One document matched: draft-westerlund-avtcore-multiplex-architecture-00.xml
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<rfc category="bcp"
docName="draft-westerlund-avtcore-multiplex-architecture-00"
ipr="trust200902">
<front>
<title abbrev="RTP Multiplexing Architecture">RTP Multiplexing
Architecture</title>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 13 11</phone>
<email>bo.burman@ericsson.com</email>
</address>
</author>
<author fullname="Colin Perkins" initials="C. " surname="Perkins">
<organization>University of Glasgow</organization>
<address>
<postal>
<street>School of Computing Science</street>
<city>Glasgow</city>
<code>G12 8QQ</code>
<country>United Kingdom</country>
</postal>
<email>csp@csperkins.org</email>
</address>
</author>
<date day="24" month="October" year="2011" />
<abstract>
<t>RTP has always been a protocol that supports multiple participants
each sending their own media streams in an RTP session. Thus relying on
the three main multiplexing points in RTP; RTP session, SSRC and Payload
Type for their various needs. However, most usages of RTP have been less
complex often with a single SSRC in each direction, with a single RTP
session per media type. But the more complex usages start to be more
common and thus guidance on how to use RTP in various complex cases are
needed. This document analyzes a number of cases and discusses the usage
of the various multiplexing points and the need for functionality when
defining RTP/RTCP extensions that utilize multiple RTP streams and
multiple RTP sessions.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>This document focuses at issues of non-basic usage of <xref
target="RFC3550">RTP</xref> where multiple media sources of the same
media type are sent over RTP. Separation of different media types is
another issue that will be discussed in this document. The intended uses
include for example multiple sources from the same end-point, multiple
streams from a single media source, multiple end-points each having a
source, or an application that needs multiple representations
(encodings) of a particular source. It will be shown that these uses are
inter-related and need a common discussion to ensure consistency. In
general, usage of the RTP session and media streams will be discussed in
detail.</t>
<t>RTP is already designed for multiple participants in a communication
session. This is not restricted to multicast, as many believe, but also
provides functionality over unicast, using either multiple transport
flows below RTP or a network node that re-distributes the RTP packets.
The node can for example be a transport translator (relay) that forwards
the packets unchanged, a translator performing media translation in
addition to forwarding, or an RTP mixer that creates new conceptual
sources from the received streams. In addition, multiple streams may
occur when a single end-point have multiple media sources of the same
media type, like multiple cameras or microphones that need to be sent
simultaneously.</t>
<t>Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per end-point and media type (typically audio and
video). Even in conferencing applications, especially voice only, the
conference focus or bridge has provided a single stream with a mix of
the other participants to each participant. It is also common to have
individual RTP sessions between each end-point and the RTP mixer.</t>
<t>SSRC is the RTP media stream identifier that helps to uniquely
identify media sources in RTP sessions. Even though available SSRC space
can theoretically handle more than 4 billion simultaneous sources, the
perceived need for handling multiple SSRCs in implementations has been
small. This has resulted in an installed legacy base that isn't fully
RTP specification compliant and will have different issues if they
receive multiple SSRCs of media, either simultaneously or in sequence.
These issues will manifest themselves in various ways, either by
software crashes or simply in limited functionality, like only decoding
and playing back the first or latest received SSRC and discarding media
related to any other SSRCs.</t>
<t>There have also arisen various cases where multiple SSRCs are used to
represent different aspects of what is in fact a single underlying media
source. A very basic case is <xref target="RFC4588">RTP
retransmission</xref> which have one SSRC for the original stream, and a
second SSRC either in the same session or in a different session to
represent the retransmitted packets to ensure that the monitoring
functions still function. Another use case is scalable encoding, such as
the <xref target="RFC6190">RTP payload format for Scalable Video Coding
(SVC)</xref>, which has an operation mode named Multiple Session
Transmission (MST) that uses one SSRC in each RTP session to send one or
more scalability layers. A third example is simulcast where a single
media source is encoded in different versions and transmitted to an RTP
mixer that picks which version to actually distribute to a given
receiver part of the RTP session.</t>
<t>This situation has created a need for a document that discusses the
existing possibilities in the RTP protocol and how these can and should
be used in applications. A new set of applications needing more advanced
functionalities from RTP is also emerging on the market, such as
telepresence and advanced video conferencing. Thus furthering the need
for a more common understanding of how multiple streams are handled in
RTP to ensure media plane interoperability.</t>
<t>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. Finally, some recommendations and examples are provided.</t>
</section>
<section title="Definitions">
<t></t>
<section title="Requirements Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
<section title="Terminology">
<t>The following terms and abbreviations are used in this
document:</t>
<t><list style="hanging">
<t hangText="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.</t>
<t hangText="Media Stream:">A sequence of RTP packets using a
single SSRC that together carry part or all of the content of a
specific Media Type from a specific sender source within a given
RTP session.</t>
<t hangText="Media Aggregate:">All Media Streams related to a
particular Source.</t>
<t hangText="Media Type:">Audio, video, text or data whose form
and meaning are defined by a specific real-time application.</t>
<t hangText="Source:">The source of a particular media stream.
Either a single media capturing device such as a video camera, or
a microphone, or a specific output of a media production function,
such as an audio mixer, or some video editing function.</t>
</list></t>
</section>
</section>
<section anchor="sec-mux-points" title="RTP Multiplex Points">
<t>This section describes the existing RTP tools that enable
multiplexing of different media streams and RTP functionalities.</t>
<section title="Session">
<t>The RTP Session is the highest semantic level in RTP and contains
all of the RTP functionality.</t>
<t>RTP in itself does not contain any Session identifier, but relies
on the underlying transport. 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). Most
commonly only the destination address, i.e. all traffic received on a
particular port, is defined as belonging to a specific RTP Session. It
is worth noting that in practice a more narrow definition of the
transport flows that are related to a give RTP session is possible. An
RTP session can for example be defined as one or more 5-tuples
(Transport Protocol, Source Address, Source Port, Destination Address,
Destination Port). Any set of identifiers of RTP and RTCP packet flows
are sufficient to determine if the flow belongs to a particular
session or not.</t>
<t>Commonly, RTP and RTCP use separate ports and the destination
transport address is in fact an address pair, but in the case of <xref
target="RFC5761">RTP/RTCP multiplex</xref> there is only a single
port.</t>
<t>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.</t>
<t>The set of participants considered part of the same RTP Session is
defined by<xref target="RFC3550"> </xref> 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.</t>
</section>
<section title="SSRC">
<t>The Synchronization Source (SSRC) identifier is used to identify
individual sources within an RTP Session. The SSRC number is globally
unique within an RTP Session and all RTP implementations must be
prepared to use procedures for SSRC collision handling, which results
in an SSRC number change. The SSRC number is randomly chosen, carried
in every RTP packet header and is not dependent on network address.
SSRC is also used as identifier to refer to separate media streams in
RTCP.</t>
<t>A media source having an SSRC identifier can be of different
types:<list style="hanging">
<t hangText="Real:">Connected to a "physical" media source, for
example a camera or microphone.</t>
<t hangText="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.</t>
<t hangText="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.</t>
</list></t>
<t>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. There exist a few other
methods to relate different SSRC where use of CNAME is inappropriate,
such as session-based <xref target="RFC4588">RTP
retransmission</xref>.</t>
<t>Note also that RTP sequence number and RTP timestamp are scoped by
SSRC and thus independent between different SSRCs.</t>
<t>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.</t>
<t>Some RTP extension mechanisms already require the RTP stacks to
handle additional SSRCs, like SSRC multiplexed <xref
target="RFC4588">RTP retransmission</xref>. However, that still only
requires handling a single media decoding chain per pair of SSRCs.</t>
</section>
<section title="CSRC">
<t>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.</t>
</section>
<section title="Payload Type">
<t>The Payload Type number is also carried in every RTP packet header
and identifies what format the RTP payload has. The term "format" here
includes whatever can be described by out-of-band signaling means for
dynamic payload types, as well as the statically allocated payload
types in <xref target="RFC3551"></xref>. In SDP the term "format"
includes media type, RTP timestamp sampling rate, codec, codec
configuration, payload format configurations, and various robustness
mechanisms such as <xref target="RFC2198">redundant
encodings</xref>.</t>
<t>The meaning of a Payload Type definition (the number) is re-used
between all media streams within an RTP session, when the definition
is either static or signaled through SDP. There however do exist cases
where each end-point have different sets of payload types due to SDP
offer/answer.</t>
<t>Although Payload Type definitions are commonly local to an RTP
Session, there are some uses where Payload Type numbers need be unique
across RTP Sessions. This is for example the case in <xref
target="RFC5583">Media Decoding Dependency </xref> where Payload Types
are used to describe media dependency across RTP Sessions.</t>
<t>Given that multiple Payload Types are defined in an RTP Session, a
media sender is free to change the Payload Type on a per packet basis.
One example of designed per-packet change of Payload Type is a speech
codec that makes use of <xref target="RFC3389">generic Comfort
Noise</xref>.</t>
<t>The RTP Payload Type in RTP is designed such that only a single
Payload Type is valid at any time instant in the SSRC's timestamp time
line, effectively time-multiplexing different Payload Types if any
switch occurs. Even when this constraint is met, having different
rates on the RTP timestamp clock 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 <xref
target="I-D.ietf-avtext-multiple-clock-rates">multiple clock rates
specification</xref>.</t>
<t>If there is a true need to send multiple Payload Types for the same
SSRC that are valid for the same RTP Timestamps, then <xref
target="RFC2198">redundant encodings</xref> can be used. Several
additional constraints than the ones mentioned above need to be met to
enable this use, one of which are that the combined payload sizes of
the different Payload Types must not exceed the transport MTU.</t>
<t>Other aspects of RTP payload format use are described in <xref
target="I-D.ietf-payload-rtp-howto">RTP Payload HowTo </xref>.</t>
</section>
</section>
<section title="Multiple Streams Alternatives">
<t>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.</t>
<t><list style="hanging">
<t hangText="SSRC Multiplexing:">Each additional Media Stream gets
its own SSRC within a RTP Session.</t>
<t hangText="Session Multiplexing:">Using additional RTP Sessions to
handle additional Media Streams</t>
<t hangText="Payload Type Multiplexing:">Using different RTP payload
types for different additional streams.</t>
</list>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 <xref target="sec-pt-mux">below section</xref>. 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 referenced and reported on.</t>
<t>This leaves us with two choices, either using SSRC multiplexing to
have multiple SSRCs from one end-point in one RTP session, or create
additional RTP sessions 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.</t>
<t>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:<list style="symbols">
<t>Multiple Media Sources of the same media type</t>
<t>Retransmission streams</t>
<t>FEC stream</t>
<t>Alternative Encoding</t>
<t>Scalability layer</t>
</list></t>
<t>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.</t>
</section>
<section anchor="sec-topologies" title="RTP Topologies and Issues">
<t>The impact of how RTP Multiplex is performed will in general vary
with how the RTP Session participants are interconnected; the <xref
target="RFC5117">RTP Topology</xref>. 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 <xref
target="RFC5117">RTP Topologies</xref>. 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.</t>
<section title="Point to Point">
<t>This is the most basic use case with an RTP session containing of
two end-points. Each end-point has one or more SSRCs.</t>
<figure align="center" title="Point to Point">
<artwork><![CDATA[
+---+ +---+
| A |<------->| B |
+---+ +---+
]]></artwork>
</figure>
<t></t>
<section anchor="sec-self-reporting" title="RTCP Reporting">
<t>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 ensure that no monitoring function
should suspect a breakage in the RTP session.</t>
<t>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.</t>
<t>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 have 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 are a true transport
issue, or simply a result of implementation.</t>
<t>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 as point to point on an RTP
level but is different. The main difference between a point to point
with two SSRC being sent from the remote end-point and a Transport
Translator with two single SSRC remote clients are that the RTT may
vary between the SSRCs (but it is not guaranteed), and that the
SSRCs may have different CNAMEs.</t>
</section>
<section title="Compound RTCP Packets">
<t>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.</t>
</section>
</section>
<section title="Point to Multipoint Using Multicast">
<t>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).</t>
<figure align="center"
title="Point to Multipoint Using Any Source Multicast">
<artwork><![CDATA[
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
]]></artwork>
</figure>
<t>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 <xref target="RFC4607">Source Specific
Multicast</xref> where only a single source (Distribution Source) can
send to the multicast group, creating a topology that looks like the
one below:</t>
<figure align="center"
title="Point to Multipoint using Source Specific Multicast">
<artwork><![CDATA[ Source-specific
+--------+ +-----+ Multicast
|Media | | | +----------------> R(1)
|Sender 1|<----->| D S | | |
+--------+ | I O | +--+ |
| 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|<----->| | |<-------------------------+
+--------+ +-----+ 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.
]]></artwork>
</figure>
<t>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 <xref
target="RFC5760">RTP extension for SSM</xref> was defined that use
unicast transmission to send RTCP from the receivers to one or more
Feedback Targets (FT).</t>
<t>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 is to use a set of
multicast groups, where each multicast group represents a particular
bit-rate. 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
transport robustification is sent in its own multicast group to allow
for interworking with legacy or to support different levels of
protection.</t>
<t>The result of this is three common behaviors for RTP
multicast:<list style="numbers">
<t>Use of multiple RTP sessions for the same media type.</t>
<t>The need for identifying RTP sessions that are related in one
of several ways.</t>
<t>The need for binding related SSRCs in different RTP sessions
together.</t>
</list></t>
<t>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.</t>
</section>
<section anchor="sec-translator"
title="Point to Multipoint Using an RTP Translator">
<t>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.</t>
<figure align="center" title="Transport Translator (Relay)">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>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, at least not without
new RTCP extensions to coherently handle the fact that some middlebox
temporarily stops a stream, preventing some receivers from reporting
on it. This consistency problem in RTCP reporting needs to be
handled.</t>
<t>The Transport Translator does not need to have an SSRC of itself,
nor need it send any RTCP reports on the flows that passes it, but it
may choose to do that.</t>
<t>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.</t>
<t>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 media stream can be
transcoded to a different bit-rate, change to 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.</t>
</section>
<section title="Point to Multipoint Using an RTP Mixer">
<t>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.</t>
<figure align="center" title="RTP Mixer">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>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.</t>
</section>
<section title="Point to Multipoint using Multiple Unicast flows">
<t>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.</t>
<figure align="center"
title="Point to Multi-Point using Multiple Unicast Transprots">
<artwork><![CDATA[
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
]]></artwork>
</figure>
<t>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.</t>
</section>
<section title="Decomposited End-Point">
<t>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.</t>
<figure align="center" title="Decomposited End-Point">
<artwork><![CDATA[
+---------------------+
| End-point A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+----\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+--------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+----/
| +------------+ |
+---------------------+
]]></artwork>
</figure>
<t>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, thus 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 isn't hidden.</t>
<t>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.</t>
<t>There exist another use case which might be considered as a
decomposited 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.</t>
<figure title="When Decomposition is a Translator">
<artwork><![CDATA[
+------+ +------+ +------+
| | | | | |
| A |------->| C |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+]]></artwork>
</figure>
<t>This type of functionality where C does something with the media
stream on behalf of A is clearly covered under the <xref
target="sec-translator">media translator definition</xref>.</t>
</section>
</section>
<section anchor="sec-pt-mux" title="Dismissing Payload Type Multiplexing">
<t>Before starting a discussion on when to use what alternative, we will
first document a number of reasons why using the payload type as a
multiplexing point for anything related to multiple streams is
unsuitable and will not be considered further.</t>
<t>If one attempts to use Payload type multiplexing beyond it's defined
usage, that has well known negative effects on RTP. To use Payload type
as the single discriminator for multiple streams implies that all the
different media streams are being sent with the same SSRC, thus using
the same timestamp and sequence number space. This has many effects:</t>
<t><list style="numbers">
<t>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.</t>
<t>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.</t>
<t>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 <xref
target="RFC4103">text/T140 payload format</xref> 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.</t>
<t>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.</t>
<t>If <xref target="RFC4588">RTP Retransmission</xref> 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.</t>
<t>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.</t>
<t>The current <xref target="RFC5104">RTCP media control
messages</xref> 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.</t>
<t>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 <xref target="RFC5761">RTP
and RTCP multiplexing</xref> where a number of payload types are
blocked due to the overlap between RTP and RTCP.</t>
<t>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.</t>
<t>It is currently not possible to signal bandwidth requirements per
media stream when using Payload Type Multiplexing.</t>
<t>Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.</t>
<t>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.</t>
</list></t>
</section>
<section anchor="sec-discussion" title="Multiple Streams Discussion">
<section title="Introduction">
<t>Using multiple media streams is a well supported feature of RTP.
However, what can be unclear for most implementors or people writing
RTP/RTCP extensions attempting to apply multiple streams, is 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.</t>
</section>
<section title="RTP/RTCP Aspects">
<t>This section discusses RTP and RTCP aspects worth considering when
selecting between SSRC multiplexing and Session multiplexing.</t>
<section anchor="sec-rtp-spec" title="The RTP Specification">
<t>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 <xref target="RFC3550"></xref>, reproduced below:</t>
<t>"For efficient protocol processing, the number of multiplexing
points should be minimized, as described in the <xref
target="ALF">integrated layer processing design principle</xref>. In
RTP, multiplexing is provided by the destination transport address
(network address and port number) which is different for each RTP
session. For example, in a teleconference composed of audio and
video media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address.</t>
<t>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: <list
style="numbers">
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>An RTP mixer would not be able to combine interleaved streams
of incompatible media into one stream.</t>
<t>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.</t>
</list></t>
<t>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.</t>
<t>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."</t>
<t>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.</t>
<t>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 <xref target="sec-pt-mux"></xref>.</t>
<t>The third argument is yet another argument against payload type
multiplexing.</t>
<t>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 is 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.</t>
<t>The fifth argument discusses network aspects that we will discuss
more below in <xref target="sec-network-aspects"></xref>. 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.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Multiple SSRC Legacy Considerations">
<t>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:</t>
<t><list style="numbers">
<t>Need to handle more than one stream simultaneously rather
than replacing an already existing stream with a new one.</t>
<t>Be capable of decoding multiple streams simultaneously.</t>
<t>Be capable of rendering multiple streams simultaneously.</t>
</list></t>
<t>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, robustification or simply to support legacy.
However, this issue does need consideration when deploying multiple
media streams within an RTP session where legacy end-points may
occur.</t>
</section>
<section title="RTP Specification Clarifications Needed">
<t>The RTP specification contains a few things that are potential
interoperability issues when using multiple SSRCs within a session.
These issues are described and discussed in <xref
target="sec-rtp-clarifications"></xref>. These should not be
considered strong arguments against using SSRC multiplexing when
otherwise appropriate, and there are some issues we expect to be
solved in the near future.</t>
</section>
<section title="Handling Varying sets of Senders">
<t>Another potential issue that needs to be considered is where a
limited 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.</t>
<t>This behavior might in certain applications be possible to limit
to a particular RTP Session and instead use multiple RTP sessions.
But in some cases it is likely unavoidable and the most appropriate
thing is to SSRC multiplex.</t>
</section>
<section title="Cross Session RTCP requests">
<t>There currently exist no functionality to make truly synchronized
and atomic RTCP requests 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 requests
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.</t>
<t>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, instead more
robust and forgiving mechanisms need to be used.</t>
</section>
<section anchor="sec-binding-related" title="Binding Related Sources">
<t>A common problem in a number of various RTP extensions has been
how to bind together related sources. This issue is common
independent of SSRC multiplexing and Session Multiplexing, and any
solution and recommendation to the problem should work equally well
for both to avoid creating barriers between using session
multiplexing and SSRC multiplexing.</t>
<t>The current solutions don't have these properties. There exist
one solution for <xref target="RFC5888">grouping RTP session
together in SDP</xref> 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 <xref target="RFC5576">group SSRCs within
a single RTP session</xref>. 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.</t>
<t>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 <xref target="RFC3550">RTP
Specification</xref> 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.</t>
<t>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 an that doesn't
fit all the RTP extensions or payload formats present in the
sessions is a issue. It is possible to create a situation where the
different mechanisms can't be combined due to the non standard SSRC
allocation behavior.</t>
<t>Existing mechanisms with known issues:<list style="hanging">
<t hangText="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, and when the first
retransmission request is sent, one must not have another
retransmission request outstanding for an SSRC which don't have
a the 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.</t>
<t hangText="Scalable Video Coding (RFC6190):">As an example of
scalable coding, <xref target="RFC6190">SVC</xref> has various
modes. The Multi Session Transmission (MST) uses Session
multiplexing to separate scalability layers. However, this
specification has failed to explicit how these layers are bound
together in cases where CNAME isn't sufficient. CNAME is no
longer sufficient when more than one media source occur within a
session that have 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
<xref target="RFC3550">RTP</xref> is to be used.</t>
<t hangText="Forward Error Correction:">If some type of FEC or
redundancy stream is being sent, it needs it's own SSRC, with
the exception of constructions like <xref
target="RFC2198">redundancy encoding</xref>. 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.</t>
</list></t>
<t>This issue appears to need action to harmonize and avoid future
shortcomings in extension specifications. A proposed solution for
handling this issue is <xref
target="I-D.westerlund-avtext-rtcp-sdes-srcname"></xref>.</t>
</section>
<section title="Forward Error Correction">
<t>There exist a number of Forward Error Correction (FEC) based
schemes for how to reduce the packet loss of the original streams.
Most of the FEC schemes will protect a single source flow. The
protection is achieved by transmitting a certain amount of redundant
information that is encoded such that it can repair one or more
packet loss over the set of packets they protect. This sequence of
redundant information also needs to be transmitted as its own media
stream, or in some cases instead of the original media stream. Thus
many of these schemes creates 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.</t>
<t>Using a Session multiplexed solution provides good support for
legacy when deploying FEC or changing the scheme used so that 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.</t>
<t>In usages involving multicast, having the FEC information on its
own multicast group and RTP session allows for flexibility, for
example when using <xref target="RFC6285">Rapid Acquisition of
Multicast Groups (RAMS)</xref>. During the RAMS burst where data is
received over unicast and where it is possible to combine with
unicast based <xref target="RFC4588">retransmission</xref>, 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 catched 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.</t>
</section>
<section title="Transport Translator Sessions">
<t>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.</t>
</section>
<section title="Multiple Media Types in one RTP session">
<t>Having different media types, like audio and video, in the same
RTP sessions is not forbidden, only recommended against as can be
seen in <xref target="sec-rtp-spec"></xref>. When using multiple
media types, there are a number of considerations:<list
style="hanging">
<t hangText="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 for two
purposes. First for the main media type and determining the
processing chain, then later for the exact configuration of the
encoder and packetization.</t>
<t hangText="Payload Type field limiations:">The total number of
Payload Types available to use in an RTP session is fairly
limited, especially if <xref target="RFC5761">Multiplexing RTP
Data and Control Packets on a Single Port</xref> is used. For
certain applications negotiating a large set of codes and
configuration may become an issue.</t>
<t hangText="Don't switch media types for an SSRC:">The primary
reasons to avoid switching from sending for example audio to
sending video using the same SSRC is the implications on a
receiver. When this happens, the processing chain in the
receiver will have to switch from one media type to another. As
the different media type's entire processing chains are
different and are connected to different outputs it is difficult
to reuse the decoding chain, which a normal codec change likely
can. Instead the entire processing chain has to be torn down and
replaced. In addition, there is likely a clock rate switching
problem, possibly resulting in synchronization loss at the point
of switching media type if some packet loss occurs.</t>
<t hangText="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.</t>
<t hangText="Decomposited end-points:">Decomposited 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
decomposited end-point will receive all media, thus doubling the
bit-rate consumption for the end-point.</t>
<t hangText="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.</t>
</list></t>
<t>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 to use this setup should
carefully review if the reasons for using a single RTP session is
sufficient to motivate this special case.</t>
</section>
</section>
<section title="Signalling Aspects">
<t>There exist various signalling solutions for establishing RTP
sessions. Many are <xref target="RFC4566">SDP</xref> based, however
SDP functionality is also dependent on the signalling protocols
carrying the SDP. Where <xref target="RFC2326">RTSP</xref> and <xref
target="RFC2974">SAP</xref> both use SDP in a declarative fashion,
<xref target="RFC3261">SIP</xref> uses SDP with the additional
definition of <xref target="RFC3264">Offer/Answer</xref>. The impact
on signalling and especially SDP needs to be considered as it can
greatly affect how to deploy a certain multiplexing point choice.</t>
<section title="Session Oriented Properties">
<t>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 that are not necessarily
strictly bound to an RTP session and could be of interest to signal
specifically for a particular media stream within the session. The
following properties have been identified as being potentially
useful to signal not only on RTP session level:<list style="symbols">
<t>Bitrate/Bandwidth exist today only at aggregate or a common
any media stream limit</t>
<t>Which SSRC that will use which RTP Payload Types</t>
</list></t>
<t>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, 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 applications requirements.</t>
<t>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.</t>
</section>
<section title="SDP Prevents Multiple Media Types">
<t>SDP encoded in its structure a 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.</t>
<t>There is a proposal in the MMUSIC WG for how one could allow
<xref target="I-D.holmberg-mmusic-sdp-bundle-negotiation">multiple
media lines describe a single underlying transport</xref> and thus
support either one RTP session with multiple media types. There is
also a solution for multiplexing multiple RTP sessions onto the same
<xref
target="I-D.westerlund-avtcore-single-transport-multiplexing">transport</xref>.</t>
</section>
</section>
<section anchor="sec-network-aspects" title="Network Apsects">
<t>The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor.</t>
<section title="Quality of Service">
<t>When it comes to Quality of Service mechanisms, they are either
flow based or marking based. <xref target="RFC2205">RSVP</xref> is
an example of a flow based mechanism, while <xref
target="RFC2474">Diff-Serv</xref> 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.</t>
<t>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.</t>
</section>
<section title="NAT and Firewall Traversal">
<t>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).</t>
<t>Below we analyze and comment on the impact of requiring more
underlying transport flows in the presence of NATs and
Firewalls:</t>
<t><list style="hanging">
<t hangText="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.</t>
<t hangText="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.</t>
<t hangText="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 immediate triggered check back. This
as ICE first finds one candidate pair that works prior to
establish multiple flows. Thus, there are no extra time until
one has found a working candidate pair. Based on that working
pair the extra time is to in parallel establish the, in most
cases 2-3, additional flows.</t>
<t hangText="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.</t>
</list></t>
<t>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 <xref
target="I-D.westerlund-avtcore-single-transport-multiplexing"></xref>.</t>
</section>
<section title="Multicast">
<t>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.</t>
<t>One limitation is that for any group, sender side adaptation to
the actual receiver properties causes a degradation for all
participants to what is supported by the receiver with the worst
conditions among the group participants. In most cases this is not
acceptable. Instead various receiver based solutions are employed to
ensure that the receivers achieve best possible performance. By
using scalable encoding and placing each scalability layer in a
different multicast group, the receiver can control the amount of
traffic it receives. To have each scalability layer on a different
multicast group, one RTP session per multicast group is used.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Multiplexing multiple RTP Session on a Single Transport">
<t>For applications that doesn't need flow based QoS and like to
save ports and NAT/FW traversal costs, there is a proposal for how
to achieve <xref
target="I-D.westerlund-avtcore-single-transport-multiplexing">multiplexing
of multiple RTP sessions over the same lower layer transport</xref>.
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.</t>
</section>
</section>
<section title="Security Aspects">
<t>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.</t>
<section title="Security Context Scope">
<t>When using <xref target="RFC3711">SRTP</xref> 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 none 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.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Key-Management for Multi-party session">
<t>Performing key-management for Multi-party session can be a
challenge. This section considers some of the issues.</t>
<t>Transport translator based session cannot use <xref
target="RFC4568">Security Description</xref> nor <xref
target="RFC5764">DTLS-SRTP</xref> without an extension as each
end-point provides it's 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 <xref
target="I-D.ietf-avt-srtp-ekt">Encrypted Key Transport</xref> are
needed or a <xref target="RFC3830">MIKEY</xref> based solution that
allows for keying all session participants with the same master
key.</t>
<t>Keying of multicast transported SRTP face similar challenges as
the transport translator case.</t>
</section>
</section>
</section>
<section title="Guidelines">
<t>This section contains a number of recommendations for implementors or
specification writers when it comes to handling multi-stream.<list
style="hanging">
<t hangText="Don't Require the same SSRC across Sessions:">As
discussed in <xref target="sec-binding-related"></xref> there exist
drawbacks in using the same SSRC in multiple RTP sessions as a
mechanism to bind related media streams together. Instead a
mechanism to explicitly signal the relation SHOULD be used, either
in RTP/RTCP or in the used signalling mechanism that establish the
RTP session(s).</t>
<t hangText="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 telepresence room where there are three
cameras, and each camera captures 2 persons sitting at the table,
sending each camera as its own SSRC within a single RTP session is
recommended.</t>
<t
hangText="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.</t>
<t hangText="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 <xref target="RFC5888">The Session
Description Protocol (SDP) Grouping Framework.</xref></t>
<t
hangText="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. If it is, then any generic extensions are
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.</t>
<t hangText="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 if a particular solution only supports
one of the multiplexing choices.</t>
</list></t>
<t>This discussion and guidelines points out 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:<list style="hanging">
<t hangText="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 <xref
target="I-D.westerlund-avtext-rtcp-sdes-srcname">proposal</xref>
exist for a new SDES item SRCNAME that also can be used with the
a=ssrc SDP attribute to provide signalling layer binding
information.</t>
<t hangText="SSRC limiations 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
<xref target="I-D.westerlund-avtcore-max-ssrc"></xref>.</t>
</list></t>
<t></t>
</section>
<section anchor="sec-rtp-clarifications"
title="RTP Specification Clarifications">
<t>This section describes a number of clarifications to the RTP
specifications that are likely necessary for aligned behavior when RTP
sessions contains more SSRCs than one local and one remote.</t>
<section title="RTCP Reporting from all SSRCs">
<t>When one have multiple SSRC in an RTP node, then 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.</t>
</section>
<section title="RTCP Self-reporting">
<t>For any RTP node that sends more than one SSRC, there exist 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.</t>
<!--MW: Moved the proposal out of the text. I am not certain this is fly and
don't have time to consider it well enough.
Our proposal is that RFC3550 is updated to clarify that one needs to report
on all SSRCs one knows exist with the sole exception of the local SSRCs
that has the same CNAME as the SSRC providing the report. That way a third
party monitor can use the CNAME data from the various SSRCs to determine
that the gap in reporting is not valid.
Note: There is nothing preventing a node to send an SSRC with the same
CNAME as one or more other SSRCs originating from another node. In fact
an obvious case for this to occur is when the creation of Forward Error
Correction data is performed at a boundary to another transport domain. Thus
any node in this case would need to report on both the actuall arrived
stream and send sender reports on the stream it creates.
The result of the above exception is that a 3rd party monitor can't
detect if there is an fault in the transport from the orignal source and
the secondary node generating the new source with shared CNAME.-->
</section>
<section title="Combined RTCP Packets">
<t>When a node contains multiple SSRCs, it is questionable if an RTCP
compound packet can only contain RTCP packets from a single SSRC or if
multiple SSRCs can include their packets in a joint compound packet.
The high level question is a matter for any receiver processing on
what to expect. In addition to that question there is the issue of how
to use the RTCP timer rules in these cases, as the existing rules are
focused on determining when a single SSRC can send.</t>
</section>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t></t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t></t>
</section>
</middle>
<back>
<references title="Normative References">
&rfc2119;
&rfc3550;
</references>
<references title="Informative References">
&rfc2198;
&rfc2205;
&rfc2326;
&rfc2474;
&rfc2974;
&rfc3261;
&rfc3264;
&rfc3389;
&rfc3551;
&rfc3711;
&rfc3830;
&rfc4103;
&rfc4566;
&rfc4568;
&rfc4588;
&rfc4607;
&rfc5104;
&rfc5117;
&rfc5583;
&rfc5576;
&rfc5760;
&rfc5761;
&rfc5764;
&rfc5888;
&rfc6190;
&rfc6285;
&draft-ietf-avtext-multiple-clock-rates;
&draft-ietf-payload-rtp-howto;
&draft-ietf-avt-srtp-ekt;
<reference anchor="I-D.westerlund-avtext-rtcp-sdes-srcname">
<front>
<title>RTCP SDES Item SRCNAME to Label Individual Sources</title>
<author fullname="Magnus Westerlund" initials="M."
surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 13 11</phone>
<email>bo.burman@ericsson.com</email>
</address>
</author>
<author fullname="Patrik Sandgren" initials="P." surname="Sandgren">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 717 97 41</phone>
<email>patrik.sandgren@ericsson.com</email>
</address>
</author>
<date day="24" month="October" year="2011" />
</front>
<seriesInfo name="Internet-Draft"
value="draft-westerlund-avtext-rtcp-sdes-srcname" />
<format target="http://www.ietf.org/internet-drafts/draft-westerlund-avtext-rtcp-sdes-srcname-00.txt"
type="TXT" />
</reference>
<reference anchor="I-D.westerlund-avtcore-max-ssrc">
<front>
<title>Multiple Synchronization sources (SSRC) in RTP Session
Signaling</title>
<author fullname="Magnus Westerlund" initials="M."
surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 13 11</phone>
<email>bo.burman@ericsson.com</email>
</address>
</author>
<author fullname="Fredrik Jansson" initials="F." surname="Jansson">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>Kista</city>
<region></region>
<code>SE-164 80</code>
<country>Sweden</country>
</postal>
<phone>+46 10 719 00 00</phone>
<facsimile></facsimile>
<email>fredrik.k.jansson@ericsson.com</email>
<uri></uri>
</address>
</author>
<date day="24" month="October" year="2011" />
</front>
<seriesInfo name="Internet-Draft"
value="draft-westerlund-avtcore-max-ssrc" />
<format target="http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-max-ssrc-00.txt"
type="TXT" />
</reference>
<reference anchor="I-D.westerlund-avtcore-single-transport-multiplexing">
<front>
<title>Multiple RTP Session on a Single Lower-Layer
Transport</title>
<author fullname="Magnus Westerlund" initials="M."
surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<date day="11" month="October" year="2011" />
</front>
<seriesInfo name="Internet-Draft"
value="draft-westerlund-avtcore-transport-multiplexing" />
<format target="http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-transport-multiplexing-00.txt"
type="TXT" />
</reference>
<reference anchor="ALF">
<front>
<title>Architectural Considerations for a New Generation of
Protocols</title>
<author initials="D." surname="Clark">
<organization>IEEE Computer Communications Review, Vol.
20(4)</organization>
</author>
<author initials="D." surname="Tennenhouse">
<organization></organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country></country>
</postal>
<phone></phone>
<facsimile></facsimile>
<email></email>
<uri></uri>
</address>
</author>
<date month="September" year="1990" />
</front>
<seriesInfo name="SIGCOMM Symposium on Communications Architectures and Protocols"
value="(Philadelphia, Pennsylvania), pp. 200--208, IEEE Computer Communications Review, Vol. 20(4)" />
</reference>
<?rfc include='reference.I-D.holmberg-mmusic-sdp-bundle-negotiation'?>
</references>
</back>
</rfc>
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