RTP Multiplexing Architecture
draft-westerlund-avtcore-multiplex-architecture-01
The information below is for an old version of the document.
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| Authors | Magnus Westerlund , Bo Burman , Colin Perkins | ||
| Last updated | 2012-03-12 | ||
| Replaced by | draft-ietf-avtcore-multiplex-guidelines, draft-ietf-avtcore-multiplex-guidelines, RFC 8872 | ||
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draft-westerlund-avtcore-multiplex-architecture-01
Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Informational Ericsson
Expires: September 13, 2012 C. Perkins
University of Glasgow
March 12, 2012
RTP Multiplexing Architecture
draft-westerlund-avtcore-multiplex-architecture-01
Abstract
Real-time Transport Protocol 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, the
Synchronization Source Identifier (SSRC), and the payload type. The
focus is put on the first two, trying 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-
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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 September 13, 2012.
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
Provisions Relating to IETF Documents
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(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Requirements Language . . . . . . . . . . . . . . . . . . 5
2.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
3. RTP Multiplex Points . . . . . . . . . . . . . . . . . . . . . 6
3.1. Session . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.2. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.3. CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.4. Payload Type . . . . . . . . . . . . . . . . . . . . . . . 9
4. Multiple Streams Alternatives . . . . . . . . . . . . . . . . 10
5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 11
5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 12
5.1.1. RTCP Reporting . . . . . . . . . . . . . . . . . . . . 12
5.1.2. Compound RTCP Packets . . . . . . . . . . . . . . . . 13
5.2. Point to Multipoint Using Multicast . . . . . . . . . . . 13
5.3. Point to Multipoint Using an RTP Translator . . . . . . . 15
5.4. Point to Multipoint Using an RTP Mixer . . . . . . . . . . 16
5.5. Point to Multipoint using Multiple Unicast flows . . . . . 17
5.6. De-composite End-Point . . . . . . . . . . . . . . . . . . 18
6. Multiple Streams Discussion . . . . . . . . . . . . . . . . . 19
6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 19
6.2. RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 19
6.2.1. The RTP Specification . . . . . . . . . . . . . . . . 19
6.2.2. Handling Varying sets of Senders . . . . . . . . . . . 22
6.2.3. Cross Session RTCP Requests . . . . . . . . . . . . . 22
6.2.4. Binding Related Sources . . . . . . . . . . . . . . . 22
6.2.5. Forward Error Correction . . . . . . . . . . . . . . . 24
6.2.6. Transport Translator Sessions . . . . . . . . . . . . 25
6.3. Interworking . . . . . . . . . . . . . . . . . . . . . . . 25
6.3.1. Interworking Applications . . . . . . . . . . . . . . 26
6.3.2. Multiple SSRC Legacy Considerations . . . . . . . . . 27
6.4. Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28
6.4.1. Session Oriented Properties . . . . . . . . . . . . . 28
6.4.2. SDP Prevents Multiple Media Types . . . . . . . . . . 29
6.4.3. Media Stream Usage . . . . . . . . . . . . . . . . . . 29
6.5. Network Aspects . . . . . . . . . . . . . . . . . . . . . 30
6.5.1. Quality of Service . . . . . . . . . . . . . . . . . . 30
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6.5.2. NAT and Firewall Traversal . . . . . . . . . . . . . . 31
6.5.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 32
6.5.4. Multiplexing multiple RTP Session on a Single
Transport . . . . . . . . . . . . . . . . . . . . . . 33
6.6. Security Aspects . . . . . . . . . . . . . . . . . . . . . 33
6.6.1. Security Context Scope . . . . . . . . . . . . . . . . 33
6.6.2. Key-Management for Multi-party session . . . . . . . . 34
6.6.3. Complexity Implications . . . . . . . . . . . . . . . 34
6.7. Multiple Media Types in one RTP session . . . . . . . . . 35
7. Arch-Types . . . . . . . . . . . . . . . . . . . . . . . . . . 37
7.1. Single SSRC per Session . . . . . . . . . . . . . . . . . 37
7.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 39
7.3. Multiple Sessions for one Media type . . . . . . . . . . . 40
7.4. Multiple Media Types in one Session . . . . . . . . . . . 41
7.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 43
8. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 43
9. Proposal for Future Work . . . . . . . . . . . . . . . . . . . 44
10. RTP Specification Clarifications . . . . . . . . . . . . . . . 45
10.1. RTCP Reporting from all SSRCs . . . . . . . . . . . . . . 45
10.2. RTCP Self-reporting . . . . . . . . . . . . . . . . . . . 45
10.3. Combined RTCP Packets . . . . . . . . . . . . . . . . . . 45
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 46
12. Security Considerations . . . . . . . . . . . . . . . . . . . 46
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 46
14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 46
14.1. Normative References . . . . . . . . . . . . . . . . . . . 46
14.2. Informative References . . . . . . . . . . . . . . . . . . 46
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 49
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 51
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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 choice of 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 translation in addition to
forwarding, or an RTP mixer that creates new conceptual sources from
the received streams. In addition, multiple streams may occur when a
single end-point have multiple media sources, 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
behavior and the implications of a particular behavior depend on
which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behavior
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. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
2.2. Terminology
The following terms and abbreviations are used in this document:
End-point: 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
end-point.
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.
Media Source: The originator or source of a particular Media Stream.
It can either be a single media capturing device such as a video
camera, a microphone, or a specific output of a media production
function, such as an audio mixer, or some video editing function.
Media Aggregate: All Media Streams related to a particular Source.
Media 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.
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RTP Session: As defined by [RFC3550], the end-points belonging to
the same RTP Session are those that share a single SSRC space.
That is, those end-points can see an SSRC identifier transmitted
by any one of the other end-points. An end-point can receive an
SSRC either as SSRC or as CSRC in RTP and RTCP packets. Thus, the
RTP Session scope is decided by the end-points' network
interconnection topology, in combination with RTP and RTCP
forwarding strategies deployed by end-points and any
interconnecting middle nodes.
Source: See Media Source.
3. RTP Multiplex Points
This section describes the existing RTP tools that enable
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.
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).
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.
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Uniqueness: Globally unique within the general communication
context for the specific end-point.
Inter-relation: Depending on the underlying transport and
signalling protocol.
Special Restrictions: None.
A source that changes its source transport address during a session
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[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 end-points and any interconnecting middle nodes.
3.2. SSRC
An RTP Session serves one or more Media Sources, each sending a Media
Stream.
Identifier: Synchronization 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.
Usage: Identify individual Media Sources within an RTP Session.
Refer to individual Media Sources in RTCP messages and SDP
signalling.
Uniqueness: Randomly chosen, globally unique within an RTP
Session and not dependent on network address.
Inter-relation: SSRC belonging to the same synchronization
context (originating from the same end-point), within or
between RTP Sessions, are indicated through use of identical
SDES CNAME items in RTCP compound packets with those SSRC as
originating source. SDP signalling can provide explicit SSRC
grouping [RFC5576]. When CNAME is inappropriate or
insufficient, there exist a few other methods to relate
different SSRC. One such case is session-based RTP
retransmission [RFC4588]. In some cases, the same SSRC
Identifier value is used to relate streams in two different RTP
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Sessions, such as in Multi-Session Transmission of scalable
video [RFC6190].
Special Restrictions: All RTP implementations must be prepared to
use procedures for SSRC collision handling, which results in an
SSRC number change. A Media Source that changes its RTP Session
identifier (e.g. source transport address) during a session must
also choose a new SSRC identifier to avoid being interpreted as
looped source. Note that RTP sequence number and RTP timestamp
are scoped by SSRC and thus independent between different SSRCs.
A media source having an SSRC identifier can be of different types:
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.
Virtual: A source that does not generate any RTP media stream in
itself (e.g. an end-point only receiving in an RTP session), but
anyway need a sender SSRC for use as source in RTCP reports.
Note that a "multimedia source" 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. 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 end-point. The main property attributed to SSRCs associated
with the same CNAME is that they are from a particular
synchronization context and may be synchronized 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.
Some RTP extension mechanisms already require the RTP stacks to
handle additional SSRCs, like SSRC multiplexed RTP retransmission
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[RFC4588]. However, that still only requires handling a single media
decoding chain per pair of SSRCs.
3.3. CSRC
The Contributing Source (CSRC) can arguably be seen as a sub-part of
a specific SSRC and thus a multiplexing point. It is optionally
included in the RTP header, 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. 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 can be represented in various encoding formats.
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 signaling means. In SDP, the term
"format" includes media type, RTP timestamp sampling rate,
codec, codec configuration, payload format configurations, and
various robustness mechanisms such as redundant encodings
[RFC2198].
Uniqueness: Scoped by sending end-point within an RTP Session.
To avoid any potential for ambiguity, it is desirable that
payload types are unique across all sending end-points within
an RTP session, but this is often not true in practice. All
SSRC in an RTP session sent from an single end-point 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
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[RFC3389].
Inter-relation: There are some uses where Payload Type numbers
need be unique across RTP Sessions. This is for example the
case in Media Decoding Dependency [RFC5583] where Payload Types
are used to describe media dependency across RTP Sessions.
Another example is session-based RTP retransmission [RFC4588].
Special Restrictions: Using different RTP timestamp clock rates for
the RTP Payload Types in use in the same RTP Session have issues
such as loss of synchronization. Payload Type clock rate
switching requires some special consideration that is described in
the multiple clock rates specification
[I-D.ietf-avtext-multiple-clock-rates].
If there is a true need to send multiple Payload Types for the same
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.
Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto].
4. Multiple Streams Alternatives
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
Payload Type Multiplexing: Using different RTP payload types for
different additional streams.
Independent of the reason to use additional media streams, achieving
it using payload type multiplexing is not a good choice as can be
seen in the Appendix A. The RTP payload type alone is not suitable
for cases where additional media streams are required. Streams need
their own SSRCs, so that they get their own sequence number space.
The SSRC itself is also important so that the media stream can be
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referenced and reported on.
This leaves us with two main choices, either using SSRC multiplexing
to have multiple SSRCs from one end-point in one RTP session, or
create an additional RTP session to hold that additional SSRC. As
the below discussion will show, in reality we cannot choose a single
one of the two solutions. To utilize RTP well and as 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.
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 Retransmission streams
o FEC stream
o Alternative Encodings
o Scalability layers
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.
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 behaviors 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 utilize the facilities of RTP are
ignored in this document.
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5.1. Point to Point
This is the most basic use case with an RTP session containing two
end-points. Each end-point has one or more SSRCs.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
5.1.1. RTCP Reporting
In cases when an end-point uses multiple SSRCs, we have found two
closely related issues. The first is if every SSRC shall report on
all other SSRC, even the ones originating from the same end-point.
The reason for this would be to ensure that no monitoring function
should suspect a breakage in the RTP session.
The second issue around RTCP reporting arise when an end-point
receives one or more media streams, and when the receiving end-point
itself sends multiple SSRC in the same RTP session. As transport
statistics are gathered per end-point and shared between the nodes,
all the end-point's SSRC will report based on the same received data,
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 end-point 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 behavior 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 end-point, any solution needs to take into account
what a particular end-point 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 end-point to a Transport
Translator, and a second scenario with two single SSRC remote end-
points sending to the same Transport Translator. The main
differences between those two scenarios are that in the second
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scenario, the RTT may vary between the SSRCs (but it is not
guaranteed), and the SSRCs may also have different CNAMEs.
5.1.2. Compound RTCP Packets
When an end-point has multiple SSRCs and it needs to send RTCP
packets on behalf of these SSRCs, the question arises if and how RTCP
packets with different source SSRCs can be sent in the same compound
packet. If it is allowed, then some consideration of the
transmission scheduling is needed.
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 2: 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 3: Point to Multipoint using Source Specific Multicast
In this topology a number of Media Senders (1 to M) are allowed to
send media to the SSM group, sends media to the distribution source
which then forwards the media streams to the multicast group. The
media streams reach the Receivers (R(1) to R(n)). The Receiver's
RTCP cannot be sent to the multicast group. To support RTCP, an RTP
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 is sent in its own multicast group
to allow for interworking with legacy or to support different levels
of protection.
The result of this is three common behaviors for RTP multicast:
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1. Use of multiple RTP sessions for the same media type.
2. The need for identifying RTP sessions that are related in one of
several possible ways.
3. The need for binding related SSRCs in different RTP sessions
together.
This indicates that Multicast is an important consideration when
working with the RTP multiplexing and multi stream architecture
questions. It is also important to note that so far there is no
special mode for basic behavior between multicast and unicast usages
of RTP. Yes, there are extensions targeted to deal with multicast
specific cases, but the general applicability does need to be
considered.
5.3. Point to Multipoint Using an RTP Translator
Transport Translators (Relays) are a very important consideration for
this document as they 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 4: Transport Translator (Relay)
One of the most important aspects with the simple relay is that it is
both easy to implement and require minimal amount of resources as
only transport headers are rewritten, no RTP modifications nor media
transcoding occur. Thus it is most likely the cheapest and most
generally deployable method for multi-point sessions. The most
obvious downside of this basic relaying is that the translator has no
control over how many streams needs to be delivered to a receiver.
Nor can it simply select to deliver only certain streams, as it
creates session inconsistencies. If some middlebox temporarily stops
a stream, this prevents some receivers from reporting on it. From
the senders 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
need it send any RTCP reports on the flows that pass it, but it may
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choose to do that.
Use of a transport translator results in that any of the end-points
will receive multiple SSRCs over a single unicast transport flow from
the translator. That is independent of the other end-points having
only a single or several SSRCs. End-points that have multiple SSRCs
put further requirements on how SSRCs can be related or bound within
and across RTP sessions and how they can be identified on an
application level. The transport translator has a signalling
requirement that also exist in any source multicast; all of the
participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It should
be noted that SDP offer/answer [RFC3264] has issues with ensuring
this property.
A Media Translator can perform a large variety of media functions
affecting the media stream passing the translator, coming from one
source and destined to a particular end-point. The translator can
transcode to a different bit-rate, transcode to use another encoder,
change the packetization of the media stream, add FEC streams, or
terminate RTP retransmissions. The latter behaviors require the
translator to use SSRCs that only exist in a particular sub-domain of
the RTP session, and it may also create additional sessions when the
mechanism applied on one side so requires.
5.4. Point to Multipoint Using an RTP Mixer
The most commonly used topology in centralized conferencing is based
on the RTP Mixer. The main reason for this is that it provides a
very consistent view of the RTP session towards each participant.
That is accomplished through the mixer having its own SSRCs and any
media sent to the participants will be sent using those SSRCs. If
the mixer wants to identify the underlying media sources for its
conceptual streams, it can identify them using CSRC. The media
stream the mixer provides can be an actual media mixing of multiple
media sources. It might also be as simple as selecting one of the
underlying sources based on some mixer policy or control signalling.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 5: RTP Mixer
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In the case where the mixer does stream selection, an application may
in fact desire multiple simultaneous streams but only as many as the
mixer can handle. As long as the mixer and an end-point can agree on
the maximum number of streams and how the streams that are delivered
are selected, this provides very good functionality. As these
streams are forwarded using the mixer's SSRCs, there are no
inconsistencies within the session.
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
end-point 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.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 6: 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 end-point may use a single local port to receive all
these transport flows, or it might have separate local reception
ports for each of the end-points.
There exists an alternative structure for establishing the above
communication scenario (Figure 6) which uses independent RTP sessions
between each pair of peers, i.e. three different RTP sessions.
Unless independently adapted the same RTP media stream could be sent
in both of the RTP sessions an end-point has. The difference exists
in the behaviors around RTCP, for example common RTCP bandwidth for
one joint session, rather than three independent pools, and the
awareness based on RTCP reports between the peers of how that third
leg is doing.
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5.6. De-composite End-Point
There is some possibility that an RTP end-point implementation in
fact reside on multiple devices, each with their own network address.
A very basic use case for this would be to separate audio and video
processing for a particular end-point, 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 end-point.
+---------------------+
| End-point A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+----\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+--------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+----/
| +------------+ |
+---------------------+
Figure 7: 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 synchronization and
clock drift handling works despite the decomposition.
However, if the audio and video were in a single RTP session then
this use case becomes problematic. This as all transport flow
receivers will need to receive all the other media streams that are
part of the session. Thus the audio component will receive also all
the video media streams, while the video component will receive all
the audio ones, doubling the site's bandwidth requirements from all
other session participants. With a joint RTP session it also becomes
evident that a given end-point, as interpreted from a CNAME
perspective, has two sets of transport flows for receiving the
streams and the decomposition is not hidden.
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 end-points based on
addresses and ports. A conclusion can also be reached that
decomposition without using separate RTP sessions has downsides and
potential for RTP/RTCP issues.
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There exist another use case which might be considered as a de-
composite end-point. However, as will be shown this should be
considered a translator instead. An example of this is when an end-
point A sends a media flow to B. On the path there is a device C that
on A's behalf does something with the media streams, for example adds
an RTP session with FEC information for A's media streams. C will in
this case need to bind the new FEC streams to A's media stream by
using the same CNAME as A.
+------+ +------+ +------+
| | | | | |
| A |------->| C |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
Figure 8: When De-composition is a Translator
This type of functionality where C does something with the media
stream on behalf of A is clearly covered under the media translator
definition (Section 5.3).
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.
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 minimized, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
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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
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."
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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. This is to
simplify the processing at any receiver of the media stream. If all
media streams that exist in an RTP session are of one media type and
one particular purpose, there is no need for deeper inspection of the
packets before processing them in both end-points and RTP aware
middle nodes.
The fifth argument discusses network aspects that we will discuss
more below in Section 6.5. It also goes into aspects of
implementation, like decomposed end-points 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 secondly use
different RTP sessions for media streams that don't share media type
and purpose, to maximize flexibility when it comes to processing and
handling of the media streams.
This mostly agrees with the discussion and recommendations in this
document. However, there has been an evolution of RTP since that
text was written which needs to be reflected in the discussion.
Additional clarifications for specific cases are also needed.
6.2.1.1. Different Media Types Recommendations
The above quote from RTP [RFC3550] includes a strong recommendation:
"For example, in a teleconference composed of audio and video
media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address."
It has been 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 document has a more detailed analysis of
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potential issues in having multiple media types in the same RTP
session in Section 6.7. An important influence for underlying
thinking for the RTP design and likely this statement can be found in
the academic paper by David Clark and David Tennenhouse
"Architectural considerations for a new generation of protocols"
[ALF].
6.2.2. Handling Varying sets of Senders
A potential issue that some application designers may need to
consider is the case where the set of simultaneously active sources
varies within a larger set of session members. As each media
decoding chain may contain state, it is important that this type of
usage ensures that a receiver can flush a decoding state for an
inactive source and if that source becomes active again, it does not
assume that this previous state exists.
This behavior will cause similar issues independent of SSRC or
Session multiplexing. It might be possible in certain applications
to limit the changes to a subset of communication session
participants by have the sub-set use particular RTP Sessions.
6.2.3. Cross Session RTCP Requests
There currently exists no functionality to make truly synchronized
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
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.
6.2.4. 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, and any solution and
recommendation related to the problem should work equally well with
both methods to avoid creating barriers between using session
multiplexing and SSRC multiplexing.
The current solutions do not have these properties. There exists one
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solution for grouping RTP session 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 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 end-point 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. In addition, being forced
to change the SSRC affects all the related media streams; instead of
having to re-synchronize only the originally conflicting stream, all
streams will suddenly need to be re-synchronized with each other.
This will prevent also the media streams not having an actual
collision from being usable during the re-synchronization and also
increases the time until synchronization is finalized. 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 behavior that needs to be
well defined by anyone using this. But, having an exception behavior
where the SSRC space is common across all session is an issue as this
behavior 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 behavior.
Existing mechanisms with known issues:
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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 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 end-point 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 more explicit mechanism. The
SDP based ssrc-group mechanism is sufficient in this case as long
as the application can rely on the signalling based solution.
Scalable Video Coding (RFC6190): As an example of scalable coding,
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.
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 harmonize and avoid future
shortcomings in extension specifications. A proposed solution for
handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].
6.2.5. 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
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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 provides good support for legacy
when deploying FEC or changing the scheme used, in the sense that it
supports the case where some set of receivers may not be able to
utilize the FEC information. By placing it in a separate RTP
session, it can easily be ignored.
In usages involving multicast, having the FEC information on its own
multicast group and 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 allow for easy
separation in the Burst/Retransmission Source (BRS) without having to
individually classify packets.
6.2.6. 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
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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. Interworking Applications
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. From an RTP perspective this could be problem free if
all the applications have made the same multiplexing choices, have
the same capabilities in number of simultaneous media streams
combined with the same set of RTP/RTCP extensions being supported.
Unfortunately this may not always be true.
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. If one's goal is to make minimal amount of work in such
a gateway, there are some multiplexing choices that one should avoid.
The lowest amount of work represents solutions where one can take an
SSRC from one RTP session in one application and forward it into
another RTP session. For example if one has one application that has
multiple SSRCs for one media type in one session and another
application that instead has chosen to use multiple RTP sessions with
only a single SSRC per end-point in each of these sessions. Then
mapping an SSRC from the side with one session into an RTP session is
possible. However mapping SSRC from different RTP sessions into a
single RTP session has the potential of creating SSRC collisions,
especially if an end-point has not generated independent random SSRC
values in each RTP session. This issue is even more likely in a case
where one side uses a single RTP session with multiple media types
and the other uses different RTP session for different media or
robustness mechanism such as retransmission [RFC4588]. Then it is
more likely or maybe even required to use the same SSRC in the
different RTP sessions.
In cases where the used structure is incompatible, the gateway will
need to make SSRC translation. Thus this incurs overhead and some
potential loss of functionality. First of all, if one translates the
SSRC in an RTP header then one will be forced to decrypt and re-
encrypt if one uses SRTP and thus also needs to be part of the
security association. Secondly, changing the SSRC also means that
one needs to translate all RTCP messages. This can be more complex,
but important so that the gateway does not end up having to terminate
the end-to-end RTCP chain. In that case the gateway will need to be
able to take the role of a true end-point in each session, which may
include functions such as bit-rate adaptation and correctly respond
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to whatever RTCP extensions are being used, and then translate them
or locally respond to them. Thirdly, an SSRC translation may require
that one changes RTP payloads; for example, an RTP retransmission
packet contains an original sequence number that must match the
sequence number used in for the corresponding packet with the new
SSRC. And for FEC packets this is even worse, as the original SSRC
is included as part of the data for which FEC redundant data is
calculated. A fourth issue is the potential for these gateways to
block evolution of the applications by blocking unknown RTP and RTCP
extensions that the regular application has been extended with.
If one uses security functions, like SRTP, they can as seen above
incur both additional risk due to the gateway needing to be in
security association between the end-points, 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 makes it hard to move a flow from one
RTP session to another as each RTP session will have one or more
different master keys and these must not be the same in multiple RTP
sessions as that can result in two-time pads that completely breaks
the confidentiality of the packets.
An additional issue around interworking is that for multi-party
applications it can be impossible to judge which different RTP
multiplexing behaviors that will be used by end-points that attempt
to join a session. Thus if one attempts to use a multiplexing choice
that has poor interworking, one may have to switch at a later stage
when someone wants to participate in a multi-party session using an
RTP application supporting only another behavior. It is likely
difficult to implement the switch without some media disruption.
To summarize, certain types of applications are likely to be inter-
worked. Sets of applications of similar type should strive to use
the same multiplexing structure to avoid the need to make an RTP
session level gateway. This as it incurs complexity costs, can force
the gateway to be part of security associations, force SSRC
translation and even payload translation which is also a potential
hinder to application evolution.
6.3.2. Multiple SSRC Legacy Considerations
Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per end-point and media type (typically audio
and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each end-point and the
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RTP mixer.
When establishing RTP sessions that may contain end-points that
aren't updated to handle multiple streams following these
recommendations, a particular application can have issues with
multiple SSRCs within a single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
RTP Session multiplexing could potentially avoid these issues if
there is only a single SSRC at each end-point, and in topologies
which appears like point to point as seen the end-point. However,
forcing the usage of session multiplexing due to this reason would be
a great mistake, as it is likely that a significant set of
applications will need a combination of SSRC multiplexing of several
media sources and session multiplexing for other aspects such as
encoding alternatives, adding robustness or simply to support legacy.
However, this issue does need consideration when deploying multiple
media streams within an RTP session where legacy end-points may
occur.
6.4. Signalling Aspects
There exist various signalling solutions for establishing RTP
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.
6.4.1. 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:
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o Bitrate/Bandwidth exist today only at aggregate or a common any
media stream limit
o Which SSRC that will use which RTP Payload Types
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.
This does in fact create a strong driver to use RTP session
multiplexing for any case where different sets of media streams with
different requirements exist.
6.4.2. SDP Prevents Multiple Media Types
SDP encoded in its structure prevention against using multiple media
types in the same RTP session. A media description in SDP can only
have a single 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. Thus a high fence against using multiple media
types in the same session was created.
There is an accepted WG item in the MMUSIC WG to define how multiple
media lines describe a single underlying transport
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus it becomes
possible in SDP to define one RTP session with multiple media types.
6.4.3. Media Stream Usage
Media streams being transported in RTP has some particular usage in
an RTP application. This usage of the media stream is in many
applications so far implicitly signalled. For example by having all
audio media streams arriving in the only audio RTP session they are
to be decoded, mixed and played out. However, in more advanced
applications that use multiple media streams there will be more than
a single usage or purpose among the set of media streams being sent
or received. RTP applications will need to signal this usage
somehow. Here the choice of SSRC multiplexing versus session
multiplexing will have significant impact. If one uses SSRC
multiplexing to its full extent one will have to explicitly indicate
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for each SSRC what its' usage and purpose are using some signalling
between the application instances.
This SSRC usage signalling will have some impact on the application
and also on any central RTP nodes. It is important in the design to
consider the implications of the need for additional signalling
between the nodes. One consideration is if a receiver can utilize
the media stream at all before it has received the signalling message
describing the media stream and its usage. Another consideration is
that any RTP central node, like an RTP mixer or translator that
selects, mixes or processes streams, in most cases will need to
receive the same signalling to know how to treat media streams with
different usage in the right fashion.
Application designers should consider putting media streams of the
same usage and/or receiving the same treatment in middleboxes in the
same RTP sessions and use the RTP session as an explicit indication
of how to deal with media streams. By having session level
indication of usage and have different RTP sessions for different
usages, the need for stream specific signalling can be reduced.
Especially signalling of the type that is time critical and needs to
be provided prior to the media stream being available.
6.5. Network Aspects
The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor.
6.5.1. 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 different version can be in a
different RTP session that can be sent over different 5-tuples.
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6.5.2. 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 end-points 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 end-point 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 end-point may use. In practice, implementation will
probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks.
NAT State: If an end-point 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
internal end-points, available external ports are typically the
scarce resource. Port limitations is primarily a problem for
larger centralized NATs where end-point 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 can take 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
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immediate triggered check back. This as ICE first finds one
candidate pair that works prior 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.
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 additional media streams within one RTP
Session and does not introduce any additional NAT traversal
complexities per media stream. In contrast, the session multiplexing
is using one RTP session per media stream. Thus additional lower
layer transport flows will be required, unless an explicit de-
multiplexing layer is added between 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-single-transport-multiplexing].
6.5.3. Multicast
Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like
behaviors with one end-point 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
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RTP session per multicast group is used.
If instead a single RTP session over multiple transports were to be
deployed, i.e. multicast groups with each layer as it's own SSRC,
then very different views of the RTP session would exist. That as
one receiver may see only a single layer (SSRC), while another may
see three SSRCs if it joined three multicast groups. This would
cause disjoint RTCP reports where a management system would not be
able to determine if a receiver isn't reporting on a particular SSRC
due to that it is not a member of that multicast group, or because it
doesn't receive it as a result of a transport failure.
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.
6.5.4. 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-single-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.
6.6. Security Aspects
On the basic level there is no significant difference in security
when having one RTP session and having multiple. However, there are
a few more detailed considerations that might need to be considered
in certain usages.
6.6.1. 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.
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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.
In the latter case it is likely easiest from signalling, transport
(if done over multicast) and security to use a different RTP session.
That way the user(s) not intended to receive a particular stream can
easily be excluded. There is no need to have SSRC specific keys,
which many of the key-management systems cannot handle.
6.6.2. Key-Management for Multi-party session
Performing key-management for Multi-party session can be a challenge.
This section considers some of the issues.
Transport translator based session cannot use Security Description
[RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end-
point provides its set of keys. In centralized conference, the
signalling counterpart is a conference server and the media plane
unicast counterpart (to which DTLS messages would be sent) is the
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.
Keying of multicast transported SRTP face similar challenges as the
transport translator case.
6.6.3. Complexity Implications
The usage of security functions can surface complexity implications
of the choice of multiplexing and topology. This becomes especially
evident in RTP topologies having any type of middlebox that processes
or modifies RTP/RTCP packets. Where there is very small overhead for
a not secured RTP translator or mixer to rewrite an SSRC value in the
RTP packet, the cost of doing it when using cryptographic security
functions is higher. For example if using SRTP [RFC3711], the actual
security context and exact crypto key are determined by the SSRC
field value. If one changes it, the encryption and authentication
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tag must be performed using another key. Thus changing the SSRC
value implies a decryption using the old SSRC and its security
context followed by an encryption using the new one.
There exist many valid cases where a middlebox will be forced to
perform such cryptographic operations due to the intended purpose of
the middlebox, for example a media transcoding RTP translator cannot
avoid performing these operations as they will produce a different
payload compared to the input. However, there exist some cases where
another topology and/or multiplexing choice could avoid the
complexities.
6.7. Multiple Media Types in one RTP session
Having different media types, like audio and video, in the same RTP
sessions is not forbidden, only recommended against as earlier
discussed in Section 6.2.1.1. When using multiple media types, there
are a number of considerations:
Payload Type gives Media Type: This solution is dependent on getting
the media type from the Payload Type. Thus overloading this de-
multiplexing point in a receiver making it serve two purposes.
First to provide the main media type and determining the
processing chain, then later for the exact configuration of the
encoder and packetization.
Payload Type field limitations: The total number of Payload Types
available to use in an RTP session is fairly limited, especially
if Multiplexing RTP Data and Control Packets on a Single Port
[RFC5761] is used. For certain applications negotiating a large
set of codes and configuration this may become an issue.
An SSRC cannot use two clock rates simultaneously: The used RTP
clock rate for an SSRC is determined from the payload type. As
discussed in Appendix A it is not possible to simultaneously use
two different clock rates for the same SSRC. Even switching clock
rate once has potential issues if packet loss occurs at the same
time. Different media types commonly have different clock rates
preventing or creating issues to use two different media types for
the same SSRC.
Do not switch media types for an SSRC: The primary reasons to avoid
switching from sending for example audio to sending video using
the same SSRC is the implications on a receiver. When this
happens, the processing chain in the receiver will have to switch
from one media type to another. As the different media type's
entire processing chains are different and are connected to
different outputs it is difficult to reuse the decoding chain,
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which a normal codec change likely can. Instead the entire
processing chain has to be torn down and replaced. In addition,
there is likely a clock rate switching problem, possibly resulting
in synchronization loss at the point of switching media type if
some packet loss occurs. So this is a behavior that shall be
avoided.
RTCP Bit-rate Issues: If the media types are significantly different
in bit-rate, the RTCP bandwidth rates assigned to each source in a
session can result in interesting effects, like that the RTCP bit-
rate share for an audio stream is larger than the actual audio
bit-rate. In itself this doesn't cause any conflicts, only
potentially unnecessary overhead. It is possible to avoid this
using AVPF or SAVPF and setting trr-int parameter, which can bring
down unnecessary regular reporting while still allowing for rapid
feedback.
De-composite end-points: De-composite nodes that rely on the regular
network to separate audio and video to different devices do not
work well with this session setup. If they are forced to work,
all media receiver parts of a de-composite end-point will receive
all media, thus doubling the bit-rate consumption for the end-
point.
Flow based QoS Separation: Flow based QoS mechanisms will see all
the media streams in the RTP session as part of a single flow.
Therefore there is no possibility to provide separated QoS
behavior for the different media types or flows.
RTP Mixers and Translators: An RTP mixer or Media Translator will
also have to support this particular session setup, where it
before could rely on the RTP session to determine what processing
options should be applied to the incoming packets.
Legacy Implementations: The use of multiple media types has the
potential for even larger issues with legacy implementations than
single media type SSRC multiplexing due to the occurrence of
multiple media types among the payload type configurations.
As can be seen, there is nothing in here that prevents using a single
RTP session for multiple media types, however it does create a number
of limitations and special case implementation requirements. So
anyone considering using this setup should carefully review if the
reasons for using a single RTP session are sufficient to motivate the
needed special handling.
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7. Arch-Types
This section discusses some arch-types of how RTP multiplexing can be
used in applications to achieve certain goals and a summary of their
implications. For each arch-type there is discussion of benefits and
downsides.
7.1. Single SSRC per Session
In this arch-type each end-point in a point-to-point session has only
a single SSRC, thus the RTP session contains only two SSRCs, one
local and one remote. This session can be used both unidirectional,
i.e. only a single media stream or bi-directional, i.e. both end-
points have one media stream each. If the application needs
additional media flows between the end-points, they will have to
establish additional RTP sessions.
The Pros:
1. This arch-type has great legacy interoperability potential as it
will not tax any RTP stack implementations.
2. The signalling has good possibilities to negotiate and describe
the exact formats and bit-rates for each media stream, especially
using today's tools in SDP.
3. It does not matter if usage or purpose of the media stream is
signalled on media stream level or session level as there is no
difference.
4. It is possible to control security association per RTP session
with current key-management.
The Cons:
a. The number of required RTP sessions cannot really be higher,
which has the implications:
* Linear growth of the amount of NAT/FW state with number of
media streams.
* Increased delay and resource consumption from NAT/FW
traversal.
* Likely larger signalling message and signalling processing
requirement due to the amount of session related information.
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* Higher potential for a single media stream to fail during
transport between the end-points.
b. When the number of RTP sessions grows, the amount of explicit
state for relating media stream also grows, linearly or possibly
exponentially, depending on how the application needs to relate
media streams.
c. The port consumption may become a problem for centralized
services, where the central node's port consumption grows rapidly
with the number of sessions.
d. For applications where the media streams are highly dynamic in
their usage, i.e. entering and leaving, the amount of signalling
can grow high. Issues arising from the timely establishment of
additional RTP sessions can also arise.
e. Cross session RTCP requests needs is likely to exist and may
cause issues.
f. If the same SSRC value is reused in multiple RTP sessions rather
than being randomly chosen, interworking with applications that
uses another multiplexing structure than this application will
have issues and require SSRC translation.
g. Cannot be used with Any Source Multicast (ASM) as one cannot
guarantee that only two end-points participate as packet senders.
Using SSM, it is possible to restrict to these requirements if no
RTCP feedback is used.
h. For most security mechanisms, each RTP session or transport flow
requires individual key-management and security association
establishment thus increasing the overhead.
i. Does not support multiparty session within a session. Instead
each multi-party participant will require an individual RTP
session to a given end-point, even if a central node is used.
RTP applications that need to inter-work with legacy RTP
applications, like VoIP and video conferencing, can potentially
benefit from this structure. However, a large number of media
descriptions in SDP can also run into issues with existing
implementations. For any application needing a larger number of
media flows, the overhead can become very significant. This
structure is also not suitable for multi-party sessions, as any given
media stream from each participant, although having same usage in the
application, must have its own RTP session. In addition, the dynamic
behavior that can arise in multi-party applications can tax the
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signalling system and make timely media establishment more difficult.
7.2. Multiple SSRCs of the Same Media Type
In this arch-type, each RTP session serves only a single media type.
The RTP session can contain multiple media streams, either from a
single end-point or due to multiple end-points. This commonly
creates a low number of RTP sessions, typically only two one for
audio and one for video with a corresponding need for two listening
ports when using RTP and RTCP multiplexing.
The Pros:
1. Low number of RTP sessions needed compared to single SSRC case.
This implies:
* Reduced NAT/FW state
* Lower NAT/FW Traversal Cost in both processing and delay.
2. Allows for early de-multiplexing in the processing chain in RTP
applications where all media streams of the same type have the
same usage in the application.
3. Works well with media type de-composite end-points.
4. Enables Flow-based QoS with different prioritization between
media types.
5. For applications with dynamic usage of media streams, i.e. they
come and go frequently, having much of the state associated with
the RTP session rather than an individual SSRC can avoid the need
for in-session signalling of meta-information about each SSRC.
6. Low overhead for security association establishment.
The Cons:
a. May have some need for cross session RTCP requests for things
that affect both media types in an asynchronous way.
b. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per end-point.
c. Will not be able to control security association for sets of
media streams within the same media type with today's key-
management mechanisms, only between SDP media descriptions.
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For RTP applications where all media streams of the same media type
share same usage, this structure provides efficiency gains in amount
of network state used and provides more faith sharing with other
media flows of the same type. At the same time, it is still
maintaining almost all functionalities when it comes to negotiation
in the signalling of the properties for the individual media type and
also enabling flow based QoS prioritization between media types. It
handles multi-party session well, independently of multicast or
centralized transport distribution, as additional sources can
dynamically enter and leave the session.
7.3. Multiple Sessions for one Media type
In this arch-type one goes one step further than in the above
(Section 7.2) by using multiple RTP sessions also for a single media
type. The main reason for going in this direction is that the RTP
application needs separation of the media streams due to their usage.
Some typical reasons for going to this arch-type are scalability over
multicast, simulcast, need for extended QoS prioritization of media
streams due to their usage in the application, or the need for fine
granular signalling using today's tools.
The Pros:
1. More suitable for Multicast usage where receivers can
individually select which RTP sessions they want to participate
in, assuming each RTP session has its own multicast group.
2. Detailed indication of the application's usage of the media
stream, where multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely signalling.
4. Enables detailed QoS prioritization for flow based mechanisms.
5. Works well with de-composite end-points.
6. Handles dynamic usage of media streams well.
7. For transport translator based multi-party sessions, this
structure allows for improved control of which type of media
streams an end-point receives.
8. The scope for who is included in a security association can be
structured around the different RTP sessions, thus enabling such
functionality with existing key-management.
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The Cons:
a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.
b. Increased amount of session configuration state.
c. May need synchronized cross-session RTCP requests and require
some consideration due to this.
d. For media streams that are part of scalability, simulcast or
transport robustness it will be needed to bind sources, which
must support multiple RTP sessions.
e. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per end-point.
f. Higher overhead for security association establishment.
g. If the applications need finer control than on media type level
over which session participants that are included in different
sets of security associations, most of today's key-management
will have difficulties establishing such a session.
For more complex RTP applications that have several different usages
for media streams of the same media type and / or uses scalability or
simulcast, this solution can enable those functions at the cost of
increased overhead associated with the additional sessions. This
type of structure is suitable for more advanced applications as well
as multicast based applications requiring differentiation to
different participants.
7.4. Multiple Media Types in one Session
This arch-type is to use a single RTP session for multiple different
media types, like audio and video, and possibly also transport
robustness mechanisms like FEC or Retransmission. Each media stream
will use its own SSRC and a given SSRC value from a particular end-
point will never use the SSRC for more than a single media type.
The Pros:
1. Single RTP session which implies:
* Minimal NAT/FW state.
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* Minimal NAT/FW Traversal Cost.
* Fate-sharing for all media flows.
2. Enables separation of the different media types based on the
payload types so media type specific end-point or central
processing can still be supported despite single session.
3. Can handle dynamic allocations of media streams well on an RTP
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
4. Minimal overhead for security association establishment.
The Cons:
a. Not suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a
single media type, due to high risk of forced SSRC translation.
b. Negotiation of bandwidth for the different media types is
currently not possible in SDP. This requires SDP extensions to
enable payload or source specific bandwidth. Likely to be a
problem due to media type asymmetry in required bandwidth.
c. Does enforce higher bandwidth and processing on de-composite end-
points.
d. Flow based QoS cannot provide separate treatment to some media
streams compared to other in the single RTP session.
e. If there is significant asymmetry between the media streams RTCP
reporting needs, there are some challenges in configuration and
usage to avoid wasting RTCP reporting on the media stream that
does not need that frequent reporting.
f. Not suitable for applications where some receivers like to
receive only a subset of the media streams, especially if
multicast or transport translator is being used.
g. Additional concern with legacy implementations that does not
support the RTP specification fully when it comes to handling
multiple SSRC per end-point, as also multiple simultaneous media
types needs to be handled.
h. If the applications need finer control over which session
participants that are included in different sets of security
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associations, most key-management will have difficulties
establishing such a session.
The analysis in this document and considerations in Section 6.7
implies that this is suitable only in a set of restricted use cases.
The aspect in the above list that can be most difficult to judge long
term is likely the potential need for interworking with other
applications and services.
7.5. Summary
There are some clear relations between these arch-types. Both the
"single SSRC per RTP session" and the "multiple media types in one
session" are cases which require full explicit signalling of the
media stream relations. However, they operate on two different
levels where the first primarily enables session level binding, and
the second needs to do it all on SSRC level. From another
perspective, the two solutions are the two extreme points when it
comes to number of RTP sessions required.
The two other arch-types "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for one Media Type" are examples of two other
cases that first of all allows for some implicit mapping of the role
or usage of the media streams based on which RTP session they appear
in. It thus potentially allows for less signalling and in particular
reduced need for real-time signalling in dynamic sessions. They also
represent points in between the first two when it comes to amount of
RTP sessions established, i.e. representing an attempt to reduce the
amount of sessions as much as possible without compromising the
functionality the session provides both on network level and on
signalling level.
8. Guidelines
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 6.2.4 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).
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Use SSRC multiplexing for additional Media Sources: In the cases an
RTP end-point needs to transmit additional media source(s) of the
same media type and purpose in the application, it is recommended
to send them as additional SSRCs in the same RTP session. For
example a tele-presence 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 purposes:
When media streams have different purpose or processing
requirements it is recommended that the different types of streams
are put in different RTP sessions.
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
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. Proposal 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
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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 end-point can handle within a given RTP Session. That
ensures that usage of SSRC multiplexing occurs when supported and
without overloading an end-point. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc].
10. RTP Specification Clarifications
This section describes a number of clarifications to the RTP
specifications that are likely necessary for aligned behavior when
RTP sessions contain more SSRCs than one local and one remote.
10.1. RTCP Reporting from all SSRCs
When one have multiple SSRC in an RTP node, all these SSRC must send
RTCP SR or RR 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. The reason for this is that a third party monitor may not
necessarily be able to determine that all these SSRC are in fact co-
located and originate from the same stack instance that gather report
data.
10.2. 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. A 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
same node.
10.3. 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
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send.
11. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
12. Security Considerations
There is discussion of the security implications of choosing SSRC vs
Session multiplexing in Section 6.6.
13. Acknowledgements
The authors would like to thanks Harald Alvestrand for providing
input into the discussion regarding multiple media types in a single
RTP session.
14. References
14.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
14.2. 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-00 (work in
progress), December 2011.
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[I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[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-avtext-multiple-clock-rates]
Petit-Huguenin, M., "Support for multiple clock rates in
an RTP session", draft-ietf-avtext-multiple-clock-rates-02
(work in progress), January 2012.
[I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-01 (work in progress),
July 2011.
[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 (work in progress),
October 2011.
[I-D.westerlund-avtcore-single-transport-multiplexing]
Westerlund, M., "Multiple RTP Session on a Single Lower-
Layer Transport",
draft-westerlund-avtcore-transport-multiplexing (work in
progress), October 2011.
[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 (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.
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[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.
[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.
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[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.
[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.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
Appendix A. Dismissing Payload Type Multiplexing
This section documents a number of reasons why using the payload type
as a multiplexing point for most things related to multiple streams
is unsuitable. If one attempts to use Payload type multiplexing
beyond it's defined usage, that has well known negative effects on
RTP. To use Payload type as the single discriminator for multiple
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streams implies that all the different media streams are being sent
with the same SSRC, thus using the same timestamp and sequence number
space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not
possible, Payload Type multiplexing cannot be used.
2. Many RTP payload formats may fragment a media object over
multiple packets, like parts of a video frame. These payload
formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all
fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in
sequence.
3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/
T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before
being used with Payload Type multiplexing.
4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which
stream a packet loss relates to.
5. If RTP Retransmission [RFC4588] is used and there is a loss, it
is possible to ask for the missing packet(s) by SSRC and
sequence number, not by Payload Type. If only some of the
Payload Type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets must be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported,
so sending feedback for a specific media stream is difficult
without extending existing RTCP reporting.
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7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing.
8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number
of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this
is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing.
11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.
12. A legacy end-point that doesn't understand the indication that
different RTP payload types are different media streams may be
slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types.
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
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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
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