One document matched: draft-westerlund-avtcore-multiplex-architecture-01.xml
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<rfc category="info"
docName="draft-westerlund-avtcore-multiplex-architecture-01"
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="12" month="March" year="2012" />
<abstract>
<t>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.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t><xref target="RFC3550">Real-time Transport Protocol (RTP)</xref> 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.</t>
<t>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.</t>
<t>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.</t>
<t>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.</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. Some arch-types of RTP usage are discussed. Finally, some
recommendations and examples are provided.</t>
<t>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.</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 carries 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 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.</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="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.</t>
<t hangText="RTP Session:">As defined by <xref
target="RFC3550"></xref>, 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.</t>
<t hangText="Source:">See Media Source.</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.</t>
<section title="Session">
<t>The RTP Session is the highest semantic level in RTP and contains
all of the RTP functionality.<list style="hanging">
<t hangText="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.<list style="hanging">
<t hangText="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 <xref target="RFC5761">RTP/RTCP
multiplex</xref> there is only a single port. Another example
is <xref target="RFC4566">SDP signalling</xref>, where the
<xref target="RFC5888">grouping framework</xref> 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.</t>
<t hangText="Usage:">Identify separate RTP Sessions.</t>
<t hangText="Uniqueness:">Globally unique within the general
communication context for the specific end-point.</t>
<t hangText="Inter-relation:">Depending on the underlying
transport and signalling protocol.</t>
</list></t>
<t hangText="Special Restrictions:">None.</t>
</list></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>An RTP Session serves one or more Media Sources, each sending a
Media Stream.<list style="hanging">
<t hangText="Identifier:">Synchronization Source (SSRC), 32-bit
unsigned number.<list style="hanging">
<t hangText="Position:">In every RTP and RTCP packet header.
May be present in RTCP payload. May be present in SDP
signalling.</t>
<t hangText="Usage:">Identify individual Media Sources within
an RTP Session. Refer to individual Media Sources in RTCP
messages and SDP signalling.</t>
<t hangText="Uniqueness:">Randomly chosen, globally unique
within an RTP Session and not dependent on network
address.</t>
<t hangText="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 <xref target="RFC5576">SSRC grouping </xref>. When
CNAME is inappropriate or insufficient, there exist a few
other methods to relate different SSRC. One such case is
session-based <xref target="RFC4588">RTP
retransmission</xref>. In some cases, the same SSRC Identifier
value is used to relate streams in two different RTP Sessions,
such as in Multi-Session Transmission of <xref
target="RFC6190">scalable video</xref>.</t>
</list></t>
<t hangText="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.</t>
</list></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.</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>Each Media Stream can be represented in various encoding
formats.<list style="hanging">
<t hangText="Identifier:">Payload Type number.<list
style="hanging">
<t hangText="Position:">In every RTP header and in SDP
signalling.</t>
<t hangText="Usage:">Identify a specific Media Stream encoding
format. The format definition may be taken from <xref
target="RFC3551"></xref> 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 <xref target="RFC2198">redundant
encodings</xref>.</t>
<t hangText="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 <xref target="RFC3389">generic Comfort
Noise</xref>.</t>
<t hangText="Inter-relation:">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. Another example
is session-based <xref target="RFC4588">RTP
retransmission</xref>.</t>
</list></t>
<t hangText="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 <xref
target="I-D.ietf-avtext-multiple-clock-rates">multiple clock rates
specification</xref>.</t>
</list></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 is 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"></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 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.</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</t>
<t>Retransmission streams</t>
<t>FEC stream</t>
<t>Alternative Encodings</t>
<t>Scalability layers</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 two
end-points. Each end-point has one or more SSRCs.</t>
<figure align="center" anchor="fig-point-to-point"
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 to 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 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.</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 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
scenario, the RTT may vary between the SSRCs (but it is not
guaranteed), and the SSRCs may also 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). There are
large commercial deployments of multicast for applications like
IPTV.</t>
<figure align="center" anchor="fig-multipoint-asm"
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" anchor="fig-multipoint-ssm"
title="Point to Multipoint using Source Specific Multicast">
<artwork><![CDATA[
+--------+ +-----+
|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.
]]></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 to 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 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.</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 possible 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" anchor="fig-translator"
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, 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.</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 pass 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. 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 <xref target="RFC3264">SDP offer/answer</xref> has issues
with ensuring this property.</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 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.</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" anchor="fig-mixer" 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" anchor="fig-multi-unicast"
title="Point to Multi-Point using Multiple Unicast Transports">
<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>
<t>There exists an alternative structure for establishing the above
<xref target="fig-multi-unicast">communication scenario</xref> 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.</t>
</section>
<section title="De-composite 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" anchor="fig-de-composite"
title="De-composite 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.</t>
<t>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.</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
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.</t>
<figure anchor="fig-de-composite-translator"
title="When De-composition 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-discussion" title="Multiple Streams Discussion">
<section title="Introduction">
<t>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.</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 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.</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 anchor="sec-multi-media-rec"
title="Different Media Types Recommendations">
<t>The above quote from <xref target="RFC3550">RTP</xref> includes
a strong recommendation:<list style="empty">
<t>"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>
</list></t>
<t>It has been identified in <xref
target="I-D.alvestrand-rtp-sess-neutral">"Why RTP Sessions Should
Be Content Neutral"</xref> 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
potential issues in having multiple media types in the same RTP
session in <xref target="sec-multiple-media-types"></xref>. 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 <xref target="ALF">"Architectural
considerations for a new generation of protocols"</xref>.</t>
</section>
</section>
<section title="Handling Varying sets of Senders">
<t>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.</t>
<t>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.</t>
</section>
<section title="Cross Session RTCP Requests">
<t>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.</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, forcing it to
use more robust and forgiving mechanisms.</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 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.</t>
<t>The current solutions do not have these properties. There exists
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 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.</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 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.</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 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 <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 its 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 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.</t>
<t>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.</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 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.</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>
<section title="Interworking">
<t>There are several different kinds of interworking, and this section
discusses two related ones. The interworking between different
applications and the implications of potentially different choices of
usage of RTP's multiplexing points. The second topic relates to what
limitations may have to be considered working with some legacy
applications.</t>
<section title="Interworking Applications">
<t>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.</t>
<t>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 <xref
target="RFC4588">retransmission</xref>. Then it is more likely or
maybe even required to use the same SSRC in the different RTP
sessions.</t>
<t>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 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.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Multiple SSRC Legacy Considerations">
<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>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, 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.</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,
while <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 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:<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 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.</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 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 an accepted WG item in the MMUSIC WG to define how <xref
target="I-D.holmberg-mmusic-sdp-bundle-negotiation">multiple media
lines describe a single underlying transport</xref> and thus it
becomes possible in SDP to define one RTP session with multiple
media types.</t>
</section>
<section title="Media Stream Usage">
<t>Media streams being transported in RTP has some particular usage
in an RTP application. This usage of the media stream is in many
applications so far implicitly signalled. For example by having all
audio media streams arriving in the only audio RTP session they are
to be decoded, mixed and played out. However, in more advanced
applications that use multiple media streams there will be more than
a single usage or purpose among the set of media streams being sent
or received. RTP applications will need to signal this usage
somehow. Here the choice of SSRC multiplexing versus session
multiplexing will have significant impact. If one uses SSRC
multiplexing to its full extent one will have to explicitly indicate
for each SSRC what its' usage and purpose are using some signalling
between the application instances.</t>
<t>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.</t>
<t>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.</t>
</section>
</section>
<section anchor="sec-network-aspects" title="Network Aspects">
<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 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.</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>
<t
hangText="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.</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 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 and where usage of multiple
media types in one RTP session is not suitable, 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 anchor="sec-security-aspects" 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 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.</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 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 <xref
target="I-D.ietf-avt-srtp-ekt">Encrypted Key Transport</xref> is
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 title="Complexity Implications">
<t>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 <xref target="RFC3711">SRTP</xref>, the actual security
context and exact crypto key are determined by the SSRC field value.
If one changes it, the encryption and authentication tag must be
performed using another key. Thus changing the SSRC value implies a
decryption using the old SSRC and its security context followed by
an encryption using the new one.</t>
<t>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.</t>
</section>
</section>
<section anchor="sec-multiple-media-types"
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 earlier
discussed in <xref target="sec-multi-media-rec"></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 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.</t>
<t hangText="Payload Type field limitations:">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 this may become an issue.</t>
<t
hangText="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 <xref target="sec-pt-mux"></xref> 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.</t>
<t hangText="Do not switch media types for an SSRC:">The primary
reasons to avoid switching from sending for example audio to
sending video using the same SSRC is the implications on a
receiver. When this happens, the processing chain in the receiver
will have to switch from one media type to another. As the
different media type's entire processing chains are different and
are connected to different outputs it is difficult to reuse the
decoding chain, which a normal codec change likely can. Instead
the entire processing chain has to be torn down and replaced. In
addition, there is likely a clock rate switching problem, possibly
resulting in synchronization loss at the point of switching media
type if some packet loss occurs. So this is a behavior that shall
be avoided.</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="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.</t>
<t hangText="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.</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>
<t hangText="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.</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 using this setup should carefully review if the
reasons for using a single RTP session are sufficient to motivate the
needed special handling.</t>
</section>
</section>
<section title="Arch-Types">
<t>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.</t>
<section title="Single SSRC per Session">
<t>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.</t>
<t>The Pros:<list style="numbers">
<t>This arch-type has great legacy interoperability potential as
it will not tax any RTP stack implementations.</t>
<t>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.</t>
<t>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.</t>
<t>It is possible to control security association per RTP session
with current key-management.</t>
</list></t>
<t>The Cons:<list style="letters">
<t>The number of required RTP sessions cannot really be higher,
which has the implications:<list style="symbols">
<t>Linear growth of the amount of NAT/FW state with number of
media streams.</t>
<t>Increased delay and resource consumption from NAT/FW
traversal.</t>
<t>Likely larger signalling message and signalling processing
requirement due to the amount of session related
information.</t>
<t>Higher potential for a single media stream to fail during
transport between the end-points.</t>
</list></t>
<t>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.</t>
<t>The port consumption may become a problem for centralized
services, where the central node's port consumption grows rapidly
with the number of sessions.</t>
<t>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.</t>
<t>Cross session RTCP requests needs is likely to exist and may
cause issues.</t>
<t>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.</t>
<t>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.</t>
<t>For most security mechanisms, each RTP session or transport
flow requires individual key-management and security association
establishment thus increasing the overhead.</t>
<t>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.</t>
</list></t>
<t>RTP applications that need to inter-work with legacy RTP
applications, like VoIP and video conferencing, can potentially
benefit from this structure. However, a large number of media
descriptions in SDP can also run into issues with existing
implementations. For any application needing a larger number of media
flows, the overhead can become very significant. This structure is
also not suitable for multi-party sessions, as any given media stream
from each participant, although having same usage in the application,
must have its own RTP session. In addition, the dynamic behavior that
can arise in multi-party applications can tax the signalling system
and make timely media establishment more difficult.</t>
</section>
<section anchor="sec-multiple-ssrc-single-session"
title="Multiple SSRCs of the Same Media Type">
<t>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.</t>
<t>The Pros:<list style="numbers">
<t>Low number of RTP sessions needed compared to single SSRC case.
This implies:<list style="symbols">
<t>Reduced NAT/FW state</t>
<t>Lower NAT/FW Traversal Cost in both processing and
delay.</t>
</list></t>
<t>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.</t>
<t>Works well with media type de-composite end-points.</t>
<t>Enables Flow-based QoS with different prioritization between
media types.</t>
<t>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.</t>
<t>Low overhead for security association establishment.</t>
</list></t>
<t>The Cons:<list style="letters">
<t>May have some need for cross session RTCP requests for things
that affect both media types in an asynchronous way.</t>
<t>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.</t>
<t>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.</t>
</list></t>
<t>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.</t>
</section>
<section title="Multiple Sessions for one Media type">
<t>In this arch-type one goes one step further than in the <xref
target="sec-multiple-ssrc-single-session">above</xref> 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.</t>
<t>The Pros:<list style="numbers">
<t>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.</t>
<t>Detailed indication of the application's usage of the media
stream, where multiple different usages exist.</t>
<t>Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely
signalling.</t>
<t>Enables detailed QoS prioritization for flow based
mechanisms.</t>
<t>Works well with de-composite end-points.</t>
<t>Handles dynamic usage of media streams well.</t>
<t>For transport translator based multi-party sessions, this
structure allows for improved control of which type of media
streams an end-point receives.</t>
<t>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.</t>
</list></t>
<t>The Cons:<list style="letters">
<t>Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.</t>
<t>Increased amount of session configuration state.</t>
<t>May need synchronized cross-session RTCP requests and require
some consideration due to this.</t>
<t>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.</t>
<t>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.</t>
<t>Higher overhead for security association establishment.</t>
<t>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.</t>
</list></t>
<t>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.</t>
</section>
<section title="Multiple Media Types in one Session">
<t>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.</t>
<t>The Pros:<list style="numbers">
<t>Single RTP session which implies:<list style="symbols">
<t>Minimal NAT/FW state.</t>
<t>Minimal NAT/FW Traversal Cost.</t>
<t>Fate-sharing for all media flows.</t>
</list></t>
<t>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.</t>
<t>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.</t>
<t>Minimal overhead for security association establishment.</t>
</list></t>
<t>The Cons:<list style="letters">
<t>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.</t>
<t>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.</t>
<t>Does enforce higher bandwidth and processing on de-composite
end-points.</t>
<t>Flow based QoS cannot provide separate treatment to some media
streams compared to other in the single RTP session.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>If the applications need finer control over which session
participants that are included in different sets of security
associations, most key-management will have difficulties
establishing such a session.</t>
</list></t>
<t>The analysis in this document and considerations in <xref
target="sec-multiple-media-types"></xref> 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.</t>
</section>
<section title="Summary">
<t>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.</t>
<t>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.</t>
</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="Do not 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. 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).</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 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.</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>, using some
appropriate grouping semantics.</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. 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.</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 which of the multiplexing choices a
particular solution supports.</t>
</list></t>
<t></t>
</section>
<section title="Proposal for Future Work">
<t>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:<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 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
<xref target="I-D.westerlund-avtcore-max-ssrc"></xref>.</t>
</list></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 contain 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, 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 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.</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>There is discussion of the security implications of choosing SSRC vs
Session multiplexing in <xref target="sec-security-aspects"></xref>.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>The authors would like to thanks Harald Alvestrand for providing
input into the discussion regarding multiple media types in a single RTP
session.</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'?>
<?rfc include='reference.I-D.alvestrand-rtp-sess-neutral'?>
</references>
<section anchor="sec-pt-mux" title="Dismissing Payload Type Multiplexing">
<t>This section documents a number of reasons why using the payload type
as a multiplexing point for most things related to multiple streams is
unsuitable. If one attempts to use Payload type multiplexing beyond it's
defined usage, that has well known negative effects on RTP. To use
Payload type as the single discriminator for multiple streams implies
that all the different media streams are being sent with the same SSRC,
thus using the same timestamp and sequence number space. This has many
effects:</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>
</back>
</rfc>
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