion
6.1. Introduction
Using multiple media streams is a well supported feature of RTP.
However, it can be unclear for most implementers or people writing
RTP/RTCP applications or extensions attempting to apply multiple
streams when it is most appropriate to add an additional SSRC in an
existing RTP session and when it is better to use multiple RTP
sessions. This section tries to discuss the various considerations
needed. The next section then concludes with some guidelines.
6.2. RTP/RTCP Aspects
This section discusses RTP and RTCP aspects worth considering when
selecting between SSRC multiplexing and Session multiplexing.
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6.2.1. The RTP Specification
RFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Let's review
Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points
should be minimised, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference
composed of audio and video media encoded separately, each medium
SHOULD be carried in a separate RTP session with its own destination
transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings.
2. An SSRC is defined to identify a single timing and sequence
number space. Interleaving multiple payload types would require
different timing spaces if the media clock rates differ and would
require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
Using a different SSRC for each medium but sending them in the same
RTP session would avoid the first three problems but not the last
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two.
On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It may also be appropriate
to multiplex streams of the same medium using different SSRC values
in other scenarios where the last two problems do not apply."
Let's consider one argument at a time. The first is an argument for
using different SSRC for each individual media stream, which still is
very applicable.
The second argument is advocating against using payload type
multiplexing, which still stands as can been seen by the extensive
list of issues found in Appendix A.
The third argument is yet another argument against payload type
multiplexing.
The fourth is an argument against multiplexing media streams that
require different handling into the same session. As we saw in the
discussion of RTP mixers, the RTP mixer has to embed application
logic in order to handle streams anyway; the separation of streams
according to stream type is just another piece of application logic,
which may or may not be appropriate for a particular application. A
type of application that can mix different media sources "blindly" is
the telephone bridge; most other type of application needs
application-specific logic to perform the mix correctly.
The fifth argument discusses network aspects that we will discuss
more below in Section 6.4. It also goes into aspects of
implementation, like decomposed endpoints where different processes
or inter-connected devices handle different aspects of the whole
multi-media session.
A summary of RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its own media/packet stream, and to use
different RTP sessions for media streams that don't share media type.
The first this document support as very valid. The later is one
thing which is further discussed in this document as something the
application developer needs to make a continuous choice for.
6.2.1.1. Different Media Types Recommendations
The above quote from RTP [RFC3550] includes a strong recommendation:
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"For example, in a teleconference composed of audio and video
media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address."
It was identified in "Why RTP Sessions Should Be Content Neutral"
[I-D.alvestrand-rtp-sess-neutral] that the above statement is poorly
supported by any of the motivations provided in the RTP
specification. This has resulted in the creation of a specification
Multiple Media Types in an RTP Session specification
[I-D.westerlund-avtcore-multi-media-rtp-session] which intend to
update this recommendation. That document has a detailed analysis of
the potential issues in having multiple media types in the same RTP
session. This document tries to provide an more over arching
consideration regarding the usage of RTP session and considers
multiple media types in one RTP session as possible choice for the
RTP application designer.
6.2.2. Multiple SSRCs in a Session
In cases when an endpoint uses multiple SSRCs, we have found two
closely related issues. The first is if every SSRC shall report on
all other SSRC, even the ones originating from the same endpoint.
The reason for this would be to ensure that no monitoring function
should suspect a breakage in the RTP session. No monitoring function
that gives an alert on non-reporting of an endpoint's own SSRCs has
been identified.
The second issue around RTCP reporting arise when an endpoint
receives one or more media streams, and when the receiving endpoint
itself sends multiple SSRC in the same RTP session. As transport
statistics are gathered per endpoint and shared between the nodes,
all the endpoint's SSRC will report based on the same received data,
the only difference will be which SSRCs sends the report. This could
be considered unnecessary overhead, but for consistency it might be
simplest to always have all sending SSRCs send RTCP reports on all
media streams the endpoint receives.
The current RTP text is silent about sending RTCP Receiver Reports
for an endpoint's own sources, but does not preclude either sending
or omitting them. The uncertainty in the expected behaviour in those
cases has likely caused variations in the implementation strategy.
This could cause an interoperability issue where it is not possible
to determine if the lack of reports is a true transport issue, or
simply a result of implementation.
Although this issue is valid already for the simple point to point
case, it needs to be considered in all topologies. From the
perspective of an endpoint, any solution needs to take into account
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what a particular endpoint can determine without explicit information
of the topology. For example, a Transport Translator (Relay)
topology will look quite similar to point to point on a transport
level but is different on RTP level. Assume a first scenario with
two SSRC being sent from an endpoint to a Transport Translator, and a
second scenario with two single SSRC remote endpoints sending to the
same Transport Translator. The main 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.
When an endpoint has multiple SSRCs and it needs to send RTCP packets
on behalf of these SSRCs, the question arises how RTCP packets with
different source SSRCs can be sent in the same compound packet. It
appears allowed, however some consideration of the transmission
scheduling is needed.
These issues are currently being discussed and a recommendation for
how to handle them are developed in "Real-Time Transport Protocol
(RTP) Considerations for Endpoints Sending Multiple Media Streams"
[I-D.lennox-avtcore-rtp-multi-stream].
6.2.3. Handling Varying sets of Senders
In some applications, the set of simultaneously active sources varies
within a larger set of session members. A receiver can then possibly
try to use a set of decoding chains that is smaller than the number
of senders, switching the decoding chains between different senders.
As each media decoding chain may contain state, either the receiver
must either be able to save the state of swapped-out senders, or the
sender must be able to send data that permits the receiver to
reinitialise when it resumes activity.
This behaviour will cause similar issues independent of SSRC or
Session multiplexing.
6.2.4. Cross Session RTCP Requests
There currently exists no functionality to make truly synchronised
and atomic RTCP messages with some type of request semantics across
multiple RTP Sessions. Instead, separate RTCP messages will have to
be sent in each session. This gives SSRC multiplexed streams a
slight advantage as RTCP messages for different streams in the same
session can be sent in a compound RTCP packet. Thus providing an
atomic operation if different modifications of different streams are
requested at the same time.
In Session multiplexed cases, the RTCP timing rules in the sessions
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and the transport aspects, such as packet loss and jitter, prevents a
receiver from relying on atomic operations, forcing it to use more
robust and forgiving mechanisms.
6.2.5. Binding Related Sources
A common problem in a number of various RTP extensions has been how
to bind related sources together. This issue is common to SSRC
multiplexing and Session Multiplexing.
Most, if not all, solutions to this problem are implemented in the
signalling plane, providing metadata information using SDP.
There exists one solution for grouping RTP sessions together in SDP
[RFC5888] to know which RTP session contains for example the FEC data
for the source data in another session. However, this mechanism does
not work on individual media flows and is thus not directly
applicable to the problem. The other solution is also SDP based and
can group SSRCs within a single RTP session [RFC5576]. Thus this
mechanism can bind media streams in SSRC multiplexed cases. Both
solutions have the shortcoming of being restricted to SDP based
signalling and also do not work in cases where the session's dynamic
properties are such that it is difficult or resource consuming to
keep the list of related SSRCs up to date.
One possible solution could be to mandate the same SSRC value being
used in all RTP session in case of session multiplexing. We do note
that Section 8.3 of the RTP Specification [RFC3550] recommends using
a single SSRC space across all RTP sessions for layered coding.
However this recommendation has some downsides and is less applicable
beyond the field of layered coding. To use the same sender SSRC in
all RTP sessions from a particular endpoint can cause issues if an
SSRC collision occurs. If the same SSRC is used as the required
binding between the streams, then all streams in the related RTP
sessions must change their SSRC. This is extra likely to cause
problems if the participant populations are different in the
different sessions. For example, in case of large number of
receivers having selected totally random SSRC values in each RTP
session as RFC 3550 specifies, a change due to a SSRC collision in
one session can then cause a new collision in another session. This
cascading effect is not severe but there is an increased risk that
this occurs for well populated sessions (the birthday paradox ensures
that if you populate a single session with 9292 SSRCs at random, the
chances are approximately 1% that at least one collision will occur).
In addition, being forced to change the SSRC affects all the related
media streams; instead of having to resynchronise only the originally
conflicting stream, all streams will suddenly need to be
resynchronised with each other. This will prevent also the media
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streams not having an actual collision from being usable during the
resynchronisation and also increases the time until synchronisation
is finalised. In addition, it requires exception handling in the
SSRC generation.
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
at least one session, like the base layer in layered encoding. In
that case the only downside is the special behaviour that needs to be
well defined by anyone using this. But, having an exception
behaviour where the SSRC space is common across all session is an
issue as this behaviour does not fit all the RTP extensions or
payload formats. It is possible to create a situation where the
different mechanisms cannot be combined due to the non standard SSRC
allocation behaviour.
Existing mechanisms with known issues:
RTP Retransmission: [RFC4588] Has two modes, one for SSRC
multiplexing and one for Session multiplexing. The session
multiplexing requires the same CNAME and mandates that the same
SSRC is used in both sessions. Using the same SSRC does work but
will as previously stated potentially have issues in certain
cases. In SSRC multiplexed mode the CNAME is used to bind media
and retransmission streams together. However, if multiple media
streams are sent from the same endpoint in the same session this
does not provide non-ambiguous binding. Therefore when the first
retransmission request for a media stream is sent, one must not
have another retransmission request outstanding for an SSRC which
don't have a binding between the original SSRC and the
retransmission stream's SSRC. This works but creates some
limitations that can be avoided by a explicit mechanism. The SDP
based ssrc-group mechanism would be sufficient in this case as
long as the application can rely on the signalling based solution.
Scalable Video Coding : As an example of scalable coding, SVC
[RFC6190] has various modes. The Multi Session Transmission (MST)
uses Session multiplexing to separate scalability layers.
However, this specification has failed to be explicit on how these
layers are bound together in cases where CNAME is not sufficient.
CNAME is no longer sufficient when more than one media source
occur within a session that has the same CNAME, for example due to
multiple video cameras capturing the same lecture hall. This
likely implies that a single SSRC space as recommend by Section
8.3 of RTP [RFC3550] is to be used.
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Forward Error Correction: If some type of FEC or redundancy stream
is being sent, it needs its own SSRC, with the exception of
constructions like redundancy encoding [RFC2198]. Thus in case of
transmitting the FEC in the same session as the source data, the
inter SSRC relation within a session is needed. In case of
sending the redundant data in a separate session from the source,
the SSRC in each session needs to be related. This occurs for
example in RFC5109 when using session separation of original and
FEC data. SSRC multiplexing is not supported, only using
redundant encoding is supported.
This issue appears to need action to harmonise and avoid future
shortcomings in extension specifications. A proposed solution for
handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].
6.2.6. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes
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
achieved by transmitting a certain amount of redundant information
that is encoded such that it can repair one or more packet loss over
the set of packets they protect. This sequence of redundant
information also needs to be transmitted as its own media stream, or
in some cases instead of the original media stream. Thus many of
these schemes create a need for binding the related flows as
discussed above. They also create additional flows that need to be
transported. Looking at the history of these schemes, there is both
SSRC multiplexed and Session multiplexed solutions and some schemes
that support both.
Using a Session multiplexed solution supports the case where some set
of receivers may not be able to utilise 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
multicast group, and therefore in its own RTP session, allows for
flexibility, for example when using Rapid Acquisition of Multicast
Groups (RAMS) [RFC6285]. During the RAMS burst where data is
received over unicast and where it is possible to combine with
unicast based retransmission [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 multicast group. This saves bandwidth to the
receiver during the 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 time join the multicast group with the FEC
information. Having the source stream and the FEC in separate groups
allows for easy separation in the Burst/Retransmission Source (BRS)
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without having to individually classify packets.
6.2.7. Transport Translator Sessions
A basic Transport Translator relays any incoming RTP and RTCP packets
to the other participants. The main difference between SSRC
multiplexing and Session multiplexing resulting from this use case is
that for SSRC multiplexing it is not possible for a particular
session participant to decide to receive a subset of media streams.
When using separate RTP sessions for the different sets of media
streams, a single participant can choose to leave one of the sessions
but not the other.
6.3. Interworking
There are several different kinds of interworking, and this section
discusses two related ones. The interworking between different
applications and the implications of potentially different choices of
usage of RTP's multiplexing points. The second topic relates to what
limitations may have to be considered working with some legacy
applications.
6.3.1. Types of Interworking
It is not uncommon that applications or services of similar usage,
especially the ones intended for interactive communication, ends up
in a situation where one want to interconnect two or more of these
applications.
In these cases one ends up in a situation where one might use a
gateway to interconnect applications. This gateway then needs to
change the multiplexing structure or adhere to limitations in each
application.
There are two fundamental approaches to gatewaying: RTP bridging,
where the gateway acts as an RTP Translator, and the two applications
are members of the same RTP session, and RTP termination, where there
are independent RTP sessions running from each interconnected
application to the gateway.
6.3.2. RTP Translator Interworking
From an RTP perspective the RTP Translator approach could work if all
the applications are using the same codecs with the same payload
types, 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.
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When one is gatewaying via an RTP Translator, a natural requirement
is that the two applications being interconnected must use the same
approach to multiplexing. Furthermore, if one of the applications is
capable of working in several modes (such as being able to use SSRC
multiplexing or RTP session multiplexing at will), and the other one
is not, successful interconnection depends on locking the more
flexible application into the operating mode where interconnection
can be successful, even if no participants using the less flexible
application are present when the RTP sessions are being created.
6.3.3. Gateway Interworking
When one terminates RTP sessions at the gateway, there are certain
tasks that the gateway must carry out:
o Generating appropriate RTCP reports for all media streams
(possibly based on incoming RTCP reports), originating from SSRCs
controlled by the gateway.
o Handling SSRC collision resolution in each application's RTP
sessions.
o Signalling, choosing and policing appropriate bit-rates for each
session.
If either of the applications has any security applied, e.g. in the
form of SRTP, the gateway must be able to decrypt incoming packets
and re-encrypt them in the other application's security context.
This is necessary even if all that's required is a simple remapping
of SSRC numbers. If this is done, the gateway also needs to be a
member of the security contexts of both sides, of course.
Other tasks a gateway may need to apply include transcoding (for
incompatible codec types), rescaling (for incompatible video size
requirements), suppression of content that is known not to be handled
in the destination application, or the addition or removal of
redundancy coding or scalability layers to fit the need of the
destination domain.
From the above, we can see that the gateway needs to have an intimate
knowledge of the application requirements; a gateway is by its nature
application specific, not a commodity product.
This fact reveals 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.
If one uses security functions, like SRTP, they can as seen above
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incur both additional risk due to the gateway needing to be in
security association between the endpoints, unless the gateway is on
the transport level, and additional complexities in form of the
decrypt-encrypt cycles needed for each forwarded packet. SRTP, due
to its keying structure, also requires that each RTP session must
have different master keys, as use of the same key in two RTP
sessions can result in two-time pads that completely breaks the
confidentiality of the packets.
6.3.4. 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 endpoint 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 endpoint and the
RTP mixer, meaning that the mixer functions as an RTP-terminating
gateway.
When establishing RTP sessions that may contain endpoints 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.
This indicates that gateways attempting to interconnect to this class
of devices must make sure that only one media stream of each type
gets delivered to the endpoint if it's expecting only one, and that
the multiplexing format is what the device expects. It is highly
unlikely that RTP translator-based interworking can be made to
function successfully in such a context.
6.4. Network Aspects
The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor.
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6.4.1. Quality of Service
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
mechanism, while Diff-Serv [RFC2474] is an example of a Marking based
one. For a marking based scheme, the method of multiplexing will not
affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between
the methods. SSRC multiplexing will result in all media streams
being part of the same 5-tuple (protocol, source address, destination
address, source port, destination port) which is the most common
selector for flow based QoS. Thus, separation of the level of QoS
between media streams is not possible. That is however possible for
session based multiplexing, where each media stream for which a
separate QoS handling is desired can be in a different RTP session
that can be sent over different 5-tuples.
6.4.2. NAT and Firewall Traversal
In today's network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW).
Below we analyze and comment on the impact of requiring more
underlying transport flows in the presence of NATs and Firewalls:
End-Point Port Consumption: A given IP address only has 65536
available local ports per transport protocol for all consumers of
ports that exist on the machine. This is normally never an issue
for an end-user machine. It can become an issue for servers that
handle large number of simultaneous streams. However, if the
application uses ICE to authenticate STUN requests, a server can
serve multiple endpoints from the same local port, and use the
whole 5-tuple (source and destination address, source and
destination port, protocol) as identifier of flows after having
securely bound them to the remote endpoint address using the STUN
request. In theory the minimum number of media server ports
needed are the maximum number of simultaneous RTP Sessions a
single endpoint may use. In practice, implementation will
probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks.
NAT State: If an endpoint sits behind a NAT, each flow it generates
to an external address will result in a state that has to be kept
in the NAT. That state is a limited resource. In home or Small
Office/Home Office (SOHO) NATs, memory or processing are usually
the most limited resources. For large scale NATs serving many
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internal endpoints, available external ports are typically the
scarce resource. Port limitations is primarily a problem for
larger centralised NATs where endpoint independent mapping
requires each flow to use one port for the external IP address.
This affects the maximum number of internal users per external IP
address. However, it is worth pointing out that a real-time video
conference session with audio and video is likely using less than
10 UDP flows, compared to certain web applications that can use
100+ TCP flows to various servers from a single browser instance.
NAT Traversal Excess Time: Making the NAT/FW traversal takes a
certain amount of time for each flow. It also takes time in a
phase of communication between accepting to communicate and the
media path being established which is fairly critical. The best
case scenario for how much extra time it takes after finding the
first valid candidate pair following the specified ICE procedures
are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing
timer, which ICE specifies to be no smaller than 20 ms. That
assumes a message in one direction, and then an immediate
triggered check back. The reason it isn't more is that ICE first
finds one candidate pair that works prior to attempting to
establish multiple flows. Thus, there is no extra time until one
has found a working candidate pair. Based on that working pair
the needed extra time is to in parallel establish the, in most
cases 2-3, additional flows. However, packet loss causes extra
delays, at least 100 ms which is the minimal retransmission timer
for ICE.
NAT Traversal Failure Rate: Due to the need to establish more than a
single flow through the NAT, there is some risk that establishing
the first flow succeeds but that one or more of the additional
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
just been successfully established. Thus only rare events such as
NAT resource overload, or selecting particular port numbers that
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 the additional media streams within one RTP
Session and does not introduce any additional NAT traversal
complexities per media stream. This can be compared with normally
one or two additional transport flows per RTP session when using
session multiplexing. Additional lower layer transport flows will be
required, unless an explicit de-multiplexing layer is added between
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RTP and the transport protocol. A proposal for how to multiplex
multiple RTP sessions over the same single lower layer transport
exist in [I-D.westerlund-avtcore-transport-multiplexing].
6.4.3. Multicast
Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like
behaviours with one endpoint transmitting to a large number of
receivers, like in IPTV. But that same semantics do result in a
certain number of limitations.
One limitation is that for any group, sender side adaptation to the
actual receiver properties causes degradation for all participants to
what is supported by the receiver with the worst conditions among the
group participants. In most cases this is not acceptable. Instead
various receiver based solutions are employed to ensure that the
receivers achieve best possible performance. By using scalable
encoding and placing each scalability layer in a different multicast
group, the receiver can control the amount of traffic it receives.
To have each scalability layer on a different multicast group, one
RTP session per multicast group is used.
RTP can't function correctly if media streams sent over different
multicast groups where considered part of the same RTP session.
First of all the different layers needs different SSRCs or the
sequence number space seen for a receiver of any sub set of the
layers would have sender side holes. Thus triggering packet loss
reactions. Also any RTCP reporting of such a session would be non
consistent and making it difficult for the sender to determine the
sessions actual state.
Thus it appears easiest and most straightforward to use multiple RTP
sessions. In addition, the transport flow considerations in
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
space.
6.4.4. Multiplexing multiple RTP Session on a Single Transport
For applications that doesn't need flow based QoS and like to save
ports and NAT/FW traversal costs and where usage of multiple media
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 [I-D.westerlund-avtcore-transport-multiplexing].
Using such a solution would allow session multiplexing without most
of the perceived downsides of additional RTP sessions creating a need
for additional transport flows.
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6.5. Security Aspects
When dealing with point-to-point, 2-member RTP sessions only, there
are few security issues that are relevant to the choice of having one
RTP session or multiple RTP sessions. However, there are a few
aspects of multiparty sessions that might warrant consideration.
6.5.1. Security Context Scope
When using SRTP [RFC3711] the security context scope is important and
can be a necessary differentiation in some applications. As SRTP's
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
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
access to the key. In most cases this is a sufficient security
property, but there are a few cases where this does create
situations.
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
media streams. This requires that everyone re-keys without
disclosing the keys to the excluded party.
A second case is when using security as an enforcing mechanism for
differentiation. Take for example a scalable layer or a high quality
simulcast version which only premium users are allowed to access.
The mechanism preventing a receiver from getting the high quality
stream can be based on the stream being encrypted with a key that
user can't access without paying premium, having the key-management
limit access to the key.
SRTP [RFC3711] has not special functions for dealing with different
sets of master keys for different SSRCs. The key-management
functions has different capabilities to establish different set of
keys, normally on a per end-point basis. DTLS-SRTP [RFC5764] and
Security Descriptions [RFC4568] for example establish different keys
for outgoing and incoming traffic from an end-point. This key usage
must be written into the cryptographic context, possibly associated
with different SSRCs.
6.5.2. Key Management for Multi-party session
Performing key-management for multi-party session can be a challenge.
This section considers some of the issues.
Multi-party sessions, such as transport translator based sessions and
multicast sessions, cannot use Security Description [RFC4568] nor
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DTLS-SRTP [RFC5764] without an extension as each endpoint provides
its set of keys. In centralised conference, the signalling
counterpart is a conference server and the media plane unicast
counterpart (to which DTLS messages would be sent) is the transport
translator. Thus an extension like Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution
that allows for keying all session participants with the same master
key.
6.5.3. 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
an RTP translator or mixer to rewrite an SSRC value in the RTP packet
of an unencrypted session, the cost of doing it when using
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.
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 endpoint 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
endpoints have one media stream each. If the application needs
additional media flows between the endpoints, 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.
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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 endpoints.
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 centralised
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
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have issues and require SSRC translation.
g. Cannot be used with Any Source Multicast (ASM) as one cannot
guarantee that only two endpoints participate as packet senders.
Using SSM, it is possible to restrict to these requirements if no
RTCP feedback is 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 endpoint, 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
behaviour 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 endpoint or due to multiple endpoints. 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.
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3. Works well with media type de-composite endpoints.
4. Enables Flow-based QoS with different prioritisation 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 endpoint.
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 prioritisation between media types. It
handles multi-party session well, independently of multicast or
centralised 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 prioritisation of media
streams due to their usage in the application, or the need for fine
granular signalling using today's tools.
The Pros:
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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 prioritisation for flow based mechanisms.
5. Works well with de-composite endpoints.
6. Handles dynamic usage of media streams well.
7. For transport translator based multi-party sessions, this
structure allows for improved control of which type of media
streams an endpoint 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 synchronised 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 endpoint.
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.
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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
endpoint 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 endpoint or central
processing can still be supported despite single session.
3. Can handle dynamic allocations of media streams well on an RTP
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
4. Minimal overhead for security association establishment.
The Cons:
a. Less suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a
single media type, due to need of 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.
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c. Not suitable for de-composite end-points as it requires higher
bandwidth and processing.
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 endpoint, 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 ??? 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
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represent points in between the first two when it comes to amount of
RTP sessions established, i.e. representing an attempt to reduce the
amount of sessions as much as possible without compromising the
functionality the session provides both on network level and on
signalling level.
8. Summary considerations and guidelines
8.1. Guidelines
This section contains a number of recommendations for implementors or
specification writers when it comes to handling multi-stream.
Do not Require the same SSRC across Sessions: As discussed in
Section 6.2.5 there exist drawbacks in using the same SSRC in
multiple RTP sessions as a mechanism to bind related media streams
together. It is instead recommended that a mechanism to
explicitly signal the relation is used, either in RTP/RTCP or in
the used signalling mechanism that establishes the RTP session(s).
Use SSRC multiplexing for additional Media Sources: In the cases an
RTP endpoint needs to transmit additional media streams of the
same media type in the application, with the same processing
requirements at the network and RTP layers, it is recommended to
send them as additional SSRCs in the same RTP session. For
example a telepresence room where there are three cameras, and
each camera captures 2 persons sitting at the table, sending each
camera as its own SSRC within a single RTP session is recommended.
Use additional RTP sessions for streams with different requirements:
When media streams have different processing requirements from the
network or the RTP layer at the endpoints, it is recommended that
the different types of streams are put in different RTP sessions.
This includes the case where different participants want different
subsets of the set of RTP streams.
When using Session Multiplexing use grouping: When using Session
Multiplexing solutions, it is recommended to be explicitly group
the involved RTP sessions using the signalling mechanism, for
example The Session Description Protocol (SDP) Grouping Framework.
[RFC5888], using some appropriate grouping semantics.
RTP/RTCP Extensions May Support SSRC and Session Multiplexing: When
defining an RTP or RTCP extension, the creator needs to consider
if this extension is applicable in both SSRC multiplexed and
Session multiplexed usages. Any extension intended to be generic
is recommended to support both. Applications that are not as
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generally applicable will have to consider if interoperability is
better served by defining a single solution or providing both
options.
Transport Support Extensions: When defining new RTP/RTCP extensions
intended for transport support, like the retransmission or FEC
mechanisms, they are recommended to include support for both SSRC
and Session multiplexing so that application developers can choose
freely from the set of mechanisms without concerning themselves
with which of the multiplexing choices a particular solution
supports.
9. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
10. Security Considerations
There is discussion of the security implications of choosing SSRC vs
Session multiplexing in Section 6.5.
11. References
11.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
11.2. Informative References
[ALF] Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols", SIGCOMM
Symposium on Communications Architectures and
Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
Computer Communications Review, Vol. 20(4),
September 1990.
[I-D.alvestrand-rtp-sess-neutral]
Alvestrand, H., "Why RTP Sessions Should Be Content
Neutral", draft-alvestrand-rtp-sess-neutral-01 (work in
progress), June 2012.
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[I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011.
[I-D.ietf-avtcore-rtp-security-options]
Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", draft-ietf-avtcore-rtp-security-options-00
(work in progress), July 2012.
[I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session",
draft-ietf-avtext-multiple-clock-rates-05 (work in
progress), May 2012.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-00 (work in
progress), February 2012.
[I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-02 (work in progress),
July 2012.
[I-D.lennox-avtcore-rtp-multi-stream]
Lennox, J. and M. Westerlund, "Real-Time Transport
Protocol (RTP) Considerations for Endpoints Sending
Multiple Media Streams",
draft-lennox-avtcore-rtp-multi-stream-00 (work in
progress), July 2012.
[I-D.lennox-mmusic-sdp-source-selection]
Lennox, J. and H. Schulzrinne, "Mechanisms for Media
Source Selection in the Session Description Protocol
(SDP)", draft-lennox-mmusic-sdp-source-selection-04 (work
in progress), March 2012.
[I-D.westerlund-avtcore-max-ssrc]
Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling",
draft-westerlund-avtcore-max-ssrc-01 (work in progress),
April 2012.
[I-D.westerlund-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple
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Media Types in an RTP Session",
draft-westerlund-avtcore-multi-media-rtp-session-00 (work
in progress), July 2012.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport",
draft-westerlund-avtcore-transport-multiplexing-02 (work
in progress), March 2012.
[I-D.westerlund-avtext-rtcp-sdes-srcname]
Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES
Item SRCNAME to Label Individual Sources",
draft-westerlund-avtext-rtcp-sdes-srcname-00 (work in
progress), October 2011.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S., and S.
Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
Functional Specification", RFC 2205, September 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 1998.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
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[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, July 2009.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010.
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[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 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
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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.
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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 endpoint 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.
Appendix B. Proposals for Future Work
The above discussion and guidelines indicates that a small set of
extension mechanisms could greatly improve the situation when it
comes to using multiple streams independently of Session multiplexing
or SSRC multiplexing. These extensions are:
Media Source Identification: A Media source identification that can
be used to bind together media streams that are related to the
same media source. A proposal
[I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
item SRCNAME that also can be used with the a=ssrc SDP attribute
to provide signalling layer binding information.
SSRC limitations within RTP sessions: By providing a signalling
solution that allows the signalling peers to explicitly express
both support and limitations on how many simultaneous media
streams an endpoint can handle within a given RTP Session. That
ensures that usage of SSRC multiplexing occurs when supported and
without overloading an endpoint. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc].
Appendix C. RTP Specification Clarifications
This section describes a number of clarifications to the RTP
specifications that are likely necessary for aligned behaviour when
RTP sessions contain more SSRCs than one local and one remote.
All of the below proposals are under consideration in
[I-D.lennox-avtcore-rtp-multi-stream].
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C.1. RTCP Reporting from all SSRCs
When one has multiple SSRC in an RTP node, all these SSRC must send
some RTP or RTCP packet as long as the SSRC exist. It is not
sufficient that only one SSRC in the node sends report blocks on the
incoming RTP streams; any SSRC that intends to remain in the session
must send some packets to avoid timing out according to the rules in
RFC 3550 section 6.3.5.
It has been hypothesised that a third party monitor may be confused
by not necessarily being able to determine that all these SSRC are in
fact co-located and originate from the same stack instance; if this
hypothesis is true, this may argue for having all the sources send
full reception reports, even though they are reporting the same
packet delivery.
The contrary argument is that such double reporting may confuse the
third party monitor even more by making it seem that utilisation of
the last-hop link to the recipient is (number of SSRCs) times higher
than what it actually is.
C.2. RTCP Self-reporting
For any RTP node that sends more than one SSRC, there is the question
if SSRC1 needs to report its reception of SSRC2 and vice versa. The
reason that they in fact need to report on all other local streams as
being received is report consistency. The hypothetical third party
monitor that considers the full matrix of media streams and all known
SSRC reports on these media streams would detect a gap in the reports
which could be a transport issue unless identified as in fact being
sources from the same node.
C.3. Combined RTCP Packets
When a node contains multiple SSRCs, it is questionable if an RTCP
compound packet can only contain RTCP packets from a single SSRC or
if multiple SSRCs can include their packets in a joint compound
packet. The high level question is a matter for any receiver
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
existing rules are focused on determining when a single SSRC can
send.
Appendix D. Signalling considerations
Signalling is not an architectural consideration for RTP itself, so
this discussion has been moved to an appendix. However, it is hugely
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important for anyone building complete applications, so it is
deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with
signalling; they cannot be addressed by tweaking, extending or
profiling RTP.
D.1. Signalling Aspects
There exist various signalling solutions for establishing RTP
sessions. Many are SDP [RFC4566] based, however SDP functionality is
also dependent on the signalling protocols carrying the SDP. Where
RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative
fashion, while SIP [RFC3261] uses SDP with the additional definition
of Offer/Answer [RFC3264]. The impact on signalling and especially
SDP needs to be considered as it can greatly affect how to deploy a
certain multiplexing point choice.
D.1.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused around
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/
media description but those are not necessarily strictly bound to an
RTP session and could be of interest to signal specifically for a
particular media stream (SSRC) within the session. The following
properties have been identified as being potentially useful to signal
not only on RTP session level:
o Bitrate/Bandwidth exist today only at aggregate or a common any
media stream limit, unless either codec-specific bandwidth
limiting or RTCP signalling using TMMBR is used.
o Which SSRC that will use which RTP Payload Types (this will be
visible from the first media packet, but is sometimes useful to
know before packet arrival).
Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy an SSRC multiplexed
solution that contains several sets of media streams with different
properties (encoding/packetization parameter, bit-rate, etc), putting
each set in a different RTP session would directly enable negotiation
of the parameters for each set. If insisting on SSRC multiplexing
only, a number of signalling extensions are needed to clarify that
there are multiple sets of media streams with different properties
and that they shall in fact be kept different, since a single set
will not satisfy the application's requirements.
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For some parameters, such as resolution and framerate, a SSRC-linked
mechanism has been proposed:
[I-D.lennox-mmusic-sdp-source-selection].
D.1.2. SDP Prevents Multiple Media Types
SDP chose to use the m= line both to delineate an RTP session and to
specify the top level of the MIME media type; audio, video, text,
image, application. This media type is used as the top-level media
type for identifying the actual payload format bound to a particular
payload type using the rtpmap attribute. This binding has to be
loosened in order to use SDP to describe RTP sessions containing
multiple MIME top level types.
There is an accepted WG item in the MMUSIC WG to define how multiple
media lines describe a single underlying transport
[I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible
in SDP to define one RTP session with media types having different
MIME top level types.
D.1.3. Signalling 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, an
application may choose to take all incoming audio RTP streams, mix
them and play them 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. The signalling
used will have to identify the media streams affected by their RTP-
level identifiers, which means that they have to be identified either
by their session or by their SSRC + session.
In some applications, the receiver cannot utilise the media stream at
all before it has received the signalling message describing the
media stream and its usage. In other applications, there exists a
default handling that is appropriate.
If all media streams in an RTP session are to be treated in the same
way, identifying the session is enough. If SSRCs in a session are to
be treated differently, signalling must identify both the session and
the SSRC.
If this signalling affects how any RTP central node, like an RTP
mixer or translator that selects, mixes or processes streams, treats
the streams, the node will also need to receive the same signalling
to know how to treat media streams with different usage in the right
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fashion.
Appendix E. Changes from -01 to -02
o Added Harald Alvestrand as co-author.
o Removed unused term "Media aggregate".
o Added term "RTP session group", noted that CNAMEs are assumed to
bind across the sessions of an RTP session group, and used it when
appropriate (TODO)
o Moved discussion of signalling aspects to appendix
o Removed all suggestion that PT can be a multiplexing point
o Normalised spelling of "endpoint" to follow RFC 3550 and not use a
hyphen.
o Added CNAME to definition list.
o Added term "Media Sink" for the thing that is identified by a
listen-only SSRC.
o Added term "RTP source" for the thing that transmits one media
stream, separating it from "Media Source". [[OUTSTANDING: Whether
to use "RTP Source" or "Media Sender" here]]
o Rewrote section on distributed endpoint, noting that this, like
any endpoint that wants a subset of a set of RTP streams, needs
multiple RTP sessions.
o Removed all substantive references to the undefined term "purpose"
from the main body of the document when it referred to the purpose
of an RTP stream.
o Moved the summary section of section 6 to the guidelines section
that it most closely supports.
o
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Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Bo Burman
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Harald Tveit Alvestrand
Google
Kungsbron 2
Stockholm, 11122
Sweden
Phone:
Fax:
Email: harald@alvestrand.no
URI:
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