rfc8853xml2.original.xml   rfc8853.xml 
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<?rfc tocompact="yes"?> <rfc xmlns:xi="http://www.w3.org/2001/XInclude" category="std"
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<?rfc symrefs="yes"?> sortRefs="true" version="3" docName="draft-ietf-mmusic-sdp-simulcast-14">
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<rfc category="std" docName="draft-ietf-mmusic-sdp-simulcast-14"
ipr="trust200902" submissionType="IETF">
<front> <front>
<title abbrev="Simulcast">Using Simulcast in SDP and RTP Sessions</title> <title abbrev="Simulcast">Using Simulcast in Session Description Protocol (S DP) and RTP Sessions</title>
<seriesInfo name="RFC" value="8853"/>
<author fullname="Bo Burman" initials="B." surname="Burman"> <author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization> <organization>Ericsson</organization>
<address> <address>
<postal> <postal>
<street>Gronlandsgatan 31</street> <street>Gronlandsgatan 31</street>
<city>SE-164 60 Stockholm</city> <city>SE-164 60 Stockholm</city>
<region/> <region/>
<code/> <code/>
<country>Sweden</country> <country>Sweden</country>
</postal> </postal>
<phone/> <phone/>
<facsimile/>
<email>bo.burman@ericsson.com</email> <email>bo.burman@ericsson.com</email>
<uri/> <uri/>
</address> </address>
</author> </author>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund"> <author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization> <organization>Ericsson</organization>
<address> <address>
<postal> <postal>
<street>Torshamnsgatan 23</street> <street>Torshamnsgatan 23</street>
<city>SE-164 83 Stockholm</city> <city>SE-164 83 Stockholm</city>
<country>Sweden</country> <country>Sweden</country>
</postal> </postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email> <email>magnus.westerlund@ericsson.com</email>
</address> </address>
</author> </author>
<author fullname="Suhas Nandakumar" initials="S." surname="Nandakumar"> <author fullname="Suhas Nandakumar" initials="S." surname="Nandakumar">
<organization>Cisco</organization> <organization>Cisco</organization>
<address> <address>
<postal> <postal>
<street>170 West Tasman Drive</street> <street>170 West Tasman Drive</street>
<city>San Jose</city> <city>San Jose</city>
<region>CA</region> <region>CA</region>
<code>95134</code> <code>95134</code>
<country>United States of America</country>
<country>USA</country>
</postal> </postal>
<phone/> <phone/>
<facsimile/>
<email>snandaku@cisco.com</email> <email>snandaku@cisco.com</email>
<uri/> <uri/>
</address> </address>
</author> </author>
<author fullname="Mo Zanaty" initials="M." surname="Zanaty"> <author fullname="Mo Zanaty" initials="M." surname="Zanaty">
<organization>Cisco</organization> <organization>Cisco</organization>
<address> <address>
<postal> <postal>
<street>170 West Tasman Drive</street> <street>170 West Tasman Drive</street>
<city>San Jose</city> <city>San Jose</city>
<region>CA</region> <region>CA</region>
<code>95134</code> <code>95134</code>
<country>United States of America</country>
<country>USA</country>
</postal> </postal>
<phone/> <phone/>
<facsimile/>
<email>mzanaty@cisco.com</email> <email>mzanaty@cisco.com</email>
<uri/> <uri/>
</address> </address>
</author> </author>
<date month="May" year="2020"/>
<date day="5" month="March" year="2019"/> <keyword>Conference</keyword>
<keyword>multi-party</keyword>
<keyword>middlebox</keyword>
<keyword>MCU</keyword>
<keyword>SFU</keyword>
<keyword>media</keyword>
<keyword>video</keyword>
<keyword>restrictions</keyword>
<keyword>RTCP</keyword>
<keyword>RID</keyword>
<keyword>RtpStreamId</keyword>
<abstract> <abstract>
<t>In some application scenarios it may be desirable to send multiple
<t>In some application scenarios, it may be desirable to send multiple
differently encoded versions of the same media source in different RTP differently encoded versions of the same media source in different RTP
streams. This is called simulcast. This document describes how to streams. This is called simulcast. This document describes how to
accomplish simulcast in RTP and how to signal it in SDP. The described accomplish simulcast in RTP and how to signal it in the Session
solution uses an RTP/RTCP identification method to identify RTP streams Description Protocol (SDP). The described solution uses an RTP/RTCP
belonging to the same media source, and makes an extension to SDP to identification method to identify RTP streams
relate those RTP streams as being different simulcast formats of that belonging to the same media source and makes an extension to SDP to
media source. The SDP extension consists of a new media level SDP indicate that those RTP streams are different simulcast formats of that
media source. The SDP extension consists of a new media-level SDP
attribute that expresses capability to send and/or receive simulcast RTP attribute that expresses capability to send and/or receive simulcast RTP
streams.</t> streams.</t>
</abstract> </abstract>
</front> </front>
<middle> <middle>
<section anchor="sec-intro" title="Introduction"> <section anchor="sec-intro" numbered="true" toc="default">
<t>Most of today's multiparty video conference solutions make use of <name>Introduction</name>
<t>Most of today's multiparty video-conference solutions make use of
centralized servers to reduce the bandwidth and CPU consumption in the centralized servers to reduce the bandwidth and CPU consumption in the
endpoints. Those servers receive RTP streams from each participant and endpoints. Those servers receive RTP streams from each participant and
send some suitable set of possibly modified RTP streams to the rest of send some suitable set of possibly modified RTP streams to the rest of
the participants, which usually have heterogeneous capabilities (screen the participants, which usually have heterogeneous capabilities (screen
size, CPU, bandwidth, codec, etc). One of the biggest issues is how to size, CPU, bandwidth, codec, etc.). One of the biggest issues is how to
perform RTP stream adaptation to different participants' constraints perform RTP stream adaptation to different participants' constraints
with the minimum possible impact on both video quality and server with the minimum possible impact on both video quality and server
performance.</t> performance.</t>
<t>Simulcast is defined in this memo as the act of simultaneously <t>Simulcast is defined in this memo as the act of simultaneously
sending multiple different encoded streams of the same media source, sending multiple different encoded streams of the same media source --
e.g. the same video source encoded with different video encoder types or e.g., the same video source encoded with different video-encoder types or
image resolutions. This can be done in several ways and for different image resolutions. This can be done in several ways and for different
purposes. This document focuses on the case where it is desirable to purposes. This document focuses on the case where it is desirable to
provide a media source as multiple encoded streams over <xref provide a media source as multiple encoded streams over <xref target="RFC3
target="RFC3550">RTP</xref> towards an intermediary so that the 550" format="default">RTP</xref> towards an intermediary so that the
intermediary can provide the wanted functionality by selecting which RTP intermediary can provide the wanted functionality by selecting which RTP
stream(s) to forward to other participants in the session, and more stream(s) to forward to other participants in the session, and more
specifically how the identification and grouping of the involved RTP specifically how the identification and grouping of the involved RTP
streams are done.</t> streams are done.</t>
<t>The intended scope of the defined mechanism is to support negotiation <t>The intended scope of the defined mechanism is to support negotiation
and usage of simulcast when using SDP offer/answer and media transport and usage of simulcast when using SDP offer/answer and media transport
over RTP. The media transport topologies considered are point to point over RTP. The media transport topologies considered are point-to-point
RTP sessions as well as centralized multi-party RTP sessions, where a RTP sessions, as well as centralized multiparty RTP sessions, where a
media sender will provide the simulcasted streams to an RTP middlebox or media sender will provide the simulcasted streams to an RTP middlebox or
endpoint, and middleboxes may further distribute the simulcast streams endpoint, and middleboxes may further distribute the simulcast streams
to other middleboxes or endpoints. Simulcast could, as part of a to other middleboxes or endpoints. Simulcast could be used point to point
distributed multi-party scenario, be used point-to-point between between
middleboxes. Usage of multicast or broadcast transport is out of scope middleboxes as part of a distributed multiparty scenario. Usage of
multicast or broadcast transport is out of scope
and left for future extensions.</t> and left for future extensions.</t>
<t>This document describes a few scenarios that motivate the use of <t>This document describes a few scenarios that motivate the use of
simulcast, and also defines the needed RTP/RTCP and SDP signaling for simulcast and also defines the needed RTP/RTCP and SDP signaling for
it.</t> it.</t>
</section> </section>
<section anchor="sec-definitions" numbered="true" toc="default">
<section anchor="sec-definitions" title="Definitions"> <name>Definitions</name>
<t/> <section numbered="true" toc="default">
<name>Terminology</name>
<section title="Terminology">
<t>This document makes use of the terminology defined in <xref <t>This document makes use of the terminology defined in <xref
target="RFC7656">RTP Taxonomy</xref>, and <xref target="RFC7667">RTP target="RFC7656" format="default">"A Taxonomy of Semantics and
Topologies</xref>. The following terms are especially noted or here Mechanisms for Real-Time
defined:<list style="hanging"> Transport Protocol (RTP) Sources"</xref> and <xref target="RFC7667"
<t hangText="RTP Mixer:">An RTP middle node, defined in <xref format="default">"RTP Topologies"</xref>. The following terms are
target="RFC7667"/> (Section 3.6 to 3.9).</t> especially noted or here defined:</t>
<dl newline="false" spacing="normal">
<t hangText="RTP Session:">An association among a group of <dt>RTP mixer:</dt>
participants communicating with RTP, as defined in <xref <dd>An RTP middlebox, in the wide sense of the term, encompassing
target="RFC3550"/> and amended by <xref target="RFC7656"/>.</t> Sections <xref target="RFC7667" section="3.6" sectionFormat="bare"/>
to <xref target="RFC7667" section="3.9" sectionFormat="bare"/> of
<t hangText="RTP Stream:">A stream of RTP packets containing media <xref target="RFC7667" format="default"/>.</dd>
data, as defined in <xref target="RFC7656"/>.</t> <dt>RTP session:</dt>
<dd>An association among a group of
<t hangText="RTP Switch:">A common short term for the terms participants communicating with RTP, as defined in <xref target="RFC
3550" format="default"/> and amended by <xref target="RFC7656" format="default"/
>.</dd>
<dt>RTP stream:</dt>
<dd>A stream of RTP packets containing media
data, as defined in <xref target="RFC7656" format="default"/>.</dd>
<dt>RTP switch:</dt>
<dd>A common short term for the terms
"switching RTP mixer", "source projecting middlebox", and "video "switching RTP mixer", "source projecting middlebox", and "video
switching MCU" as discussed in <xref target="RFC7667"/>.</t> switching Multipoint Control Unit (MCU)", as discussed in <xref targ
et="RFC7667" format="default"/>.</dd>
<t hangText="Simulcast Stream:">One encoded stream or dependent <dt>Simulcast stream:</dt>
<dd>One encoded stream or dependent
stream from a set of concurrently transmitted encoded streams and stream from a set of concurrently transmitted encoded streams and
optional dependent streams, all sharing a common media source, as optional dependent streams, all sharing a common media source, as
defined in <xref target="RFC7656"/>. For example, HD and thumbnail defined in <xref target="RFC7656" format="default"/>. For example, H D and thumbnail
video simulcast versions of a single media source sent video simulcast versions of a single media source sent
concurrently as separate RTP Streams.</t> concurrently as separate RTP streams.</dd>
<dt>Simulcast format:</dt>
<t hangText="Simulcast Format:">Different formats of a simulcast <dd>Different formats of a simulcast
stream serve the same purpose as alternative RTP payload types in stream serve the same purpose as alternative RTP payload types in
non-simulcast SDP: to allow multiple alternative media formats for nonsimulcast SDP: to allow multiple alternative media formats for
a given RTP stream. As for multiple RTP payload types on the a given RTP stream. As for multiple RTP payload types on the
m-line in <xref target="RFC3264">offer/answer</xref>, any one of "m=" line in <xref target="RFC3264" format="default">offer/answer</x ref>, any one of
the negotiated alternative formats can be used in a single RTP the negotiated alternative formats can be used in a single RTP
stream at a given point in time, but not more than one (based on stream at a given point in time, but not more than one (based on
RTP timestamp). What format is used can change dynamically from RTP timestamp). What format is used can change dynamically from
one RTP packet to another.</t> one RTP packet to another.</dd>
</list></t> </dl>
</section> </section>
<section numbered="true" toc="default">
<section title="Requirements Language"> <name>Requirements Language</name>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", <t>
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
"OPTIONAL" in this document are to be interpreted as described in BCP "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL
14 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and only NOT</bcp14>", "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
when, they appear in all capitals, as shown here.</t> "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
"<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are
to be interpreted as
described in BCP&nbsp;14 <xref target="RFC2119" format="default"/> <xref
target="RFC8174" format="default"/>
when, and only when, they appear in all capitals, as shown here.
</t>
</section> </section>
</section> </section>
<section anchor="sec-use-cases" numbered="true" toc="default">
<section anchor="sec-use-cases" title="Use Cases"> <name>Use Cases</name>
<t>The use cases of simulcast described in this document relate to a <t>The use cases of simulcast described in this document relate to a
multi-party communication session where one or more central nodes are multiparty communication session where one or more central nodes are
used to adapt the view of the communication session towards individual used to adapt the view of the communication session towards individual
participants, and facilitate the media transport between participants. participants and facilitate the media transport between participants.
Thus, these cases target the RTP Mixer type of topology.</t> Thus, these cases target the RTP mixer type of topology.</t>
<t>There are two principal approaches for an RTP mixer to provide this
<t>There are two principal approaches for an RTP Mixer to provide this
adapted view of the communication session to each receiving adapted view of the communication session to each receiving
participant:<list style="symbols"> participant:</t>
<t>Transcoding (decoding and re-encoding) received RTP streams with <ul spacing="normal">
<li>Transcoding (decoding and re-encoding) received RTP streams with
characteristics adapted to each receiving participant. This often characteristics adapted to each receiving participant. This often
include mixing or composition of media sources from multiple includes mixing or composition of media sources from multiple
participants into a mixed media source originated by the RTP Mixer. participants into a mixed media source originated by the RTP mixer.
The main advantage of this approach is that it achieves close to The main advantage of this approach is that it achieves
optimal adaptation to individual receiving participants. The main close-to-optimal adaptation to individual receiving
participants. The main
disadvantages are that it can be very computationally expensive to disadvantages are that it can be very computationally expensive to
the RTP Mixer, typically degrades media Quality of Experience (QoE) the RTP mixer, typically degrades media Quality of Experience (QoE)
such as end-to-end delay for the receiving participants, and such as creating end-to-end delay for the receiving participants, and
requires RTP Mixer access to media content.</t> requires the RTP mixer to have access to media content.</li>
<li>Switching a subset of all received RTP streams or substreams to
<t>Switching a subset of all received RTP streams or sub-streams to
each receiving participant, where the used subset is typically each receiving participant, where the used subset is typically
specific to each receiving participant. The main advantages of this specific to each receiving participant. The main advantages of this
approach are that it is computationally cheap to the RTP Mixer, has approach are that it is computationally cheap to the RTP mixer, has
very limited impact on media QoE, and does not require RTP Mixer very limited impact on media QoE, and does not require the RTP mixer
(full) access to media content. The main disadvantage is that it can to have (full) access to media content. The main disadvantage is
be difficult to combine a subset of received RTP streams into a that it can be difficult to combine a subset of received RTP streams in
perfect fit to the resource situation of a receiving participant. It to a
perfect fit for the resource situation of a receiving participant. It
is also a disadvantage that sending multiple RTP streams consumes is also a disadvantage that sending multiple RTP streams consumes
more network resources from the sending participant to the RTP more network resources from the sending participant to the RTP
Mixer.</t> mixer.</li>
</list></t> </ul>
<t>The use of simulcast relates to the latter approach, where it is more <t>The use of simulcast relates to the latter approach, where it is more
important to reduce the load on the RTP Mixer and/or minimize QoE impact important to reduce the load on the RTP mixer and/or minimize QoE impact
than to achieve an optimal adaptation of resource usage.</t> than to achieve an optimal adaptation of resource usage.</t>
<section anchor="sec-diverse-receivers" numbered="true" toc="default">
<section anchor="sec-diverse-receivers" <name>Reaching a Diverse Set of Receivers</name>
title="Reaching a Diverse Set of Receivers">
<t>The media sources provided by a sending participant potentially <t>The media sources provided by a sending participant potentially
need to reach several receiving participants that differ in terms of need to reach several receiving participants that differ in terms of
available resources. The receiver resources that typically differ available resources. The receiver resources that typically differ
include, but are not limited to:<list style="hanging"> include, but are not limited to:</t>
<t hangText="Codec:">This includes codec type (such as RTP payload <dl newline="false" spacing="normal">
<dt>Codec:</dt>
<dd>This includes codec type (such as RTP payload
format MIME type) and can include codec configuration. A couple of format MIME type) and can include codec configuration. A couple of
codec resources that differ only in codec configuration will be codec resources that differ only in codec configuration will be
"different" if they are somehow not "compatible", like if they "different" if they are somehow not "compatible", such as if they
differ in video codec profile, or the transport packetization differ in video codec profile or the transport packetization
configuration.</t> configuration.</dd>
<dt>Sampling:</dt>
<t hangText="Sampling:">This relates to how the media source is <dd>This relates to how the media source is
sampled, in spatial as well as in temporal domain. For video sampled, in spatial as well as temporal domain. For video
streams, spatial sampling affects image resolution and temporal streams, spatial sampling affects image resolution, and temporal
sampling affects video frame rate. For audio, spatial sampling sampling affects video frame rate. For audio, spatial sampling
relates to the number of audio channels and temporal sampling relates to the number of audio channels, and temporal sampling
affects audio bandwidth. This may be used to suit different affects audio bandwidth. This may be used to suit different
rendering capabilities or needs at the receiving endpoints.</t> rendering capabilities or needs at the receiving endpoints.</dd>
<dt>Bitrate:</dt>
<t hangText="Bitrate:">This relates to the number of bits sent per <dd>This relates to the number of bits sent per
second to transmit the media source as an RTP stream, which second to transmit the media source as an RTP stream, which
typically also affects the QoE for the receiving user.</t> typically also affects the QoE for the receiving user.</dd>
</list>Letting the sending participant create a simulcast of a few </dl>
<t>Letting the sending participant create a simulcast of a few
differently configured RTP streams per media source can be a good differently configured RTP streams per media source can be a good
tradeoff when using an RTP switch as middlebox, instead of sending a trade-off when using an RTP switch as middlebox, instead of sending a
single RTP stream and using an RTP mixer to create individual single RTP stream and using an RTP mixer to create individual
transcodings to each receiving participant.</t> transcodings to each receiving participant.</t>
<t>This requires that the receiving participants can be categorized in <t>This requires that the receiving participants can be categorized in
terms of available resources and that the sending participant can terms of available resources and that the sending participant can
choose a matching configuration for a single RTP stream per category choose a matching configuration for a single RTP stream per category
and media source. For example, a set of receiving participants differ and media source. For example, a set of receiving participants differ
only in screen resolution; some are able to display video with at most only in screen resolution; some are able to display video with at most
360p resolution and some support 720p resolution. A sending 360p resolution, and some support 720p resolution. A sending
participant can then reach all receivers with best possible resolution participant can then reach all receivers with best possible resolution
by creating a simulcast of RTP streams with 360p and 720p resolution by creating a simulcast of RTP streams with 360p and 720p resolution
for each sent video media source.</t> for each sent video media source.</t>
<t>The maximum number of simulcasted RTP streams that can be sent is <t>The maximum number of simulcasted RTP streams that can be sent is
mainly limited by the amount of processing and uplink network mainly limited by the amount of processing and uplink network
resources available to the sending participant.</t> resources available to the sending participant.</t>
</section> </section>
<section anchor="sec-application-specific" numbered="true" toc="default">
<section anchor="sec-application-specific" <name>Application-Specific Media Source Handling</name>
title="Application Specific Media Source Handling">
<t>The application logic that controls the communication session may <t>The application logic that controls the communication session may
include special handling of some media sources. It is, for example, include special handling of some media sources. It is, for example,
commonly the case that the media from a sending participant is not commonly the case that the media from a sending participant is not
sent back to itself.</t> sent back to itself.</t>
<t>It is also common that a currently active speaker participant is <t>It is also common that a currently active speaker participant is
shown in larger size or higher quality than other participants (the shown in larger size or higher quality than other participants (the
sampling or bitrate aspects of <xref target="sec-diverse-receivers"/>) sampling or bitrate aspects of <xref target="sec-diverse-receivers"
format="default"/>)
in a receiving client. Many conferencing systems do not send the in a receiving client. Many conferencing systems do not send the
active speaker's media back to the sender itself, which means there is active speaker's media back to the sender itself, which means there is
some other participant's media that instead is forwarded to the active some other participant's media that instead is forwarded to the active
speaker; typically the previous active speaker. This way, the speaker -- typically the previous active speaker. This way, the
previously active speaker is needed both in larger size (to current previously active speaker is needed both in larger size (to current
active speaker) and in small size (to the rest of the participants), active speaker) and in small size (to the rest of the participants),
which can be solved with a simulcast from the previously active which can be solved with a simulcast from the previously active
speaker to the RTP switch.</t> speaker to the RTP switch.</t>
</section> </section>
<section anchor="sec-receiver-preferences" numbered="true" toc="default">
<section anchor="sec-receiver-preferences" <name>Receiver Media-Source Preferences</name>
title="Receiver Media Source Preferences">
<t>The application logic that controls the communication session may <t>The application logic that controls the communication session may
allow receiving participants to state preferences on the allow receiving participants to state preferences on the
characteristics of the RTP stream they like to receive, for example in characteristics of the RTP stream they like to receive, for example in
terms of the aspects listed in <xref target="sec-diverse-receivers"/>. terms of the aspects listed in <xref target="sec-diverse-receivers" form at="default"/>.
Sending a simulcast of RTP streams is one way of accommodating Sending a simulcast of RTP streams is one way of accommodating
receivers with conflicting or otherwise incompatible preferences.</t> receivers with conflicting or otherwise incompatible preferences.</t>
</section> </section>
</section> </section>
<section anchor="sec-overview" numbered="true" toc="default">
<section anchor="sec-overview" title="Overview"> <name>Overview</name>
<t>This memo defines <xref target="RFC4566">SDP</xref> signaling that <t>This memo defines <xref target="RFC4566" format="default">SDP</xref> si
gnaling that
covers the above described simulcast use cases and functionalities. A covers the above described simulcast use cases and functionalities. A
number of requirements for such signaling are elaborated in <xref number of requirements for such signaling are elaborated in <xref target="
target="sec-requirements"/>.</t> sec-requirements" format="default"/>.</t>
<t>The RID mechanism, as defined in <xref <t>The Restriction Identifier (RID) mechanism, as defined in <xref target=
target="I-D.ietf-mmusic-rid"/>, enables an SDP offerer or answerer to "RFC8851" format="default"/>, enables an SDP offerer or answerer to
specify a number of different RTP stream restrictions for a rid-id by specify a number of different RTP stream restrictions for a rid-id by
using the "a=rid" line. Examples of such restrictions are maximum using the "a=rid" line. Examples of such restrictions are maximum
bitrate, maximum spatial video resolution (width and height), maximum bitrate, maximum spatial video resolution (width and height), maximum
video framerate, etc. Each rid-id may also be restricted to use only a video frame rate, etc. Each rid-id may also be restricted to use only a
subset of the RTP payload types in the associated SDP media description. subset of the RTP payload types in the associated SDP media description.
Those RTP payload types can have their own configurations and parameters Those RTP payload types can have their own configurations and parameters
affecting what can be sent or received, using the "a=fmtp" line as well affecting what can be sent or received, using the "a=fmtp" line as well
as other SDP attributes.</t> as other SDP attributes.</t>
<t>A new SDP media-level attribute, "a=simulcast", is defined. The
<t>A new SDP media level attribute "a=simulcast" is defined. The attribute describes, independently for "send" and "receive" directions, th
attribute describes, independently for send and receive directions, the e
number of simulcast RTP streams as well as potential alternative formats number of simulcast RTP streams as well as potential alternative formats
for each simulcast RTP stream. Each simulcast RTP stream, including for each simulcast RTP stream. Each simulcast RTP stream, including
alternatives, is identified using the RID identifier (rid-id), defined alternatives, is identified using the RID identifier (rid-id), defined
in <xref target="I-D.ietf-mmusic-rid"/>.</t> in <xref target="RFC8851" format="default"/>.</t>
<!-- DO NOT EDIT -->
<figure align="left"> <sourcecode type="sdp">
<artwork align="left"><![CDATA[a=simulcast:send 1;2,3 recv 4 a=simulcast:send 1;2,3 recv 4
]]></artwork> </sourcecode>
</figure> <!-- End DNE -->
<t>If this line is included in an SDP offer, the "send" part
<t>If the above line is included in an SDP offer, the "send" part
indicates the offerer's capability and proposal to send two simulcast indicates the offerer's capability and proposal to send two simulcast
RTP streams. Each simulcast stream is described by one or more RTP RTP streams. Each simulcast stream is described by one or more RTP
stream identifiers (rid-id), each group of rid-ids for a simulcast stream identifiers (rid-ids), and each group of rid-ids for a simulcast
stream is separated by a semicolon (";"). When a simulcast stream has stream is separated by a semicolon (";"). When a simulcast stream has
multiple rid-ids that are separated by a comma (","), they describe multiple rid-ids that are separated by a comma (","), they describe
alternative representations for that particular simulcast RTP stream. alternative representations for that particular simulcast RTP stream.
Thus, the above "send" part is interpreted as an intention to send two Thus, the "send" part shown above is interpreted as an intention to send t wo
simulcast RTP streams. The first simulcast RTP stream is identified and simulcast RTP streams. The first simulcast RTP stream is identified and
restricted according to rid-id 1. The second simulcast RTP stream can be restricted according to rid-id 1. The second simulcast RTP stream can be
sent as two alternatives, identified and restricted according to rid-ids sent as two alternatives, identified and restricted according to rid-ids
2 and 3. The "recv" part of the above line indicates that the offerer 2 and 3. The "recv" part of the line shown here indicates that the offerer
desires to receive a single RTP stream (no simulcast) according to desires to receive a single RTP stream (no simulcast) according to
rid-id 4.</t> rid-id 4.</t>
<t>A more complete example SDP-offer media description is provided
in <xref target="fig-md-offer" format="default"/>.</t>
<!-- DO NOT EDIT -->
<t>A more complete example SDP offer media description is provided <figure anchor="fig-md-offer">
below:</t> <name>Example Simulcast Media Description in Offer</name>
<sourcecode type="sdp">
<figure align="center" anchor="fig-md-offer"
title="Example Simulcast Media Description in Offer">
<artwork align="left"><![CDATA[
m=video 49300 RTP/AVP 97 98 99 m=video 49300 RTP/AVP 97 98 99
a=rtpmap:97 H264/90000 a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000 a=rtpmap:98 H264/90000
a=rtpmap:99 VP8/90000 a=rtpmap:99 VP8/90000
a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000
a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600
a=fmtp:99 max-fs=240; max-fr=30 a=fmtp:99 max-fs=240; max-fr=30
a=rid:1 send pt=97;max-width=1280;max-height=720 a=rid:1 send pt=97;max-width=1280;max-height=720
a=rid:2 send pt=98;max-width=320;max-height=180 a=rid:2 send pt=98;max-width=320;max-height=180
a=rid:3 send pt=99;max-width=320;max-height=180 a=rid:3 send pt=99;max-width=320;max-height=180
a=rid:4 recv pt=97 a=rid:4 recv pt=97
a=simulcast:send 1;2,3 recv 4 a=simulcast:send 1;2,3 recv 4
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
<t>The above SDP media description can be interpreted at a high level to <t>The SDP media description in <xref target="fig-md-offer" format="defaul
say that the offerer is capable of sending two simulcast RTP streams, t"/> can be
interpreted at a high level to
say that the offerer is capable of sending two simulcast RTP streams:
one H.264 encoded stream in up to 720p resolution, and one additional one H.264 encoded stream in up to 720p resolution, and one additional
stream encoded as either H.264 or VP8 with a maximum resolution of stream encoded as either H.264 or VP8 with a maximum resolution of
320x180 pixels. The offerer can receive one H.264 stream with maximum 320x180 pixels. The offerer can receive one H.264 stream with maximum
720p resolution.</t> 720p resolution.</t>
<t>The receiver of this SDP offer can generate an SDP answer that <t>The receiver of this SDP offer can generate an SDP answer that
indicates what it accepts. It uses the "a=simulcast" attribute to indicates what it accepts. It uses the "a=simulcast" attribute to
indicate simulcast capability and specify what simulcast RTP streams and indicate simulcast capability and specify what simulcast RTP streams and
alternatives to receive and/or send. An example of such answering alternatives to receive and/or send. An example of such an answering
"a=simulcast" attribute, corresponding to the above offer, is:</t> "a=simulcast" attribute, corresponding to the above offer, is:</t>
<!-- DO NOT EDIT -->
<figure align="left"> <sourcecode type="sdp">
<artwork align="left"><![CDATA[a=simulcast:recv 1;2 send 4 a=simulcast:recv 1;2 send 4
]]></artwork> </sourcecode>
</figure> <!-- End DNE -->
<t>With this SDP answer, the answerer indicates in the "recv" part that <t>With this SDP answer, the answerer indicates in the "recv" part that
it wants to receive the two simulcast RTP streams. It has removed an it wants to receive the two simulcast RTP streams. It has removed an
alternative that it doesn't support (rid-id 3). The send part confirms alternative that it doesn't support (rid-id 3). The "send" part confirms
to the offerer that it will receive one stream for this media source to the offerer that it will receive one stream for this media source
according to rid-id 4. The corresponding, more complete example SDP according to rid-id 4. The corresponding, more complete example SDP
answer media description could look like:</t> answer media description could look like <xref target="fig-md-answer" form
at="default"/>.</t>
<figure align="center" anchor="fig-md-answer" <!-- DO NOT EDIT -->
title="Example Simulcast Media Description in Answer"> <figure anchor="fig-md-answer">
<artwork align="left"><![CDATA[ <name>Example Simulcast Media Description in Answer</name>
<sourcecode type="sdp">
m=video 49674 RTP/AVP 97 98 m=video 49674 RTP/AVP 97 98
a=rtpmap:97 H264/90000 a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000 a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000
a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600
a=rid:1 recv pt=97;max-width=1280;max-height=720 a=rid:1 recv pt=97;max-width=1280;max-height=720
a=rid:2 recv pt=98;max-width=320;max-height=180 a=rid:2 recv pt=98;max-width=320;max-height=180
a=rid:4 send pt=97 a=rid:4 send pt=97
a=simulcast:recv 1;2 send 4 a=simulcast:recv 1;2 send 4
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
<t>It is assumed that a single SDP media description is used to describe <t>It is assumed that a single SDP media description is used to describe
a single media source. This is aligned with the concepts defined in a single media source. This is aligned with the concepts defined in
<xref target="RFC7656"/> and will work in a WebRTC context, both with <xref target="RFC7656" format="default"/> and will work in a WebRTC contex
and without <xref t, both with
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref> grouping and without BUNDLE grouping of media descriptions <xref target="RFC8843" f
of media descriptions.</t> ormat="default"/>.</t>
<t>To summarize, the "a=simulcast" line describes "send"- and
<t>To summarize, the "a=simulcast" line describes send and receive "receive"-direction simulcast streams separately. Each direction can in
direction simulcast streams separately. Each direction can in turn turn describe one or more simulcast streams, separated by semicolons. The
describe one or more simulcast streams, separated by semicolon. The
identifiers describing simulcast streams on the "a=simulcast" line are identifiers describing simulcast streams on the "a=simulcast" line are
rid-id, as defined by "a=rid" lines in <xref rid-ids, as defined by "a=rid" lines in <xref target="RFC8851" format="def
target="I-D.ietf-mmusic-rid"/>. Each simulcast stream can be offered as ault"/>. Each simulcast stream can be offered as
a list of alternative rid-id, with each alternative separated by comma a list of alternative rid-ids, with each alternative separated by a comma
(not in the examples above). A detailed specification can be found in as shown in the example offer in <xref target="fig-md-offer"/>. A detailed
<xref target="sec-details"/> and more detailed examples are outlined in specification can be found in
<xref target="sec-ex"/>.</t> <xref target="sec-details" format="default"/>, and more detailed examples
are outlined in
<xref target="sec-ex" format="default"/>.</t>
</section> </section>
<section anchor="sec-details" numbered="true" toc="default">
<section anchor="sec-details" title="Detailed Description"> <name>Detailed Description</name>
<t>This section further details the overview <xref <t>This section provides further details to the overview in <xref target="
target="sec-overview">above</xref>. First, formal syntax is <xref sec-overview" format="default"/>. First, formal syntax is <xref target="sec-attr
target="sec-attr">provided</xref>, followed by the rest of the SDP " format="default">provided</xref>, followed by the rest of the SDP
attribute definition in <xref target="sec-cap"/>. <xref attribute definition in <xref target="sec-cap" format="default"/>. <xref t
target="sec-relating">Relating Simulcast Streams </xref> provides the arget="sec-relating" format="default">"Relating Simulcast Streams"</xref> provid
definition of the RTP/RTCP mechanisms used. The section is concluded es the
definition of the RTP/RTCP mechanisms used. The section concludes
with a number of examples.</t> with a number of examples.</t>
<section anchor="sec-attr" numbered="true" toc="default">
<section anchor="sec-attr" title="Simulcast Attribute"> <name>Simulcast Attribute</name>
<t>This document defines a new SDP media-level "a=simulcast" <t>This document defines a new SDP media-level "a=simulcast"
attribute, with value according to the following <xref attribute, with value according to the syntax in <xref target="fig-abnf"
target="RFC5234">ABNF</xref> syntax and its update for <xref format="default"/>, which uses <xref target="RFC5234" format="default">ABNF</xr
target="RFC7405">Case-Sensitive String Support in ABNF</xref>:</t> ef> and its update, <xref target="RFC7405" format="default">"Case-Sensitive Stri
ng Support in ABNF"</xref>:</t>
<figure align="center" anchor="fig-abnf" <!-- DO NOT EDIT -->
title="ABNF for Simulcast Value"> <figure anchor="fig-abnf">
<artwork align="center"><![CDATA[ <name>ABNF for Simulcast Value</name>
<sourcecode type="abnf">
sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] ) sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] )
sc-send = %s"send" SP sc-str-list sc-send = %s"send" SP sc-str-list
sc-recv = %s"recv" SP sc-str-list sc-recv = %s"recv" SP sc-str-list
sc-str-list = sc-alt-list *( ";" sc-alt-list ) sc-str-list = sc-alt-list *( ";" sc-alt-list )
sc-alt-list = sc-id *( "," sc-id ) sc-alt-list = sc-id *( "," sc-id )
sc-id-paused = "~" sc-id-paused = "~"
sc-id = [sc-id-paused] rid-id sc-id = [sc-id-paused] rid-id
; SP defined in [RFC5234] ; SP defined in [RFC5234]
; rid-id defined in [I-D.ietf-mmusic-rid] ; rid-id defined in [RFC8851]
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
<t><list style="empty">
<t>Note to RFC Editor: Replace "I-D.ietf-mmusic-rid" in the above
figure with RFC number of draft-ietf-mmusic-rid before publication
of this document.</t>
</list></t>
<t>The "a=simulcast" attribute has a parameter in the form of one or <t>The "a=simulcast" attribute has a parameter in the form of one or
two simulcast stream descriptions, each consisting of a direction two simulcast stream descriptions, each consisting of a direction
("send" or "recv"), followed by a list of one or more simulcast ("send" or "recv"), followed by a list of one or more simulcast
streams. Each simulcast stream consists of one or more alternative streams. Each simulcast stream consists of one or more alternative
simulcast formats. Each simulcast format is identified by a simulcast simulcast formats. Each simulcast format is identified by a simulcast
stream identifier (rid-id). The rid-id MUST have the form of an RTP stream identifier (rid-id). The rid-id <bcp14>MUST</bcp14> have the form
stream identifier, as described by <xref of an RTP
target="I-D.ietf-mmusic-rid">RTP Payload Format stream identifier, as described by <xref target="RFC8851" format="defaul
Restrictions</xref>.</t> t">"RTP Payload Format Restrictions"</xref>.</t>
<t>In the list of simulcast streams, each simulcast stream is <t>In the list of simulcast streams, each simulcast stream is
separated by a semicolon (";"). Each simulcast stream can in turn be separated by a semicolon (";"). Each simulcast stream can, in turn, be
offered in one or more alternative formats, represented by rid-ids, offered in one or more alternative formats, represented by rid-ids,
separated by a comma (","). Each rid-id can also be specified as separated by commas (","). Each rid-id can also be specified as
initially <xref target="RFC7728">paused</xref>, indicated by initially <xref target="RFC7728" format="default">paused</xref>, indicat
ed by
prepending a "~" to the rid-id. The reason to allow separate initial prepending a "~" to the rid-id. The reason to allow separate initial
pause states for each rid-id is that pause capability can be specified pause states for each rid-id is that pause capability can be specified
individually for each RTP payload type referenced by an rid-id. Since individually for each RTP payload type referenced by a rid-id. Since
pause capability specified via the "a=rtcp-fb" attribute applies only pause capability specified via the "a=rtcp-fb" attribute applies only
to specified payload types and rid-id specified by "a=rid" can refer to specified payload types, and a rid-id specified by "a=rid" can refer
to multiple different payload types, it is unfeasible to pause streams to multiple different payload types, it is unfeasible to pause streams
with rid-id where any of the related RTP payload type(s) do not have with rid-id where any of the related RTP payload type(s) do not have
pause capability.</t> pause capability.</t>
</section> </section>
<section anchor="sec-cap" numbered="true" toc="default">
<section anchor="sec-cap" title="Simulcast Capability"> <name>Simulcast Capability</name>
<t>Simulcast capability is expressed through a new media level <xref <t>Simulcast capability is expressed through a new media-level <xref tar
target="sec-attr">SDP attribute, "a=simulcast"</xref>. The use of this get="sec-attr" format="default">SDP attribute, "a=simulcast"</xref>. The use of
this
attribute at the session level is undefined. Implementations of this attribute at the session level is undefined. Implementations of this
specification MUST NOT use it at the session level and MUST ignore it specification <bcp14>MUST NOT</bcp14> use it at the session level and <b cp14>MUST</bcp14> ignore it
if received at the session level. Extensions to this specification may if received at the session level. Extensions to this specification may
define such session level usage. Each SDP media description MUST define such session-level usage. Each SDP media description <bcp14>MUST< /bcp14>
contain at most one "a=simulcast" line.</t> contain at most one "a=simulcast" line.</t>
<t>There are separate and independent sets of simulcast streams in the
<t>There are separate and independent sets of simulcast streams in "send" and "receive" directions. When listing multiple directions, each
send and receive directions. When listing multiple directions, each direction <bcp14>MUST NOT</bcp14> occur more than once on the same line.
direction MUST NOT occur more than once on the same line.</t> </t>
<t>Simulcast streams using undefined rid-ids <bcp14>MUST NOT</bcp14> be
<t>Simulcast streams using undefined rid-id MUST NOT be used as valid used as valid
simulcast streams by an RTP stream receiver. The direction for an simulcast streams by an RTP stream receiver. The direction for a
rid-id MUST be aligned with the direction specified for the rid-id <bcp14>MUST</bcp14> be aligned with the direction specified for t
he
corresponding RTP stream identifier on the "a=rid" line.</t> corresponding RTP stream identifier on the "a=rid" line.</t>
<t>The listed number of simulcast streams for a direction sets a limit <t>The listed number of simulcast streams for a direction sets a limit
to the number of supported simulcast streams in that direction. The to the number of supported simulcast streams in that direction. The
order of the listed simulcast streams in the "send" direction suggests order of the listed simulcast streams in the "send" direction suggests
a proposed order of preference, in decreasing order: the rid-id listed a proposed order of preference, in decreasing order: the rid-id listed
first is the most preferred and subsequent streams have progressively first is the most preferred, and subsequent streams have progressively
lower preference. The order of the listed rid-id in the "recv" lower preference. The order of the listed rid-ids in the "recv"
direction expresses which simulcast streams that are preferred, with direction expresses which simulcast streams are preferred, with
the leftmost being most preferred. This can be of importance if the the leftmost being most preferred. This can be of importance if the
number of actually sent simulcast streams have to be reduced for some number of actually sent simulcast streams has to be reduced for some
reason.</t> reason.</t>
<t>rid-id that have explicit <xref <t>rid-ids that have explicit <xref target="RFC5583"
target="RFC5583">dependencies</xref> <xref format="default">dependencies</xref> <xref target="RFC8851"
target="I-D.ietf-mmusic-rid"/> to other rid-id (even in the same media format="default"/> to other rid-ids (even in the same media
description) MAY be used.</t> description) <bcp14>MAY</bcp14> be used.</t>
<t>Use of more than a single, alternative simulcast format for a <t>Use of more than a single, alternative simulcast format for a
simulcast stream MAY be specified as part of the attribute parameters simulcast stream <bcp14>MAY</bcp14> be specified as part of the
by expressing the simulcast stream as a comma-separated list of attribute parameters by expressing the simulcast stream as a
alternative rid-id. The order of the rid-id alternatives within a comma-separated list of alternative rid-ids. The order of the rid-id
simulcast stream is significant; the rid-id alternatives are listed alternatives within a simulcast stream is significant; the rid-id
from (left) most preferred to (right) least preferred. For the use of alternatives are listed from (left) most preferred to (right) least
simulcast, this overrides the normal codec preference as expressed by preferred. For the use of simulcast, this overrides the normal codec
format type ordering on the "m=" line, using regular SDP rules. This preference as expressed by format-type ordering on the "m=" line,
is to enable a separation of general codec preferences and simulcast using regular SDP rules. This is to enable a separation of general
stream configuration preferences. However, the choice of which codec preferences and simulcast-stream configuration
alternative to use per simulcast stream is independent, and there is preferences. However, the choice of which alternative to use per
currently no mechanism to align the choice between alternative rid-ids simulcast stream is independent, and there is currently no mechanism
between different simulcast streams.</t> for the offerer to force the answerer to choose the same alternative
for multiple simulcast streams.
</t>
<t>A simulcast stream can use a codec defined such that the same RTP <t>A simulcast stream can use a codec defined such that the same RTP
SSRC can change RTP payload type multiple times during a session, synchronization source (SSRC) can change RTP payload type multiple
possibly even on a per-packet basis. A typical example can be a speech times during a session, possibly even on a per-packet basis. A typical
codec that makes use of <xref target="RFC3389">Comfort Noise</xref> example is a speech codec that makes use of formats for <xref
and/or <xref target="RFC4733">DTMF</xref> formats.</t> target="RFC3389" format="default">Comfort Noise</xref> and/or <xref
target="RFC4733" format="default">dual-tone multifrequency
(DTMF)</xref>.</t>
<t>If <xref target="RFC7728">RTP stream pause/resume</xref> is <t>If <xref target="RFC7728" format="default">RTP stream
supported, any rid-id MAY be prefixed by a "~" character to indicate pause/resume</xref> is supported, any rid-id <bcp14>MAY</bcp14> be
that the corresponding simulcast stream is initially paused already prefixed by a "~" character to indicate that the corresponding
from start of the RTP session. In this case, support for RTP stream simulcast stream is paused already from the start of the RTP
pause/resume MUST also be included under the same "m=" line where session. In this case, support for RTP stream pause/resume
<bcp14>MUST</bcp14> also be included under the same "m=" line where
"a=simulcast" is included. All RTP payload types related to such an "a=simulcast" is included. All RTP payload types related to such an
initially paused simulcast stream MUST be listed in the SDP as initially paused simulcast stream <bcp14>MUST</bcp14> be listed in the
pause/resume capable as specified by <xref target="RFC7728"/>, e.g. by SDP as pause/resume capable as specified by <xref target="RFC7728"
using the "*" wildcard format for "a=rtcp-fb".</t> format="default"/> -- e.g., by using the "*" wildcard format for
"a=rtcp-fb".</t>
<t>An initially paused simulcast stream in "send" direction for the <t>An initially paused simulcast stream in the "send" direction for the
endpoint sending the SDP MUST be considered equivalent to an endpoint sending the SDP <bcp14>MUST</bcp14> be considered equivalent to
unsolicited locally paused stream, and be handled accordingly. an
unsolicited locally paused stream and handled accordingly.
Initially paused simulcast streams are resumed as described by the RTP Initially paused simulcast streams are resumed as described by the RTP
pause/resume specification. An RTP stream receiver that wishes to pause/resume specification. An RTP stream receiver that wishes to
resume an unsolicited locally paused stream needs to know the SSRC of resume an unsolicited locally paused stream needs to know the SSRC of
that stream. The SSRC of an initially paused simulcast stream can be that stream.
obtained from an RTP stream sender RTCP Sender Report (SR) including
both the desired SSRC as "SSRC of sender", and the rid-id value in an
<xref target="I-D.ietf-avtext-rid">RtpStreamId RTCP SDES
item</xref>.</t>
<t>If the endpoint sending the SDP includes an "recv" direction The SSRC of an initially paused simulcast stream can be obtained from
an RTP stream sender RTCP Sender Report (SR) or Receiver Report (RR)
that includes both the desired SSRC as initial SSRC in the source
description (SDES) chunk, optionally a <xref target="RFC8843"
format="default">MID SDES item</xref> (if used and if rid-ids are not
unique across "m=" lines), and the rid-id value in an <xref
target="RFC8852" format="default">RtpStreamId RTCP SDES
item</xref>.</t>
<t>If the endpoint sending the SDP includes a "recv"-direction
simulcast stream that is initially paused, then the remote RTP sender simulcast stream that is initially paused, then the remote RTP sender
receiving the SDP SHOULD put its RTP stream in a unsolicited locally receiving the SDP <bcp14>SHOULD</bcp14> put its RTP stream in an unsolic ited locally
paused state. The simulcast stream sender does not put the stream in paused state. The simulcast stream sender does not put the stream in
the locally paused state if there are other RTP stream receivers in the locally paused state if there are other RTP stream receivers in
the session that do not mark the simulcast stream as initially paused. the session that do not mark the simulcast stream as initially paused.
However, in centralized conferencing the RTP sender usually does not However, in centralized conferencing, the RTP sender usually does not
see the SDP signalling from RTP receivers and cannot make this see the SDP signaling from RTP receivers and cannot make this
determination. The reason to require an initially paused "recv" stream determination. The reason for requiring that an initially paused "recv"
to be considered locally paused by the remote RTP sender, instead of stream
making it equivalent to implicitly sending a pause request, is because be considered locally paused by the remote RTP sender instead of
making it equivalent to implicitly sending a pause request is that
the pausing RTP sender cannot know which receiving SSRC owns the the pausing RTP sender cannot know which receiving SSRC owns the
restriction when Temporary Maximum Media Stream Bit Rate Request restriction when Temporary Maximum Media Stream Bit Rate Request
(TMMBR) and Temporary Maximum Media Stream Bit Rate Notification (TMMBR) and Temporary Maximum Media Stream Bit Rate Notification
(TMMBN) are used for pause/resume signaling (<xref (TMMBN) are used for pause/resume signaling (<xref target="RFC7728"
target="RFC7728">Section 5.6 of </xref>) since the RTP receiver's SSRC sectionFormat="of" section="5.6" />); this is because the RTP
in send direction is sometimes not yet known.</t> receiver's SSRC
in the "send" direction is sometimes not yet known.</t>
<t>Use of the <xref target="RFC2198">redundant audio data</xref> <t>Use of the redundant audio data format <xref target="RFC2198" format=
format could be seen as a form of simulcast for loss protection "default"/>
purposes, but is not considered conflicting with the mechanisms could be seen as a form of simulcast for loss-protection
described in this memo and MAY therefore be used as any other format. purposes, but it is not considered conflicting with the mechanisms
In this case the "red" format, rather than the carried formats, SHOULD described in this memo and <bcp14>MAY</bcp14> therefore be used as any o
ther format.
In this case, the "red" format, rather than the carried formats, <bcp14>
SHOULD</bcp14>
be the one to list as a simulcast stream on the "a=simulcast" be the one to list as a simulcast stream on the "a=simulcast"
line.</t> line.</t>
<t>The media formats and corresponding characteristics of simulcast <t>The media formats and corresponding characteristics of simulcast
streams SHOULD be chosen such that they are different, e.g. as streams <bcp14>SHOULD</bcp14> be chosen such that they are different -- e.g., as
different SDP formats with differing "a=rtpmap" and/or "a=fmtp" lines, different SDP formats with differing "a=rtpmap" and/or "a=fmtp" lines,
or as differently defined RTP payload format restrictions. If this or as differently defined RTP payload format restrictions. If this
difference is not required, it is RECOMMENDED to use <xref difference is not required, it is <bcp14>RECOMMENDED</bcp14> to use RTP
target="RFC7104">RTP duplication</xref> procedures instead of duplication
simulcast. To avoid complications in implementations, a single rid-id procedures <xref target="RFC7104" format="default"/> instead of simulcast
MUST NOT occur more than once per "a=simulcast" line. Note that this . To avoid
complications in implementations, a single rid-id
<bcp14>MUST NOT</bcp14> occur more than once per "a=simulcast" line. Not
e that this
does not eliminate use of simulcast as an RTP duplication mechanism, does not eliminate use of simulcast as an RTP duplication mechanism,
since it is possible to define multiple different rid-id that are since it is possible to define multiple different rid-ids that are
effectively equivalent.</t> effectively equivalent.</t>
</section> </section>
<section anchor="sec-offer-answer" numbered="true" toc="default">
<section anchor="sec-offer-answer" title="Offer/Answer Use"> <name>Offer/Answer Use</name>
<t><list style="empty"> <dl>
<t>Note: The inclusion of "a=simulcast" or the use of simulcast <dt>Note:</dt> <dd>The inclusion of "a=simulcast" or the use of simulcast
does not change any of the interpretation or Offer/Answer does not change any of the interpretation or Offer/Answer
procedures for other SDP attributes, like "a=fmtp" or "a=rid".</t> procedures for other SDP attributes, such as "a=fmtp" or "a=rid".</d
</list></t> d>
</dl>
<section title="Generating the Initial SDP Offer"> <section numbered="true" toc="default">
<t>An offerer wanting to use simulcast for a media description SHALL <name>Generating the Initial SDP Offer</name>
<t>An offerer wanting to use simulcast for a media description <bcp14>
SHALL</bcp14>
include one "a=simulcast" attribute in that media description in the include one "a=simulcast" attribute in that media description in the
offer. An offerer listing a set of receive simulcast streams and/or offer. An offerer listing a set of receive simulcast streams and/or
alternative formats as rid-id in the offer MUST be prepared to alternative formats as rid-ids in the offer <bcp14>MUST</bcp14> be pre pared to
receive RTP streams for any of those simulcast streams and/or receive RTP streams for any of those simulcast streams and/or
alternative formats from the answerer.</t> alternative formats from the answerer.</t>
</section> </section>
<section numbered="true" toc="default">
<section title="Creating the SDP Answer"> <name>Creating the SDP Answer</name>
<t>An answerer that does not understand the concept of simulcast <t>An answerer that does not understand the concept of simulcast
will also not know the attribute and will remove it in the SDP will also not know the attribute and will remove it in the SDP
answer, as defined in existing <xref target="RFC3264">SDP answer, as defined in existing SDP offer/answer procedures <xref targe
Offer/Answer</xref> procedures. Since SDP session level simulcast is t="RFC3264" format="default"/>. Since SDP session-level simulcast is
undefined in this memo, an answerer that receives an offer with the undefined in this memo, an answerer that receives an offer with the
"a=simulcast" attribute on SDP session level SHALL remove it in the "a=simulcast" attribute on the SDP session level <bcp14>SHALL</bcp14> remove it in the
answer. An answerer that understands the attribute but receives answer. An answerer that understands the attribute but receives
multiple "a=simulcast" attributes in the same media description multiple "a=simulcast" attributes in the same media description
SHALL disable use of simulcast by removing all "a=simulcast" lines <bcp14>SHALL</bcp14> disable use of simulcast by removing all "a=simul cast" lines
for that media description in the answer.</t> for that media description in the answer.</t>
<t>An answerer that does understand the attribute and wants to
<t>An answerer that does understand the attribute and that wants to support simulcast in an indicated direction <bcp14>SHALL</bcp14> rever
support simulcast in an indicated direction SHALL reverse se
directionality of the unidirectional direction parameters; "send" directionality of the unidirectional direction parameters -- "send"
becomes "recv" and vice versa, and include it in the answer.</t> becomes "recv" and vice versa -- and include it in the answer.</t>
<t>An answerer that receives an offer with simulcast containing an <t>An answerer that receives an offer with simulcast containing an
"a=simulcast" attribute listing alternative rid-id MAY keep all the "a=simulcast" attribute listing alternative rid-ids <bcp14>MAY</bcp14>
alternative rid-id in the answer, but it MAY also choose to remove keep all the
any non-desirable alternative rid-id in the answer. The answerer alternative rid-ids in the answer, but it <bcp14>MAY</bcp14> also choo
MUST NOT add any alternative rid-id in send direction in the answer se to remove
any nondesirable alternative rid-ids in the answer. The answerer
<bcp14>MUST NOT</bcp14> add any alternative rid-ids in the "send" dire
ction in the answer
that were not present in the offer receive direction. The answerer that were not present in the offer receive direction. The answerer
MUST be prepared to receive any of the receive direction rid-id <bcp14>MUST</bcp14> be prepared to receive any of the receive-directio
alternatives and MAY send any of the send direction alternatives n rid-id
alternatives and <bcp14>MAY</bcp14> send any of the "send"-direction a
lternatives
that are part of the answer.</t> that are part of the answer.</t>
<t>An answerer that receives an offer with simulcast that lists a <t>An answerer that receives an offer with simulcast that lists a
number of simulcast streams, MAY reduce the number of simulcast number of simulcast streams <bcp14>MAY</bcp14> reduce the number of si
streams in the answer, but MUST NOT add simulcast streams.</t> mulcast
streams in the answer, but it <bcp14>MUST NOT</bcp14> add simulcast st
reams.</t>
<t>An answerer that receives an offer without RTP stream <t>An answerer that receives an offer without RTP stream
pause/resume capability MUST NOT mark any simulcast streams as pause/resume capability <bcp14>MUST NOT</bcp14> mark any simulcast str eams as
initially paused in the answer.</t> initially paused in the answer.</t>
<t>An RTP stream answerer capable of pause/resume that receives an
<t>An RTP stream pause/resume capable answerer that receives an offer with RTP stream pause/resume capability <bcp14>MAY</bcp14> mark
offer with RTP stream pause/resume capability MAY mark any rid-id any rid-ids
that refer to pause/resume capable formats as initially paused in that refer to pause/resume capable formats as initially paused in
the answer.</t> the answer.</t>
<t>An answerer that receives indication in an offer of a rid-id
<t>An answerer that receives indication in an offer of an rid-id being initially paused <bcp14>SHOULD</bcp14> mark that rid-id as initi
being initially paused SHOULD mark that rid-id as initially paused ally paused
also in the answer, regardless of direction, unless it has good also in the answer, regardless of direction, unless it has good
reason for the rid-id not being initially paused. One reason to reason for the rid-id not being initially paused. One reason to
remove an initial pause in the answer compared to the offer could, remove an initial pause in the answer compared to the offer could be,
for example, be that all receive direction simulcast streams for a for example, that all "receive"-direction simulcast streams for a
media source the answerer accepts in the answer would otherwise be media source the answerer accepts in the answer would otherwise be
paused.</t> paused.</t>
</section> </section>
<section numbered="true" toc="default">
<section title="Offerer Processing the SDP Answer"> <name>Offerer Processing the SDP Answer</name>
<t>An offerer that receives an answer without "a=simulcast" MUST NOT <t>An offerer that receives an answer without "a=simulcast" <bcp14>MUS
T NOT</bcp14>
use simulcast towards the answerer. An offerer that receives an use simulcast towards the answerer. An offerer that receives an
answer with "a=simulcast" without any rid-id in a specified answer with "a=simulcast" without any rid-id in a specified
direction MUST NOT use simulcast in that direction.</t> direction <bcp14>MUST NOT</bcp14> use simulcast in that direction.</t>
<t>An offerer that receives an answer where some rid-id alternatives <t>An offerer that receives an answer where some rid-id alternatives
are kept MUST be prepared to receive any of the kept send direction are kept <bcp14>MUST</bcp14> be prepared to receive any of the kept "s
rid-id alternatives, and MAY send any of the kept receive direction end"-direction
rid-id alternatives and <bcp14>MAY</bcp14> send any of the kept "recei
ve"-direction
rid-id alternatives.</t> rid-id alternatives.</t>
<t>An offerer that receives an answer where some of the rid-ids are
<t>An offerer that receives an answer where some of the rid-id are removed compared to the offer <bcp14>MAY</bcp14> release the correspon
removed compared to the offer MAY release the corresponding ding
resources (codec, transport, etc) in its receive direction and MUST resources (codec, transport, etc) in its "receive" direction and <bcp1
NOT send any RTP packets corresponding to the removed rid-id.</t> 4>MUST
NOT</bcp14> send any RTP packets corresponding to the removed rid-ids.
<t>An offerer that offered some of its rid-id as initially paused </t>
and that receives an answer that does not indicate RTP stream <t>An offerer that offered some of its rid-ids as initially paused
pause/resume capability, MUST NOT initially pause any simulcast and receives an answer that does not indicate RTP stream
pause/resume capability <bcp14>MUST NOT</bcp14> initially pause any si
mulcast
streams.</t> streams.</t>
<t>An offerer with RTP stream pause/resume capability that receives <t>An offerer with RTP stream pause/resume capability that receives
an answer where some rid-id are marked as initially paused, SHOULD an answer where some rid-ids are marked as initially paused <bcp14>SHO
initially pause those RTP streams regardless if they were marked as ULD</bcp14>
initially pause those RTP streams, even if they were marked as
initially paused also in the offer, unless it has good reason for initially paused also in the offer, unless it has good reason for
those RTP streams not being initially paused. One such reason could, those RTP streams not being initially paused. One such reason could be
for example, be that the answerer would otherwise initially not ,
for example, that the answerer would otherwise initially not
receive any media of that type at all.</t> receive any media of that type at all.</t>
</section> </section>
<section numbered="true" toc="default">
<section title="Modifying the Session"> <name>Modifying the Session</name>
<t>Offers inside an existing session follow the same rules as for <t>Offers inside an existing session follow the same rules as for
initial SDP offer, with these additions:<list style="numbers"> initial SDP offer, with these additions:</t>
<t>rid-id marked as initially paused in the offerer's send <ol spacing="normal" type="1">
direction SHALL reflect the offerer's opinion of the current <li>rid-ids marked as initially paused in the offerer's "send"
direction <bcp14>SHALL</bcp14> reflect the offerer's opinion of th
e current
pause state at the time of creating the offer. This is purely pause state at the time of creating the offer. This is purely
informational, and <xref target="RFC7728">RTP stream informational, and RTP stream
pause/resume</xref> signaling in the ongoing session SHALL take pause/resume signaling <xref target="RFC7728" format="default"/> i
precedence in case of any conflict or ambiguity.</t> n the ongoing
session <bcp14>SHALL</bcp14> take precedence in case of any conflic
<t>rid-id marked as initially paused in the offerer's receive t or
direction SHALL (as in an initial offer) reflect the offerer's ambiguity.</li>
<li>rid-ids marked as initially paused in the offerer's "receive"
direction <bcp14>SHALL</bcp14> (as in an initial offer) reflect th
e offerer's
desired rid-id pause state. Except for the case where the desired rid-id pause state. Except for the case where the
offerer already paused the corresponding RTP stream through offerer already paused the corresponding RTP stream through
<xref target="RFC7728">RTP stream pause/resume</xref> signaling <xref target="RFC7728" format="default">RTP stream pause/resume</x
, this is identical to the conditions at an initial offer.</t> ref> signaling,
</list></t> this is identical to the conditions at an initial offer.</li>
</ol>
<t>Creation of SDP answers and processing of SDP answers inside an <t>Creation of SDP answers and processing of SDP answers inside an
existing session follow the same rules as described above for existing session follow the same rules as described above for
initial SDP offer/answer.</t> initial SDP offer/answer.</t>
<t>Session modification restrictions in <xref
<t>Session modification restrictions in section 6.5 of <xref target="RFC8851" sectionFormat="of" section="6.5">"RTP Payload Format
target="I-D.ietf-mmusic-rid">RTP payload format restrictions</xref> Restrictions"</xref>
also apply.</t> also apply.</t>
</section> </section>
</section> </section>
<section numbered="true" toc="default">
<section title="Use with Declarative SDP"> <name>Use with Declarative SDP</name>
<t>This document does not define the use of "a=simulcast" in <t>This document does not define the use of "a=simulcast" in
declarative SDP, partly motivated by use of the <xref declarative SDP, partly because use of the <xref target="RFC8851" format
target="I-D.ietf-mmusic-rid">simulcast format identification</xref> ="default">simulcast format identification</xref>
not being defined for use in declarative SDP. If concrete use cases is not defined for use in declarative SDP. If concrete use cases
for simulcast in declarative SDP are identified in the future, the for simulcast in declarative SDP are identified in the future, the
authors of this memo expect that additional specifications will authors of this memo expect that additional specifications will
address such use.</t> address such use.</t>
</section> </section>
<section anchor="sec-relating" numbered="true" toc="default">
<section anchor="sec-relating" title="Relating Simulcast Streams"> <name>Relating Simulcast Streams</name>
<t>Simulcast RTP streams MUST be related on RTP level through <xref <t>Simulcast RTP streams <bcp14>MUST</bcp14> be related on the RTP
target="I-D.ietf-avtext-rid">RtpStreamId</xref>, as specified in the level through <xref target="RFC8852"
SDP <xref target="sec-cap">"a=simulcast" attribute </xref> parameters. format="default">RtpStreamId</xref>, as specified in the
SDP <xref target="sec-cap" format="default">"a=simulcast" attribute
</xref> parameters.
This is sufficient as long as there is only a single media source per This is sufficient as long as there is only a single media source per
SDP media description. When using <xref SDP media description. When using <xref target="RFC8843" format="default
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref>, where ">BUNDLE</xref>, where
multiple SDP media descriptions jointly specify a single RTP session, multiple SDP media descriptions jointly specify a single RTP session,
the SDES MID identification mechanism in BUNDLE allows relating RTP the SDES MID (Media Identification) mechanism in BUNDLE allows relating
streams back to individual media descriptions, after which the above RTP
described RtpStreamId relations can be used. Use of the <xref streams back to individual media descriptions, after which the
target="RFC8285">RTP header extension</xref> for both MID and RtpStreamId relations described above can be used.
RtpStreamId identifications can be important to ensure rapid initial
reception, required to correctly interpret and process the RTP
streams. Implementers of this specification MUST support the RTCP
source description (SDES) item method and SHOULD support RTP header
extension method to signal RtpStreamId on RTP level.<list
style="hanging">
<t hangText="NOTE:">For the case where it is clear from SDP that
RTP PT uniquely maps to corresponding RtpStreamId, an RTP receiver
can use RTP PT to relate simulcast streams. This can sometimes
enable decoding even in advance to receiving RtpStreamId
information in RTCP SDES and/or RTP header extensions.</t>
</list></t>
<t>RTP streams MUST only use a single alternative rid-id at a time Use of the RTP header extension for the <xref target="RFC7941">RTCP
(based on RTP timestamps), but MAY change format (and rid-id) on a source description items</xref> for both MID
per-RTP packet basis. This corresponds to the existing (non-simulcast) and RtpStreamId identifications can be important to ensure rapid
initial reception, required to correctly interpret and process the RTP
streams. Implementers of this specification <bcp14>MUST</bcp14>
support the RTCP source description (SDES) item method and
<bcp14>SHOULD</bcp14> support RTP header extension method to signal
RtpStreamId on the RTP level.</t>
<dl newline="false" spacing="normal">
<dt>NOTE:</dt>
<dd>For the case where it is clear from SDP that the
RTP PT uniquely maps to a corresponding RtpStreamId, an RTP receiver
can use RTP PT to relate simulcast streams. This can sometimes
enable decoding even in advance of receiving RtpStreamId
information in RTCP SDES and/or RTP header extensions.</dd>
</dl>
<t>RTP streams <bcp14>MUST</bcp14> only use a single alternative rid-id
at a time
(based on RTP timestamps) but <bcp14>MAY</bcp14> change format (and rid-
id) on a
per-RTP packet basis. This corresponds to the existing (nonsimulcast)
SDP offer/answer case when multiple formats are included on the "m=" SDP offer/answer case when multiple formats are included on the "m="
line in the SDP answer, enabling per-RTP packet change of RTP payload line in the SDP answer, enabling per-RTP packet change of RTP payload
type.</t> type.</t>
</section> </section>
<section anchor="sec-ex" numbered="true" toc="default">
<section anchor="sec-ex" title="Signaling Examples"> <name>Signaling Examples</name>
<t>These examples describe a client to video conference service, using <t>These examples describe a client-to-video-conference service, using
a centralized media topology with an RTP mixer.</t> a centralized media topology with an RTP mixer.</t>
<!-- DO NOT EDIT -->
<figure align="center" anchor="fig-mixer-four-party" <figure anchor="fig-mixer-four-party">
title="Four-party Mixer-based Conference"> <name>Four-Party Mixer-Based Conference</name>
<artwork align="center"><![CDATA[ <artwork align="center" name="" type="" alt=""><![CDATA[
+---+ +-----------+ +---+ +---+ +-----------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| F |<---->| |<---->| J | | F |<---->| |<---->| J |
+---+ +-----------+ +---+]]></artwork> +---+ +-----------+ +---+]]></artwork>
</figure> </figure>
<!-- End DNE -->
<section anchor="sec-ex-single-source" title="Single-Source Client"> <section anchor="sec-ex-single-source" numbered="true" toc="default">
<name>Single-Source Client</name>
<t>Alice is calling in to the mixer with a simulcast-enabled client <t>Alice is calling in to the mixer with a simulcast-enabled client
capable of a single media source per media type. The client can send capable of a single media source per media type. The client can send
a simulcast of 2 video resolutions and frame rates: HD 1280x720p a simulcast of 2 video resolutions and frame rates: HD 1280x720p
30fps and thumbnail 320x180p 15fps. This is defined below using the 30fps and thumbnail 320x180p 15fps. This is defined below using the
<xref target="RFC6236">"imageattr"</xref>. In this example, only the <xref target="RFC6236" format="default">"imageattr"</xref>. In this ex
"pt" "a=rid" parameter is used, effectively achieving a 1:1 mapping ample, only the
between RtpStreamId and media formats (RTP payload types), to "pt" "a=rid" parameter is used to
describe simulcast stream formats. Alice's Offer:</t> describe simulcast stream formats, effectively achieving a 1:1 mapping
between RtpStreamId and media formats (RTP payload types). Alice's Off
er:</t>
<figure align="center" anchor="fig-up-offer" <figure anchor="fig-up-offer">
title="Single-Source Simulcast Offer"> <name>Single-Source Simulcast Offer</name>
<artwork align="left"><![CDATA[ <sourcecode type="sdp">
v=0 v=0
o=alice 2362969037 2362969040 IN IP4 192.0.2.156 o=alice 2362969037 2362969040 IN IP4 192.0.2.156
s=Simulcast Enabled Client s=Simulcast-Enabled Client
c=IN IP4 192.0.2.156 c=IN IP4 192.0.2.156
t=0 0 t=0 0
m=audio 49200 RTP/AVP 0 m=audio 49200 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
m=video 49300 RTP/AVP 97 98 m=video 49300 RTP/AVP 97 98
a=rtpmap:97 H264/90000 a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000 a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000
a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=rid:1 send pt=97 a=rid:1 send pt=97
a=rid:2 send pt=98 a=rid:2 send pt=98
a=rid:3 recv pt=97 a=rid:3 recv pt=97
a=simulcast:send 1;2 recv 3 a=simulcast:send 1;2 recv 3
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
<t>The only thing in the SDP that indicates simulcast capability is <t>The only thing in the SDP that indicates simulcast capability is
the line in the video media description containing the "simulcast" the line in the video media description containing the "simulcast"
attribute. The included "a=fmtp" and "a=imageattr" parameters attribute. The included "a=fmtp" and "a=imageattr" parameters
indicates that sent simulcast streams can differ in video indicate that sent simulcast streams can differ in video
resolution. The RTP header extension for RtpStreamId is offered to resolution. The RTP header extension for RtpStreamId is offered to
avoid issues with the initial binding between RTP streams (SSRCs) avoid issues with the initial binding between RTP streams (SSRCs)
and the RtpStreamId identifying the simulcast stream and its and the RtpStreamId identifying the simulcast stream and its
format.</t> format.</t>
<t>The answer from the server indicates that it, too, is simulcast
<t>The Answer from the server indicates that it too is simulcast
capable. Should it not have been simulcast capable, the capable. Should it not have been simulcast capable, the
"a=simulcast" line would not have been present and communication "a=simulcast" line would not have been present, and communication
would have started with the media negotiated in the SDP. Also the would have started with the media negotiated in the SDP. Also, the
usage of the RtpStreamId RTP header extension is accepted.</t> usage of the RtpStreamId RTP header extension is accepted.</t>
<!-- DO NOT EDIT -->
<figure align="center" anchor="fig-up-answer" <figure anchor="fig-up-answer">
title="Single-Source Simulcast Answer"> <name>Single-Source Simulcast Answer</name>
<artwork align="left"><![CDATA[ <sourcecode type="sdp">
v=0 v=0
o=server 823479283 1209384938 IN IP4 192.0.2.2 o=server 823479283 1209384938 IN IP4 192.0.2.2
s=Answer to Simulcast Enabled Client s=Answer to Simulcast-Enabled Client
c=IN IP4 192.0.2.43 c=IN IP4 192.0.2.43
t=0 0 t=0 0
m=audio 49672 RTP/AVP 0 m=audio 49672 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
m=video 49674 RTP/AVP 97 98 m=video 49674 RTP/AVP 97 98
a=rtpmap:97 H264/90000 a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000 a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000
a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=rid:1 recv pt=97 a=rid:1 recv pt=97
a=rid:2 recv pt=98 a=rid:2 recv pt=98
a=rid:3 send pt=97 a=rid:3 send pt=97
a=simulcast:recv 1;2 send 3 a=simulcast:recv 1;2 send 3
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
]]></artwork> </sourcecode></figure>
</figure> <!-- End DNE -->
<t>Since the server is the simulcast media receiver, it reverses the <t>Since the server is the simulcast media receiver, it reverses the
direction of the "simulcast" and "rid" attribute parameters.</t> direction of the "simulcast" and "rid" attribute parameters.</t>
</section> </section>
<section anchor="sec-ex-multi-source" numbered="true" toc="default">
<section anchor="sec-ex-multi-source" title="Multi-Source Client"> <name>Multisource Client</name>
<t>Fred is calling in to the same conference as in the example above <t>Fred is calling in to the same conference as in the example above
with a two-camera, two-display system, thus capable of handling two with a two-camera, two-display system, thus capable of handling two
separate media sources in each direction, where each media source is separate media sources in each direction, where each media source is
simulcast-enabled in the send direction. Fred's client is restricted simulcast enabled in the "send" direction. Fred's client is restricted
to a single media source per media description.</t> to a single media source per media description.</t>
<t>The first two simulcast streams for the first media source use <t>The first two simulcast streams for the first media source use
different codecs, <xref target="RFC6190">H264-SVC</xref> and <xref different codecs, <xref target="RFC6190" format="default">H264-SVC</xr
target="RFC6184">H264</xref>. These two simulcast streams also have ef> and <xref target="RFC6184" format="default">H264</xref>. These two simulcast
a temporal dependency. Two different video codecs, <xref streams also have
target="RFC7741">VP8</xref> and H264, are offered as alternatives a temporal dependency. Two different video codecs, <xref target="RFC77
41" format="default">VP8</xref> and H264, are offered as alternatives
for the third simulcast stream for the first media source. Only the for the third simulcast stream for the first media source. Only the
highest fidelity simulcast stream is sent from start, the lower highest-fidelity simulcast stream is sent from start, the
fidelity streams being initially paused.</t> lower-fidelity streams being initially paused.</t>
<t>The second media source is offered with three different simulcast <t>The second media source is offered with three different simulcast
streams. All video streams of this second media source are loss streams. All video streams of this second media source are loss
protected by <xref target="RFC4588">RTP retransmission</xref>. Also protected by <xref target="RFC4588" format="default">RTP retransmissio
here, all but the highest fidelity simulcast stream are initially n</xref>. In
paused. Note that the lower resolution is more prioritized than the addition, all but the highest-fidelity simulcast stream are
medium resolution simulcast stream.</t> initially paused. Note that the lower resolution is more prioritized
than the medium-resolution simulcast stream.</t>
<t>Fred's client is also using BUNDLE to send all RTP streams from <t>Fred's client is also using BUNDLE to send all RTP streams from
all media descriptions in the same RTP session on a single media all media descriptions in the same RTP session on a single media
transport. Although using many different simulcast streams in this transport. Although using many different simulcast streams in this
example, the use of RtpStreamId as simulcast stream identification example, the use of RtpStreamId as simulcast stream identification
enables use of a low number of RTP payload types. Note that the use enables use of a low number of RTP payload types.
of both <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref> and
<xref target="I-D.ietf-mmusic-rid">"a=rid"</xref> recommends using
the <xref target="RFC8285">RTP header extension</xref> for carrying
these RTP stream identification fields, which is consequently also
included in the SDP. Note also that for "a=rid", the corresponding
RtpStreamId SDES attribute RTP header extension is named <xref
target="I-D.ietf-avtext-rid">rtp-stream-id</xref>.</t>
<figure anchor="fig-ms-offer" Note that when using both <xref target="RFC8843"
title="Fred's Multi-Source Simulcast Offer"> format="default">BUNDLE</xref> and <xref target="RFC8851"
<artwork><![CDATA[ format="default">"a=rid"</xref>, it is recommended to use the RTP
header extension for the <xref target="RFC7941" format="default">RTCP
source descriptions items</xref> for carrying
these RTP stream-identification fields, which is consequently also
included in the SDP.
Note also that for "a=rid",
the corresponding RtpStreamId SDES attribute RTP header extension is
named <xref target="RFC8852"
format="default">rtp-stream-id</xref>.</t>
<!-- DO NOT EDIT -->
<figure anchor="fig-ms-offer">
<name>Fred's Multisource Simulcast Offer</name>
<sourcecode type="sdp">
v=0 v=0
o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d
s=Offer from Simulcast Enabled Multi-Source Client s=Offer from Simulcast-Enabled Multi-Source Client
c=IN IP6 2001:db8::c000:27d c=IN IP6 2001:db8::c000:27d
t=0 0 t=0 0
a=group:BUNDLE foo bar zen a=group:BUNDLE foo bar zen
m=audio 49200 RTP/AVP 99 m=audio 49200 RTP/AVP 99
a=mid:foo a=mid:foo
a=rtpmap:99 G722/8000 a=rtpmap:99 G722/8000
m=video 49600 RTP/AVPF 100 101 103 m=video 49600 RTP/AVPF 100 101 103
a=mid:bar a=mid:bar
a=rtpmap:100 H264-SVC/90000 a=rtpmap:100 H264-SVC/90000
a=rtpmap:101 H264/90000 a=rtpmap:101 H264/90000
skipping to change at line 979 skipping to change at line 926
a=rtpmap:104 rtx/90000 a=rtpmap:104 rtx/90000
a=fmtp:104 apt=96;rtx-time=200 a=fmtp:104 apt=96;rtx-time=200
a=rid:1 send max-fs=921600;max-fps=30 a=rid:1 send max-fs=921600;max-fps=30
a=rid:2 send max-fs=614400;max-fps=15 a=rid:2 send max-fs=614400;max-fps=15
a=rid:3 send max-fs=230400;max-fps=30 a=rid:3 send max-fs=230400;max-fps=30
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=rtcp-fb:* ccm pause nowait a=rtcp-fb:* ccm pause nowait
a=simulcast:send 1;~3;~2 a=simulcast:send 1;~3;~2
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
</section> </section>
<section numbered="true" toc="default">
<section title="Simulcast and Redundancy"> <name>Simulcast and Redundancy</name>
<t>The example in this section looks at applying simulcast with <t>The example in this section looks at applying simulcast with
audio and video redundancy formats. The audio media description uses audio and video redundancy formats.
codec and bitrate restrictions, combining it with <xref
target="RFC2198">RTP Payload for Redundant Audio Data</xref> for
enhanced packet loss resilience. The video media description applies
both resolution and bitrate restrictions, combining it with FEC in
the form of <xref
target="I-D.ietf-payload-flexible-fec-scheme">Flexible FEC</xref>
and <xref target="RFC4588">RTP Retransmission</xref>.</t>
<t>The audio source is offered to be sent as two simulcast streams. The audio media description uses codec and bitrate restrictions,
The first simulcast stream is encoded with Opus, restricted to 50 combined with the <xref target="RFC2198" format="default">RTP
kbps (rid-id=5), and the second simulcast stream is encoded either payload for redundant audio data</xref> for enhanced packet-loss
with G.711 (rid-id=7) or with G.711 combined with LPC for redundancy resilience. The video media description applies both resolution and
(rid-id=6). In this example, stand-alone LPC is not offered as an bitrate restrictions, combined with Forward Error Correction (FEC)
possible payload type for the second simulcast stream's RID, which in the form of <xref target="RFC8627" format="default">flexible
could e.g. be motivated by not providing sufficient quality.</t> FEC</xref> and <xref target="RFC4588" format="default">RTP
retransmission</xref>.</t>
<t>
The audio source is offered to be sent as two simulcast
streams. The first simulcast stream is encoded with Opus,
restricted to 64 kbps (rid-id=1), and the second simulcast stream
(rid-id=2) is encoded with either G.711, or G.711 combined with
linear predictive coding (LPC) for redundancy and explicit comfort
noise (CN). Both simulcast streams include telephone-event
capability. In this example, stand-alone LPC is not offered as a
possible payload type for the second simulcast stream's RID, which
could be motivated by, for example, not providing sufficient
quality.
</t>
<t>The video source is offered to be sent as two simulcast streams, <t>The video source is offered to be sent as two simulcast streams,
both with two alternative simulcast formats. Redundancy and repair both with two alternative simulcast formats. Redundancy and repair
are offered in the form of both Flexible FEC and RTP Retransmission. are offered in the form of both flexible FEC and RTP retransmission.
The Flexible FEC is not bound to any particular RTP streams and is The flexible FEC is not bound to any particular RTP streams and is
therefore possible to use across all RTP streams that are being sent therefore able to be used across all RTP streams that are being sent
as part of this media description.</t> as part of this media description.</t>
<!-- DO NOT EDIT -->
<figure anchor="fig-sim-red" <figure anchor="fig-sim-red">
title="Simulcast and Redundancy Example"> <name>Simulcast and Redundancy Example</name>
<artwork><![CDATA[v=0 <sourcecode type="sdp">
o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d
s=Offer from Simulcast Enabled Client using Redundancy s=Offer from Simulcast-Enabled Client using Redundancy
c=IN IP6 2001:db8::c000:27d c=IN IP6 2001:db8::c000:27d
t=0 0 t=0 0
a=group:BUNDLE foo bar a=group:BUNDLE foo bar
m=audio 49200 RTP/AVP 97 98 99 100 101 102 m=audio 49200 RTP/AVP 97 98 99 100 101 102
a=mid:foo a=mid:foo
a=rtpmap:97 G711/8000 a=rtpmap:97 G711/8000
a=rtpmap:98 LPC/8000 a=rtpmap:98 LPC/8000
a=rtpmap:99 OPUS/48000/1 a=rtpmap:99 OPUS/48000/1
a=rtpmap:100 RED/8000/1 a=rtpmap:100 RED/8000/1
a=rtpmap:101 CN/8000 a=rtpmap:101 CN/8000
skipping to change at line 1046 skipping to change at line 1001
a=mid:bar a=mid:bar
a=rtpmap:103 H264/90000 a=rtpmap:103 H264/90000
a=rtpmap:104 VP8/90000 a=rtpmap:104 VP8/90000
a=rtpmap:105 rtx/90000 a=rtpmap:105 rtx/90000
a=rtpmap:106 rtx/90000 a=rtpmap:106 rtx/90000
a=rtpmap:107 flexfec/90000 a=rtpmap:107 flexfec/90000
a=fmtp:103 profile-level-id=42c00d;max-fs=3600;max-mbps=108000 a=fmtp:103 profile-level-id=42c00d;max-fs=3600;max-mbps=108000
a=fmtp:104 max-fs=3600; max-fr=30 a=fmtp:104 max-fs=3600; max-fr=30
a=fmtp:105 apt=103;rtx-time=200 a=fmtp:105 apt=103;rtx-time=200
a=fmtp:106 apt=104;rtx-time=200 a=fmtp:106 apt=104;rtx-time=200
a=fmtp:107 repair-window=2000 a=fmtp:107 repair-window=100000
a=rid:1 send pt=103;max-width=1280;max-height=720;max-fps=30 a=rid:1 send pt=103;max-width=1280;max-height=720;max-fps=30
a=rid:2 send pt=104;max-width=1280;max-height=720;max-fps=30 a=rid:2 send pt=104;max-width=1280;max-height=720;max-fps=30
a=rid:3 send pt=103;max-width=640;max-height=360;max-br=300000 a=rid:3 send pt=103;max-width=640;max-height=360;max-br=300000
a=rid:4 send pt=104;max-width=640;max-height=360;max-br=300000 a=rid:4 send pt=104;max-width=640;max-height=360;max-br=300000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=rtcp-fb:* ccm pause nowait a=rtcp-fb:* ccm pause nowait
a=simulcast:send 1,2;3,4 a=simulcast:send 1,2;3,4
]]></artwork> </sourcecode>
</figure> </figure>
<!-- End DNE -->
<t/>
</section> </section>
</section> </section>
</section> </section>
<section anchor="sec-rtp-aspects" numbered="true" toc="default">
<section anchor="sec-rtp-aspects" title="RTP Aspects"> <name>RTP Aspects</name>
<t>This section discusses what the different entities in a simulcast <t>This section discusses what the different entities in a simulcast
media path can expect to happen on RTP level. This is explored from media path can expect to happen on the RTP level. This is explored from
source to sink by starting in an endpoint with a media source that is source to sink by starting in an endpoint with a media source that is
simulcasted to an RTP middlebox. That RTP middlebox sends media sources simulcasted to an RTP middlebox. That RTP middlebox sends media sources
both to other RTP middleboxes (cascaded middleboxes), as well as to other RTP middleboxes (cascaded middleboxes), as well as
selecting some simulcast format of the media source and sending it to selecting some simulcast format of the media source and sending it to
receiving endpoints. Different types of RTP middleboxes and their usage receiving endpoints. Different types of RTP middleboxes and their usage
of the different simulcast formats results in several different of the different simulcast formats results in several different
behaviors.</t> behaviors.</t>
<section numbered="true" toc="default">
<section title="Outgoing from Endpoint with Media Source"> <name>Outgoing from Endpoint with Media Source</name>
<t>The most straightforward simulcast case is the RTP streams being <t>The most straightforward simulcast case is the RTP streams being
emitted from the endpoint that originates a media source. When emitted from the endpoint that originates a media source. When
simulcast has been negotiated in the sending direction, the endpoint simulcast has been negotiated in the sending direction, the endpoint
can transmit up to the number of RTP streams needed for the negotiated can transmit up to the number of RTP streams needed for the negotiated
simulcast streams for that media source. Each RTP stream (SSRC) is simulcast streams for that media source. Each RTP stream (SSRC) is
identified by <xref target="sec-relating">associating</xref> it with identified by associating it (<xref target="sec-relating" format="defaul t"/>) with
an RtpStreamId SDES item, transmitted in RTCP and possibly also as an an RtpStreamId SDES item, transmitted in RTCP and possibly also as an
RTP header extension. In cases where multiple media sources have been RTP header extension. In cases where multiple media sources have been
negotiated for the same RTP session and thus <xref negotiated for the same RTP session and thus <xref target="RFC8843" form
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref> is used, at="default">BUNDLE</xref> is used, the MID SDES item will also be
also the MID SDES item will be sent similarly to the RtpStreamId.</t> sent, similarly to the RtpStreamId.</t>
<t>Each RTP stream might not be continuously transmitted due to any of <t>Each RTP stream might not be continuously transmitted due to any of
the following reasons; temporarily paused using <xref the following reasons: temporarily paused using <xref target="RFC7728" f
target="RFC7728">Pause/Resume</xref>, sender side application logic ormat="default">Pause/Resume</xref>, sender-side application logic
temporarily pausing it, or lack of network resources to transmit this temporarily pausing it, or lack of network resources to transmit this
simulcast stream. However, all simulcast streams that have been simulcast stream. However, all simulcast streams that have been
negotiated have active and maintained SSRC (at least in regular RTCP negotiated have active and maintained SSRCs (at least in regular RTCP
reports), even if no RTP packets are currently transmitted. The reports), even if no RTP packets are currently transmitted. The
relation between an RTP Stream (SSRC) and a particular simulcast relation between an RTP stream (SSRC) and a particular simulcast
stream is not expected to change, except in exceptional situations stream is not expected to change, except in exceptional situations
such as SSRC collisions. At SSRC changes, the usage of MID and such as SSRC collisions. At SSRC changes, the usage of MID and
RtpStreamId should enable the receiver to correctly identify the RTP RtpStreamId should enable the receiver to correctly identify the RTP
streams even after an SSRC change.</t> streams even after an SSRC change.</t>
</section> </section>
<section numbered="true" toc="default">
<section title="RTP Middlebox to Receiver"> <name>RTP Middlebox to Receiver</name>
<t>RTP streams in a multi-party RTP session can be used in multiple <t>RTP streams in a multiparty RTP session can be used in multiple
different ways, when the session utilizes simulcast at least on the different ways when the session utilizes simulcast at least on the
media source to middlebox legs. This is to a large degree due to the media-source-to-middlebox legs. This is to a large degree due to the
different RTP middlebox behaviors, but also the needs of the different RTP middlebox behaviors, but also the needs of the
application. This text assumes that the RTP middlebox will select a application. This text assumes that the RTP middlebox will select a
media source and choose which simulcast stream for that media source media source and choose which simulcast stream for that media source
to deliver to a specific receiver. In many cases, at most one to deliver to a specific receiver. In many cases, at most one
simulcast stream per media source will be forwarded to a particular simulcast stream per media source will be forwarded to a particular
receiver at any instant in time, even if the selected simulcast stream receiver at any instant in time, even if the selected simulcast stream
may vary. For cases where this does not hold due to application needs, may vary. For cases where this does not hold due to application needs,
then the RTP stream aspects will fall under the middlebox to middlebox the RTP stream aspects will fall under the middlebox-to-middlebox
case <xref target="sec-rtp-box-box"/>.</t> case (<xref target="sec-rtp-box-box" format="default"/>).</t>
<t>The selection of which simulcast streams to forward towards the <t>The selection of which simulcast streams to forward towards the
receiver, is application specific. However, in conferencing receiver is application specific. However, in conferencing
applications, active speaker selection is common. In case the number applications, active speaker selection is common. In case the number
of media sources possible to forward, N, is less than the total amount of media sources possible to forward, N, is less than the total number
of media sources available in an multi-media session, the current and of media sources available in a multimedia session, the current and
previous speakers (up to N in total) are often the ones forwarded. To previous speakers (up to N in total) are often the ones forwarded. To
avoid the need for media specific processing to determine the current avoid the need for media-specific processing to determine the current
speaker(s) in the RTP middlebox, the endpoint providing a media source speaker(s) in the RTP middlebox, the endpoint providing a media source
may include meta data, such as the <xref target="RFC6464">RTP Header may include metadata, such as the <xref target="RFC6464" format="default
Extension for Client-to-Mixer Audio Level Indication</xref>.</t> ">RTP header
extension for client-to-mixer audio level indication</xref>.</t>
<t>The possibilities for stream switching are media type specific, but <t>The possibilities for stream switching are media type specific, but
for media types with significant interframe dependencies in the for media types with significant interframe dependencies in the
encoding, like most video coding, the switching needs to be made at encoding, like most video coding, the switching needs to be made at
suitable switching points in the media stream that breaks or otherwise suitable switching points in the media stream that breaks or otherwise
deals with the dependency structure. Even if switching points can be deals with the dependency structure. Even if switching points can be
included periodically, it is common to use mechanisms like <xref included periodically, it is common to use mechanisms like <xref target=
target="RFC5104">Full Intra Requests</xref> to request switching "RFC5104" format="default">Full Intra Requests</xref> to request switching
points from the endpoint performing the encoding of the media points from the endpoint performing the encoding of the media
source.</t> source.</t>
<t>Inclusion of the RtpStreamId SDES item for an SSRC in the
<t>Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox middlebox-to-receiver direction should only occur when use of
to receiver direction should only occur when use of RtpStreamId has RtpStreamId has
been negotiated in that direction. It is worth noting that one can been negotiated in that direction. It is worth noting that one can
signal multiple RtpStreamIds when simulcast signalling indicates only signal multiple RtpStreamIds when simulcast signaling indicates only
a single simulcast stream, allowing one to use all of the RtpStreamIds a single simulcast stream, allowing one to use all of the RtpStreamIds
as alternatives for that simulcast stream. One reason for including as alternatives for that simulcast stream. One reason for including
the RtpStreamId in the middlebox to receiver direction for an RTP the RtpStreamId in the middlebox-to-receiver direction for an RTP
stream is to let the receiver know which restrictions apply to the stream is to let the receiver know which restrictions apply to the
currently delivered RTP stream. In case the RtpStreamId is negotiated currently delivered RTP stream. In case the RtpStreamId is negotiated
to be used, it is important to remember that the used identifiers will to be used, it is important to remember that the used identifiers will
be specific to each signalling session. Even if the central entity can be specific to each signaling session. Even if the central entity can
attempt to coordinate, it is likely that the RtpStreamIds need to be attempt to coordinate, it is likely that the RtpStreamIds need to be
translated to the leg specific values. The below cases will have as translated to the leg-specific values. The below cases will assume
base line that RtpStreamId is not used in the mixer to receiver that RtpStreamId is not used in the mixer to receiver
direction.</t> direction.</t>
<section numbered="true" toc="default">
<section title="Media-Switching Mixer"> <name>Media-Switching Mixer</name>
<t>This section discusses the behavior in cases where the RTP <t>This section discusses the behavior in cases where the RTP
middlebox behaves like the Media-Switching Mixer (Section 3.6.2) in middlebox behaves like the media-switching mixer in
<xref target="RFC7667">RTP Topologies</xref>. The fundamental aspect RTP topologies (<xref target="RFC7667"
sectionFormat="of" section="3.6.2"/>). The
fundamental aspect
here is that the media sources delivered from the middlebox will be here is that the media sources delivered from the middlebox will be
the mixer's conceptual or functional ones. For example, one media the mixer's conceptual or functional ones. For example, one media
source may be the main speaker in high resolution video, while a source may be the main speaker in high-resolution video, while a
number of other media sources are thumbnails of each number of other media sources are thumbnails of each
participant.</t> participant.</t>
<t>The above results in the RTP stream produced by the mixer being
<t>The above results in that the RTP stream produced by the mixer is
one that switches between a number of received incoming RTP streams one that switches between a number of received incoming RTP streams
for different media sources and in different simulcast versions. The for different media sources and in different simulcast versions. The
mixer selects the media source to be sent as one of the RTP streams, mixer selects the media source to be sent as one of the RTP streams
and then selects among the available simulcast streams for the most and then selects among the available simulcast streams for the most
appropriate one. The selection criteria include available bandwidth appropriate one. The selection criteria include available bandwidth
on the mixer to receiver path and restrictions based on the on the mixer-to-receiver path and restrictions based on the
functional usage of the RTP stream delivered to the receiver. As an functional usage of the RTP stream delivered to the receiver. As an
example of the latter, it is unnecessary to forward a full HD video example of the latter, it is unnecessary to forward a full HD video
to a receiver if the display area is just a thumbnail. Thus, to a receiver if the display area is just a thumbnail. Thus,
restrictions may exist to not allow some simulcast streams to be restrictions may exist to not allow some simulcast streams to be
forwarded for some of the mixer's media sources.</t> forwarded for some of the mixer's media sources.</t>
<t>This will result in a single RTP stream being used for each of <t>This will result in a single RTP stream being used for each of
the RTP mixer's media sources. This RTP stream is at any point in the RTP mixer's media sources. At any point in time, this RTP stream
time a selection of one particular RTP stream arriving to the mixer, is a selection of one particular RTP stream arriving to the mixer,
where the RTP header field values are rewritten to provide a where the RTP header-field values are rewritten to provide a
consistent, single RTP stream. If the RTP mixer doesn't receive any consistent, single RTP stream. If the RTP mixer doesn't receive any
incoming stream matched to this media source, the SSRC will not incoming stream matched to this media source, the SSRC will not
transmit, but be kept alive using RTCP. The SSRC and thus RTP stream transmit but be kept alive using RTCP. The SSRC and thus RTP stream
for the mixer's media source is expected to be long term stable. It for the mixer's media source is expected to be long-term stable. It
will only be changed by signalling or other disruptive events. Note will only be changed by signaling or other disruptive events. Note
that although the above talks about a single RTP stream, there can that although the above talks about a single RTP stream, there can
in some cases be multiple RTP streams carrying the selected in some cases be multiple RTP streams carrying the selected
simulcast stream for the originating media source, including simulcast stream for the originating media source, including
redundancy or other auxiliary RTP streams.</t> redundancy or other auxiliary RTP streams.</t>
<t>The mixer may communicate the identity of the originating media <t>The mixer may communicate the identity of the originating media
source to the receiver by including the CSRC field with the source to the receiver by including the Contributing Source (CSRC) fie ld with the
originating media source's SSRC value. Note that due to the originating media source's SSRC value. Note that due to the
possibility that the RTP mixer switches between simulcast versions possibility that the RTP mixer switches between simulcast versions
of the media source, the CSRC value may change, even if the media of the media source, the CSRC value may change, even if the media
source is kept the same.</t> source is kept the same.</t>
<t>It is important to note that any MID SDES item from the <t>It is important to note that any MID SDES item from the
originating media source needs to be removed and not be associated originating media source needs to be removed and not be associated
with the RTP stream's SSRC. That is, there is nothing in the with the RTP stream's SSRC. That is, there is nothing in the
signalling between the mixer and the receiver that is structured signaling between the mixer and the receiver that is structured
around the originating media sources, only the mixer's media around the originating media sources, only the mixer's media
sources. If they would be associated with the SSRC, the receiver sources. If they were associated with the SSRC, the receiver
would likely believe that there has been an SSRC collision, and that would likely believe that there has been an SSRC collision and
the RTP stream is spurious as it doesn't carry the identifiers used the RTP stream is spurious, because it doesn't carry the identifiers u
sed
to relate it to the correct context. However, this is not true for to relate it to the correct context. However, this is not true for
CSRC values, as long as they are never used as SSRC. In these cases CSRC values, as long as they are never used as an SSRC. In these cases ,
one could provide CNAME and MID as SDES items. A receiver could use one could provide CNAME and MID as SDES items. A receiver could use
this to determine which CSRC values that are associated with the this to determine which CSRC values that are associated with the
same originating media source.</t> same originating media source.</t>
<t>If RtpStreamIds are used in the scenario described by this <t>If RtpStreamIds are used in the scenario described by this
section, it should be noted that the RtpStreamId on a particular section, it should be noted that the RtpStreamId on a particular
SSRC will change based on the actual simulcast stream selected for SSRC will change based on the actual simulcast stream selected for
switching. These RtpStreamId identifiers will be local to this leg's switching. These RtpStreamId identifiers will be local to this leg's
signalling context. In addition, the defined RtpStreamIds and their signaling context. In addition, the defined RtpStreamIds and their
parameters need to cover all the media sources and simulcast streams parameters need to cover all the media sources and simulcast streams
received by the RTP mixer that can be switched into this media received by the RTP mixer that can be switched into this media
source, sent by the RTP mixer.</t> source, sent by the RTP mixer.</t>
</section> </section>
<section numbered="true" toc="default">
<section title="Selective Forwarding Middlebox"> <name>Selective Forwarding Middlebox</name>
<t>This section discusses the behavior in cases where the RTP <t>This section discusses the behavior in cases where the RTP
middlebox behaves like the Selective Forwarding Middlebox (Section middlebox behaves like the Selective Forwarding Middlebox in RTP
3.7) in <xref target="RFC7667">RTP Topologies</xref>. Applications topologies (<xref target="RFC7667"
for this type of RTP middlebox results in that each originating sectionFormat="of" section="3.7"/>). Applications
media source will have a corresponding media source on the leg for this type of RTP middlebox result in each originating
media source having a corresponding media source on the leg
between the middlebox and the receiver. A Selective Forwarding between the middlebox and the receiver. A Selective Forwarding
Middlebox (SFM) could go as far as exposing all the simulcast Middlebox (SFM) could go as far as exposing all the simulcast
streams for an media source, however this section will focus on streams for a media source; however, this section will focus on
having a single simulcast stream that can contain any of the having a single simulcast stream that can contain any of the
simulcast formats. This section will assume that the SFM projection simulcast formats. This section will assume that the SFM projection
mechanism works on media source level, and maps one of the media mechanism works on the media-source level and maps one of the media
source's simulcast streams onto one RTP stream from the SFM to the source's simulcast streams onto one RTP stream from the SFM to the
receiver.</t> receiver.</t>
<t>This usage will result in the individual RTP stream(s) for
<t>This usage will result in that the individual RTP stream(s) for one media source being able to switch between being active and
one media source can switch between being active to paused, based on paused, based on
the subset of media sources the SFM wants to provide the receiver the subset of media sources the SFM wants to provide the receiver
for the moment. With SFMs there exist no reasons to use CSRC to for the moment. With SFMs, there exist no reasons to use CSRC to
indicate the originating stream, as there is a one to one media indicate the originating stream, as there is a one-to-one
source mapping. If the application requires knowing the simulcast media-source mapping. If the application requires knowing the
simulcast
version received to function well, then RtpStreamId should be version received to function well, then RtpStreamId should be
negotiated on the SFM to receiver leg. Which simulcast stream that negotiated on the SFM to receiver leg. Which simulcast stream that
is being forwarded is not made explicit unless RtpStreamId is used is being forwarded is not made explicit unless RtpStreamId is used
on the leg.</t> on the leg.</t>
<t>Any MID SDES items being sent by the SFM to the receiver are only <t>Any MID SDES items being sent by the SFM to the receiver are only
those agreed between the SFM and the receiver, and no MID values those agreed between the SFM and the receiver, and no MID values
from the originating side of the SFM are to be forwarded.</t> from the originating side of the SFM are to be forwarded.</t>
<t>An SFM could expose corresponding RTP streams for all the media
<t>A SFM could expose corresponding RTP streams for all the media sources and their simulcast streams and then, for any media source
sources and their simulcast streams, and then for any media source that is to be provided, forward one selected simulcast stream.
that is to be provided forward one selected simulcast stream. However, this is not recommended, as it would unnecessarily increase
However, this is not recommended as it would unnecessarily increase
the number of RTP streams and require the receiver to timely detect the number of RTP streams and require the receiver to timely detect
switching between simulcast streams. The above usage requires the switching between simulcast streams. The above usage requires the
same SFM functionality for switching, while avoiding the same SFM functionality for switching, while avoiding the
uncertainties of timely detecting that a RTP stream ends. The uncertainties of timely detecting that a RTP stream ends. The
benefit would be that the received simulcast stream would be benefit would be that the received simulcast stream would be
implicitly provided by which RTP stream would be active for a media implicitly provided by which RTP stream would be active for a media
source. However, using RtpStreamId to make this explicit also source. However, using RtpStreamId to make this explicit also
exposes which alternative format is used. The conclusion is that exposes which alternative format is used. The conclusion is that
using one RTP stream per simulcast stream is unnecessary. The issue using one RTP stream per simulcast stream is unnecessary. The issue
with timely detecting end of streams, independent if they are with timely detecting end of streams, independent of whether they are
stopped temporarily or long term, is that there is no explicit stopped temporarily or long term, is that there is no explicit
indication that the transmission has intentionally been stopped. The indication that the transmission has intentionally been stopped. The
RTCP based <xref target="RFC7728">Pause and Resume mechanism</xref> RTCP-based <xref target="RFC7728" format="default">pause and resume
mechanism</xref>
includes a PAUSED indication that provides the last RTP sequence includes a PAUSED indication that provides the last RTP sequence
number transmitted prior to the pause. Due to usage, the timeliness number transmitted prior to the pause. Due to usage, the timeliness
of this solution depends on when delivery using RTCP can occur in of this solution depends on when delivery using RTCP can occur in
relation to the transmission of the last RTP packet. If no explicit relation to the transmission of the last RTP packet. If no explicit
information is provided at all, then detection based on non information is provided at all, then detection based on
increasing RTCP SR field values and timers need to be used to nonincreasing RTCP SR field values and timers need to be used to
determine pause in RTP packet delivery. This results in that one can determine pause in RTP packet delivery. As a result, when the last
usually not determine when the last RTP packet arrives (if it RTP packet arrives (if it arrives), one usually
arrives) that this will be the last. That it was the last is cannot determine that this will be the last. That it was the last is
something that one learns later.</t> something that one learns later.</t>
</section> </section>
</section> </section>
<section anchor="sec-rtp-box-box" numbered="true" toc="default">
<section anchor="sec-rtp-box-box" title="RTP Middlebox to RTP Middlebox"> <name>RTP Middlebox to RTP Middlebox</name>
<t>This relates to the transmission of simulcast streams between RTP <t>This relates to the transmission of simulcast streams between RTP
middleboxes or other usages where one wants to enable the delivery of middleboxes or other usages where one wants to enable the delivery of
multiple simultaneous simulcast streams per media source, but the multiple simultaneous simulcast streams per media source, but the
transmitting entity is not the originating endpoint. For a particular transmitting entity is not the originating endpoint. For a particular
direction between middlebox A and B, this looks very similar to the direction between middleboxes A and B, this looks very similar to the
originating to middlebox case on a media source basis. However, in originating-to-middlebox case on a media-source basis. However, in
this case there is usually multiple media sources, originating from this case, there are usually multiple media sources, originating from
multiple endpoints. This can create situations where limitations in multiple endpoints. This can create situations where limitations in
the number of simultaneously received media streams can arise, for the number of simultaneously received media streams can arise -- for
example due to limitation in network bandwidth. In this case, a subset example, due to limitation in network bandwidth. In this case, a subset
of not only the simulcast streams, but also media sources can be of not only the simulcast streams but also media sources can be
selected. This results in that individual RTP streams can be become selected. As a result, individual RTP streams can become
paused at any point and later being resumed based on various paused at any point and later be resumed based on various criteria.</t>
criteria.</t>
<t>The MIDs used between A and B are the ones agreed between these two <t>The MIDs used between A and B are the ones agreed between these two
identities in signalling. The RtpStreamId values will also be provided identities in signaling. The RtpStreamId values will also be provided
to ensure explicit information about which simulcast stream they are. to ensure explicit information about which simulcast stream they are.
The RTP stream to MID and RtpStreamId associations should here be long The RTP-stream-to-MID and -RtpStreamId associations should here be
term stable.</t> long-term stable.</t>
</section> </section>
</section> </section>
<section anchor="sec-network-aspects" numbered="true" toc="default">
<section anchor="sec-network-aspects" title="Network Aspects"> <name>Network Aspects</name>
<t>Simulcast is in this memo defined as the act of sending multiple <t>Simulcast is in this memo defined as the act of sending multiple
alternative encoded streams of the same underlying media source. When alternative encoded streams of the same underlying media
transmitting multiple independent streams that originate from the same source. Transmitting multiple independent streams that originate from
source, it could potentially be done in several different ways using the same
source could potentially be done in several different ways using
RTP. A general discussion on considerations for use of the different RTP RTP. A general discussion on considerations for use of the different RTP
multiplexing alternatives can be found in <xref multiplexing alternatives can be found in <xref target="I-D.ietf-avtcore-m
target="I-D.ietf-avtcore-multiplex-guidelines">Guidelines for ultiplex-guidelines" format="default">"Guidelines for using the Multiplexing Fea
Multiplexing in RTP</xref>. Discussion and clarification on how to tures of
handle multiple streams in an RTP session can be found in <xref RTP to Support Multiple Media Streams"</xref>. Discussion and
target="RFC8108"/>.</t> clarification on how to handle multiple streams in an RTP session can be
found in <xref target="RFC8108" format="default"/>.</t>
<t>The network aspects that are relevant for simulcast are:<list <t>The network aspects that are relevant for simulcast are:</t>
style="hanging"> <dl newline="false" spacing="normal">
<t hangText="Quality of Service:">When using simulcast it might be <dt>Quality of Service (QoS):</dt>
<dd>When using simulcast, it might be
of interest to prioritize a particular simulcast stream, rather than of interest to prioritize a particular simulcast stream, rather than
applying equal treatment to all streams. For example, lower bitrate applying equal treatment to all streams. For example, lower-bitrate
streams may be prioritized over higher bitrate streams to minimize streams may be prioritized over higher-bitrate streams to minimize
congestion or packet losses in the low bitrate streams. Thus, there congestion or packet losses in the low-bitrate streams. Thus, there
is a benefit to use a simulcast solution with good QoS support.</t> is a benefit to using a simulcast solution with good QoS support.</dd>
<t hangText="NAT/FW Traversal:">Using multiple RTP sessions incurs
more cost for NAT/FW traversal unless they can re-use the same
transport flow, which can be achieved by <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Multiplexing
Negotiation Using SDP Port Numbers</xref>.</t>
</list></t>
<dt>NAT/FW Traversal (Network Address Translator / Firewall Traversal):<
/dt>
<dd>Using multiple RTP sessions incurs
more cost for NAT/FW traversal unless they can reuse the same
transport flow, which can be achieved by <xref target="RFC8843" format
="default">multiplexing negotiation using SDP port
numbers</xref>.</dd>
</dl>
<t/> <t/>
<section numbered="true" toc="default">
<section title="Bitrate Adaptation"> <name>Bitrate Adaptation</name>
<t>Use of multiple simulcast streams can require a significant amount <t>Use of multiple simulcast streams can require a significant amount
of network resources. The aggregate bandwidth for all simulcast of network resources. The aggregate bandwidth for all simulcast
streams for a media source (and thus SDP media description) is bounded streams for a media source (and thus SDP media description) is bounded
by any SDP "b=" line applicable to that media source. It is assumed by any SDP "b=" line applicable to that media source. It is assumed
that a suitable congestion control mechanism is used by the that a suitable congestion-control mechanism is used by the
application to ensure that it doesn't cause persistent congestion. If application to ensure that it doesn't cause persistent congestion. If
the amount of available network resources varies during an RTP session the amount of available network resources varies during an RTP session
such that it does not match what is negotiated in SDP, the bitrate such that it does not match what is negotiated in SDP, the bitrate
used by the different simulcast streams may have to be reduced used by the different simulcast streams may have to be reduced
dynamically. When a simulcasting media source uses a single media dynamically. When a simulcasting media source uses a single media
transport for all of the simulcast streams, it is likely that a joint transport for all of the simulcast streams, it is likely that a joint
congestion control across all simulcast streams is used for that media congestion control across all simulcast streams is used for that media
source. What simulcast streams to prioritize when allocating available source. What simulcast streams to prioritize when allocating available
bitrate among the simulcast streams in such adaptation SHOULD be taken bitrate among the simulcast streams in such adaptation <bcp14>SHOULD</bc p14> be taken
from the simulcast stream order on the "a=simulcast" line and ordering from the simulcast stream order on the "a=simulcast" line and ordering
of alternative simulcast formats <xref target="sec-cap"/>. Simulcast of alternative simulcast formats (<xref target="sec-cap" format="default "/>). Simulcast
streams that have pause/resume capability and that would be given such streams that have pause/resume capability and that would be given such
low bitrate by the adaptation process that they are considered not low bitrate by the adaptation process that they are considered not
really useful can be temporarily paused until the limiting condition really useful can be temporarily paused until the limiting condition
clears.</t> clears.</t>
</section> </section>
</section> </section>
<section anchor="sec-limitation" numbered="true" toc="default">
<section anchor="sec-limitation" title="Limitation"> <name>Limitation</name>
<t>The chosen approach has a limitation that relates to the use of a <t>The chosen approach has a limitation that relates to the use of a
single RTP session for all simulcast formats of a media source, which single RTP session for all simulcast formats of a media source, which
comes from sending all simulcast streams related to a media source under comes from sending all simulcast streams related to a media source under
the same SDP media description.</t> the same SDP media description.</t>
<t>It is not possible to use different simulcast streams on different <t>It is not possible to use different simulcast streams on different
media transports, limiting the possibilities to apply different QoS to media transports, which limits the possibilities for applying different Qo S to
different simulcast streams. When using unicast, QoS mechanisms based on different simulcast streams. When using unicast, QoS mechanisms based on
individual packet marking are feasible, since they do not require individual packet marking are feasible, since they do not require
separation of simulcast streams into different RTP sessions to apply separation of simulcast streams into different RTP sessions to apply
different QoS.</t> different QoS.</t>
<t>It is also not possible to separate different simulcast streams into <t>It is also not possible to separate different simulcast streams into
different multicast groups to allow a multicast receiver to pick the different multicast groups to allow a multicast receiver to pick the
stream it wants, rather than receive all of them. In this case, the only stream it wants, rather than receive all of them. In this case, the only
reasonable implementation is to use different RTP sessions for each reasonable implementation is to use different RTP sessions for each
multicast group so that reporting and other RTCP functions operate as multicast group so that reporting and other RTCP functions operate as
intended. Such simulcast usage in multicast context is out of scope for intended. Such simulcast usage in a multicast context is out of scope for
the current document and would require additional specification.</t> the current document and would require additional specification.</t>
</section> </section>
<section anchor="sec-iana" numbered="true" toc="default">
<section anchor="sec-iana" title="IANA Considerations"> <name>IANA Considerations</name>
<t>This document requests to register a new media-level SDP attribute, <t>This document registers a new media-level SDP attribute,
"simulcast", in the "att-field (media level only)" registry within the "simulcast", in the "att-field (media level only)" registry within the
SDP parameters registry, according to the procedures of <xref "Session Description Protocol (SDP) Parameters" registry, according to the
target="RFC4566"/> and <xref procedures of <xref target="RFC4566" format="default"/> and <xref target="
target="I-D.ietf-mmusic-sdp-mux-attributes"/>.<list style="hanging"> RFC8859" format="default"/>.</t>
<t hangText="Contact name, email:">The IESG (iesg@ietf.org)</t> <dl newline="false" spacing="normal">
<dt>Contact name, email:</dt>
<t hangText="Attribute name:">simulcast</t> <dd>The IESG (iesg@ietf.org)</dd>
<dt>Attribute name:</dt>
<t hangText="Long-form attribute name:">Simulcast stream <dd>simulcast</dd>
description</t> <dt>Long-form attribute name:</dt>
<dd>Simulcast stream description</dd>
<t hangText="Charset dependent:">No</t> <dt>Charset dependent:</dt>
<dd>No</dd>
<t hangText="Attribute value:">sc-value; see <xref <dt>Attribute value:</dt>
target="sec-attr"/> of RFC XXXX.</t> <dd>sc-value; see <xref target="sec-attr" format="default"/> of RFC
8853.</dd>
<t hangText="Purpose:">Signals simulcast capability for a set of RTP <dt>Purpose:</dt>
streams</t> <dd>Signals simulcast capability for a set of RTP
streams</dd>
<t hangText="MUX category:">NORMAL</t> <dt>Mux category:</dt>
</list>Note to RFC Editor: Please replace "RFC XXXX" with the assigned <dd>NORMAL</dd>
number of this RFC.</t> </dl>
</section> </section>
<section anchor="sec-security" numbered="true" toc="default">
<section anchor="sec-security" title="Security Considerations"> <name>Security Considerations</name>
<t>The simulcast capability, configuration attributes, and parameters <t>The simulcast capability, configuration attributes, and parameters
are vulnerable to attacks in signaling.</t> are vulnerable to attacks in signaling.</t>
<t>A false inclusion of the "a=simulcast" attribute may result in <t>A false inclusion of the "a=simulcast" attribute may result in
simultaneous transmission of multiple RTP streams that would otherwise simultaneous transmission of multiple RTP streams that would otherwise
not be generated. The impact is limited by the media description joint not be generated. The impact is limited by the media description joint
bandwidth, shared by all simulcast streams irrespective of their number. bandwidth, shared by all simulcast streams irrespective of their number.
There may however be a large number of unwanted RTP streams that will However, there may be a large number of unwanted RTP streams that will
impact the share of bandwidth allocated for the originally wanted RTP impact the share of bandwidth allocated for the originally wanted RTP
stream.</t> stream.</t>
<t>A hostile removal of the "a=simulcast" attribute will result in <t>A hostile removal of the "a=simulcast" attribute will result in
simulcast not being used.</t> simulcast not being used.</t>
<t>Neither of the above will likely have any major consequences and can <t>
be mitigated by signaling that is at least integrity and source Integrity protection and source authentication of all SDP signaling,
authenticated to prevent an attacker to change it.</t> including simulcast attributes, can mitigate the risks of such attacks
that attempt to alter signaling.
</t>
<t>Security considerations related to the use of "a=rid" and the <t>Security considerations related to the use of "a=rid" and the
RtpStreamId SDES item is covered in <xref target="I-D.ietf-mmusic-rid"/> RtpStreamId SDES item are covered in <xref target="RFC8851" format="defaul
and <xref target="I-D.ietf-avtext-rid"/>. There are no additional t"/>
and <xref target="RFC8852" format="default"/>. There are no additional
security concerns related to their use in this specification.</t> security concerns related to their use in this specification.</t>
</section> </section>
<section anchor="sec-contributors" title="Contributors">
<t>Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
contributed with important material to the first versions of this
document. Robert Hansen and Cullen Jennings, from Cisco, Peter Thatcher,
from Google, and Adam Roach, from Mozilla, contributed significantly to
subsequent versions.</t>
</section>
<section anchor="sec-ack" title="Acknowledgements">
<t>The authors would like to thank Bernard Aboba, Thomas Belling, Roni
Even, Adam Roach, Inaki Baz Castillo, Paul Kyzivat, and Arun Arunachalam
for the feedback they provided during the development of this
document.</t>
</section>
</middle> </middle>
<back> <back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.4566'?>
<?rfc include='reference.RFC.5234'?>
<?rfc include='reference.RFC.7405'?>
<?rfc include='reference.RFC.7728'?>
<?rfc include='reference.RFC.8174'?>
<?rfc include='reference.I-D.ietf-mmusic-rid'?>
<?rfc include='reference.I-D.ietf-avtext-rid'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-mux-attributes'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.2198'?>
<?rfc include='reference.RFC.3264'?>
<?rfc include='reference.RFC.3389'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.4733'?>
<?rfc include='reference.RFC.5104'?> <displayreference target="I-D.ietf-avtcore-multiplex-guidelines" to="MULTIPLEX"/
>
<?rfc include='reference.RFC.5109'?>
<?rfc include='reference.RFC.5583'?>
<?rfc include='reference.RFC.6184'?>
<?rfc include='reference.RFC.6190'?>
<?rfc include='reference.RFC.6236'?>
<?rfc include='reference.RFC.6464'?>
<?rfc include='reference.RFC.7104'?> <references>
<name>References</name>
<references>
<name>Normative References</name>
<xi:include
href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.
2119.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.3264.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.3550.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.4566.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.5234.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7405.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7728.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.8174.xml"/>
<?rfc include='reference.RFC.7656'?> <!-- draft-ietf-mmusic-rid in C238 -->
<reference anchor="RFC8851" target="https://www.rfc-editor.org/info/rfc8
851">
<front>
<title>RTP Payload Format Restrictions</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach" role="
editor">
<organization/>
</author>
<date month="April" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8851"/>
<seriesInfo name="RFC" value="8851"/>
</reference>
<?rfc include='reference.RFC.7667'?> <!-- draft-ietf-avtext-rid-09 in C238 -->
<reference anchor="RFC8852" target="https://www.rfc-editor.org/info/rfc8
852">
<front>
<title>RTP Stream Identifier Source Description (SDES)</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach">
<organization/>
</author>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar
">
<organization/>
</author>
<author initials="P" surname="Thatcher" fullname="Peter Thatcher">
<organization/>
</author>
<date month="April" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8852"/>
<seriesInfo name="RFC" value="8852"/>
</reference>
<?rfc include='reference.RFC.7741'?> <!-- draft-ietf-mmusic-sdp-mux-attributes-17 in C238 -->
<reference anchor="RFC8859" target="https://www.rfc-editor.org/info/rfc8
859">
<front>
<title>A Framework for SDP Attributes when Multiplexing</title>
<seriesInfo name="DOI" value="10.17487/RFC8859"/>
<seriesInfo name="RFC" value="8859"/>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar
">
<organization/>
</author>
<date month="April" year="2020"/>
</front>
</reference>
<?rfc include='reference.RFC.8108'?> <!-- draft-ietf-mmusic-sdp-bundle-negotiation in C238 -->
<reference anchor="RFC8843" target="https://www.rfc-editor.org/info/rfc8
843">
<front>
<title>Negotiating Media Multiplexing Using the Session Description
Protocol (SDP)</title>
<seriesInfo name="DOI" value="10.17487/RFC8843"/>
<seriesInfo name="RFC" value="8843"/>
<author initials="C" surname="Holmberg" fullname="">
<organization/>
</author>
<author initials="H" surname="Alvestrand" fullname="">
<organization/>
</author>
<author initials="C" surname="Jennings" fullname="">
<organization/>
</author>
<date month="April" year="2020"/>
</front>
</reference>
</references>
<references>
<name>Informative References</name>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.2198.xml"/>
<?rfc include='reference.RFC.8285'?> <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.3389.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.4588.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.4733.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.5104.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.5109.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.5583.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.6184.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.6190.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.6236.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.6464.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7104.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7656.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7667.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7741.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.8108.xml"/>
<xi:include
href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.
7941.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.8627.xml"/>
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?> <!-- draft-ietf-avtcore-multiplex-guidelines-11 in IESG Evaluation -->
<xi:include
href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml3/reference.I-D.ietf-a
vtcore-multiplex-guidelines.xml"/>
<?rfc include='reference.I-D.ietf-payload-flexible-fec-scheme'?> </references>
</references> </references>
<section anchor="sec-requirements" numbered="true" toc="default">
<section anchor="sec-requirements" title="Requirements"> <name>Requirements</name>
<t>The following requirements are met by the defined solution to support <t>The following requirements are met by the defined solution to support
the <xref target="sec-use-cases">use cases</xref>:<list style="hanging"> the <xref target="sec-use-cases" format="default">use cases</xref>:</t>
<t anchor="req-1" hangText="REQ-1:">Identification:<list
style="hanging">
<t anchor="req-1.1" hangText="REQ-1.1:">It must be possible to
identify a set of simulcasted RTP streams as originating from
the same media source in SDP signaling.</t>
<t anchor="req-1.2" hangText="REQ-1.2:">An RTP endpoint must be <dl newline="false" spacing="normal">
capable of identifying the simulcast stream a received RTP <dt>REQ-1:</dt>
<dd anchor="req-1">
<t>Identification:</t>
<dl newline="false" spacing="normal">
<dt>REQ-1.1:</dt>
<dd anchor="req-1.1">It must be possible to
identify a set of simulcasted RTP streams as originating from
the same media source in SDP signaling.</dd>
<dt>REQ-1.2:</dt>
<dd anchor="req-1.2">An RTP endpoint must be
capable of identifying the simulcast stream that a received RTP
stream is associated with, knowing the content of the SDP stream is associated with, knowing the content of the SDP
signalling.</t> signaling.</dd>
</list></t> </dl>
</dd>
<t anchor="req-2" hangText="REQ-2:">Transport usage. The solution <dt>REQ-2:</dt>
must work when using:<list style="hanging"> <dd anchor="req-2">
<t anchor="req-2.1" hangText="REQ-2.1:">Legacy SDP with separate <t>Transport usage. The solution
media transports per SDP media description.</t> must work when using:</t>
<dl newline="false" spacing="normal">
<t anchor="req-2.2" hangText="REQ-2.2:"><xref <dt>REQ-2.1:</dt>
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Bundled</xref> <dd anchor="req-2.1">Legacy SDP with separate
SDP media descriptions.</t> media transports per SDP media description.</dd>
</list></t> <dt>REQ-2.2:</dt>
<dd anchor="req-2.2">
<t anchor="req-3" hangText="REQ-3:">Capability negotiation. It must <xref target="RFC8843" format="default">Bundled</xref>
be possible that:<list style="hanging"> SDP media descriptions.</dd>
<t anchor="req-3.1" hangText="REQ-3.1:">Sender can express </dl>
capability of sending simulcast.</t> </dd>
<t anchor="req-3.2" hangText="REQ-3.2:">Receiver can express
capability of receiving simulcast.</t>
<t anchor="req-3.3" hangText="REQ-3.3:">Sender can express
maximum number of simulcast streams that can be provided.</t>
<t anchor="req-3.4" hangText="REQ-3.4:">Receiver can express
maximum number of simulcast streams that can be received.</t>
<t anchor="req-3.5" hangText="REQ-3.5:">Sender can detail the <dt>REQ-3:</dt>
<dd anchor="req-3">
<t>Capability negotiation. The
following must be possible:</t>
<dl newline="false" spacing="normal">
<dt>REQ-3.1:</dt>
<dd anchor="req-3.1">The sender can express
capability of sending simulcast.</dd>
<dt>REQ-3.2:</dt>
<dd anchor="req-3.2">The receiver can express
capability of receiving simulcast.</dd>
<dt>REQ-3.3:</dt>
<dd anchor="req-3.3">The sender can express
the maximum number of simulcast streams that can be
provided.</dd>
<dt>REQ-3.4:</dt>
<dd anchor="req-3.4">The receiver can express the
maximum number of simulcast streams that can be received.</dd>
<dt>REQ-3.5:</dt>
<dd anchor="req-3.5">The sender can detail the
characteristics of the simulcast streams that can be characteristics of the simulcast streams that can be
provided.</t> provided.</dd>
<dt>REQ-3.6:</dt>
<t anchor="req-3.6" hangText="REQ-3.6:">Receiver can detail the <dd anchor="req-3.6">The receiver can detail the
characteristics of the simulcast streams that it prefers to characteristics of the simulcast streams that it prefers to
receive.</t> receive.</dd>
</list></t> </dl>
</dd>
<t anchor="req-4" hangText="REQ-4:">Distinguishing features. It must <dt>REQ-4:</dt>
<dd anchor="req-4">Distinguishing features. It must
be possible to have different simulcast streams use different codec be possible to have different simulcast streams use different codec
parameters, as can be expressed by SDP format values and RTP payload parameters, as can be expressed by SDP format values and RTP payload
types.</t> types.</dd>
<dt>REQ-5:</dt>
<t anchor="req-5" hangText="REQ-5:">Compatibility. It must be <dd anchor="req-5">
<t>Compatibility. It must be
possible to use simulcast in combination with other RTP mechanisms possible to use simulcast in combination with other RTP mechanisms
that generate additional RTP streams:<list style="hanging"> that generate additional RTP streams:</t>
<t anchor="req-5.1" hangText="REQ-5.1:"><xref <dl newline="false" spacing="normal">
target="RFC4588">RTP Retransmission</xref>.</t> <dt>REQ-5.1:</dt>
<dd anchor="req-5.1">
<t anchor="req-5.2" hangText="REQ-5.2:"><xref <xref target="RFC4588" format="default">RTP retransmission</xref>.
target="RFC5109">RTP Forward Error Correction</xref>.</t> </dd>
<dt>REQ-5.2:</dt>
<t anchor="req-5.3" hangText="REQ-5.3:">Related payload types <dd anchor="req-5.2">
such as audio Comfort Noise and/or DTMF.</t> <xref target="RFC5109" format="default">RTP Forward Error Correcti
on</xref>.</dd>
<t hangText="REQ-5.4:">A single simulcast stream can consist of <dt>REQ-5.3:</dt>
<dd anchor="req-5.3">Related payload types
such as audio Comfort Noise and/or DTMF.</dd>
<dt>REQ-5.4:</dt>
<dd>A single simulcast stream can consist of
multiple RTP streams, to support codecs where a dependent stream multiple RTP streams, to support codecs where a dependent stream
is dependent on a set of encoded and dependent streams, each is dependent on a set of encoded and dependent streams, each
potentially carried in their own RTP stream.</t> potentially carried in their own RTP stream.</dd>
</list></t> </dl>
</dd>
<t anchor="req-6" hangText="REQ-6:">Interoperability. The solution <dt>REQ-6:</dt>
must be possible to use in:<list style="hanging"> <dd anchor="req-6">
<t anchor="req-6.1" hangText="REQ-6.1:">Interworking with <t>Interoperability. The solution
non-simulcast legacy clients using a single media source per must be possible to use in:</t>
media type.</t> <dl newline="false" spacing="normal">
<dt>REQ-6.1:</dt>
<dd anchor="req-6.1">Interworking with
nonsimulcast legacy clients using a single media source per
media type.</dd>
<dt>REQ-6.2:</dt>
<dd anchor="req-6.2">WebRTC environment with
a single media source per SDP media description.</dd>
</dl>
</dd>
</dl>
<t anchor="req-6.2" hangText="REQ-6.2:">WebRTC environment with </section>
a single media source per SDP media description.</t> <section anchor="sec-ack" numbered="false" toc="default">
</list></t> <name>Acknowledgements</name>
</list></t> <t>The authors would like to thank <contact fullname="Bernard Aboba"/>, <c
ontact
fullname="Thomas Belling"/>, <contact fullname="Roni Even"/>, <contact
fullname="Adam Roach"/>, <contact fullname="Iñaki Baz Castillo"/>,
<contact fullname="Paul Kyzivat"/>, and <contact fullname="Arun
Arunachalam"/> for the feedback they provided during the development of
this document.</t>
</section> </section>
<section title="Changes From Earlier Versions"> <section anchor="sec-contributors" numbered="false" toc="default">
<t>NOTE TO RFC EDITOR: Please remove this section prior to <name>Contributors</name>
publication.</t>
<section title="Modifications Between WG Version -13 and -14">
<t><list style="symbols">
<t>c= and t= line order corrected in SDP examples</t>
</list></t>
</section>
<section title="Modifications Between WG Version -12 and -13">
<t><list style="symbols">
<t>Examples corrected to follow RID ABNF</t>
<t>Example <xref target="fig-ms-offer"/> now comments on priority
for second media source.</t>
<t>Clarified a SHOULD limitation.</t>
<t>Added urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id in
examples with RTX.</t>
<t>ABNF now uses RFC 7405 to indicate case sensitivity</t>
<t>Various minor editorials and nits.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -11 and -12">
<t><list style="symbols">
<t>Modified Normative statement regarding RTP stream duplication
in Section 5.2.</t>
<t>Clarified assumption about use of congestion control by
applications.</t>
<t>Changed to use RFC 8174 boilerplate instead of RFC 2119.</t>
<t>Clarified explanation of syntax for simulcast attribute in
Section 4.</t>
<t>Editorial clarification in Section 5.2 and 5.3.2.</t>
<t>Various minor editorials and nits.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -10 and -11">
<t><list style="symbols">
<t>Added new SDP example section on Simulcast and Redundancy,
including both RED (RFC2198), RTP RTX (RFC4588), and FEC
(draft-ietf-payload-flexible-fec-scheme).</t>
<t>Removed restriction that "related" payload formats in an RTP
stream (such as CN and DTMF) must not have their own rid-id, since
there is no reason to forbid this and corresponding clarification
is made in draft-ietf-mmusic-rid.</t>
<t>Removed any mention of source-specific signaling and the
reference to RFC5576, since draft-ietf-mmusic-rid is not defined
for source-specific signaling.</t>
<t>Changed some SDP examples to use a=rid restrictions instead of
a=imageattr.</t>
<t>Changed reference from the obsoleted RFC 5285 to RFC 8285.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -09 and -10">
<t><list style="symbols">
<t>Amended overview section with a bit more explanation on the
examples, and added an rid-id alternative for one of the
streams.</t>
<t>Removed SCID also from the Terminology section, which was
forgotten in -09 when changing SCID to rid-id.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -08 and -09">
<t><list style="symbols">
<t>Changed SCID to rid-id, to align with ietf-draft-mmusic-rid
naming.</t>
<t>Changed Overview to be based on examples and shortened it.</t>
<t>Changed semantics of initially paused rid-id in modified SDP
offers from requiring it to follow actual RFC 7728 pause state to
an informational offerer's opinion at the time of offer creation,
not in any way overriding or amending RFC 7728 signaling.</t>
<t>Replaced text on ignoring all but the first of multiple
"a=simulcast" lines in a media description with mandating that at
most one "a=simulcast" line is included.</t>
<t>Clarified with a note that, for the case it is clear from the
SDP that RTP PT uniquely maps to RtpStreamId, an RTP receiver can
use RTP PT to relate simulcast streams.</t>
<t>Moved Section 4 Requirements to become Appendix A.</t>
<t>Editorial corrections and clarifications.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -07 and -08">
<t><list style="symbols">
<t>Correcting syntax of SDP examples in section 6.6.1, as found by
Inaki Baz Castillo.</t>
<t>Changing ABNF to only define the sc-value, not the SDP
attribute itself, as suggested by Paul Kyzivat.</t>
<t>Changing I-D reference to newly published RFC 8108.</t>
<t>Adding list of modifications between -06 and -07.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -06 and -07">
<t><list style="symbols">
<t>A scope clarification, as result of the discussion with Roni
Even.</t>
<t>A reformulation of the identification requirements for
simulcast stream.</t>
<t>Correcting the statement related to source specific signalling
(RFC 5576) to address Roni Even's comment.</t>
<t>Update of the last paragraph in Section 6.2 regarding simulcast
stream differences as well as forbidding multiple instances of the
same SCID within a single a=simulcast line.</t>
<t>Removal of note in Section 6.4 as result of issue raised by
Roni Even.</t>
<t>Use of "m=" has been changed to media description and a few
other editorial improvements and clarifications.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -05 and -06">
<t><list style="symbols">
<t>Added section on RTP Aspects</t>
<t>Added a requirement (5-4) on that capability exchange must be
capable of handling multi RTP stream cases.</t>
<t>Added extmap attribute also on first signalling example as it
is a recommended to use mechanism.</t>
<t>Clarified the definition of the simulcast attribute and how
simulcast streams relates to simulcast formats and SCIDs.</t>
<t>Updated References list and moved around some references
between informative and normative categories.</t>
<t>Editorial improvements and corrections.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -04 and -05">
<t><list style="symbols">
<t>Aligned with recent changes in draft-ietf-mmusic-rid and
draft-ietf-avtext-rid.</t>
<t>Modified the SDP offer/answer section to follow the generally
accepted structure, also adding a brief text on modifying the
session that is aligned with draft-ietf-mmusic-rid.</t>
<t>Improved text around simulcast stream identification (as
opposed to the simulcast stream itself) to consistently use the
acronym SCID and defined that in the Terminology section.</t>
<t>Changed references for RTP-level pause/resume and VP8 payload
format that are now published as RFC.</t>
<t>Improved IANA registration text.</t>
<t>Removed unused reference to
draft-ietf-payload-flexible-fec-scheme.</t>
<t>Editorial improvements and corrections.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -03 and -04">
<t><list style="symbols">
<t>Changed to only use RID identification, as was consensus during
IETF 94.</t>
<t>ABNF improvements.</t>
<t>Clarified offer-answer rules for initially paused streams.</t>
<t>Changed references for RTP topologies and RTP taxonomy
documents that are now published as RFC.</t>
<t>Added reference to the new RID draft in AVTEXT.</t>
<t>Re-structured section 6 to provide an easy reference by the
updated IANA section.</t>
<t>Added a sub-section 7.1 with a discussion of bitrate
adaptation.</t>
<t>Editorial improvements.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -02 and -03">
<t><list style="symbols">
<t>Removed text on multicast / broadcast from use cases, since it
is not supported by the solution.</t>
<t>Removed explicit references to unified plan draft.</t>
<t>Added possibility to initiate simulcast streams in paused
mode.</t>
<t>Enabled an offerer to offer multiple stream identification (pt
or rid) methods and have the answerer choose which to use.</t>
<t>Added a preference indication also in send direction
offers.</t>
<t>Added a section on limitations of the current proposal,
including identification method specific limitations.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -01 and -02">
<t><list style="symbols">
<t>Relying on the new RID solution for codec constraints and
configuration identification. This has resulted in changes in
syntax to identify if pt or RID is used to describe the simulcast
stream.</t>
<t>Renamed simulcast version and simulcast version alternative to
simulcast stream and simulcast format respectively, and improved
definitions for them.</t>
<t>Clarification that it is possible to switch between simulcast
version alternatives, but that only a single one be used at any
point in time.</t>
<t>Changed the definition so that ordering of simulcast formats
for a specific simulcast stream do have a preference order.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -00 and -01">
<t><list style="symbols">
<t>No changes. Only preventing expiry.</t>
</list></t>
</section>
<section title="Modifications Between Individual Version -00 and WG Versio <t><contact fullname="Morgan Lindqvist"/> and <contact fullname="Fredrik
n -00"> Jansson"/>, both from Ericsson, have contributed with important material
<t><list style="symbols"> to the first draft versions of this document. <contact fullname="Robert
<t>Added this appendix.</t> Hanton"/> and <contact fullname="Cullen Jennings"/> from Cisco, <contact
</list></t> fullname="Peter Thatcher"/> from Google, and <contact fullname="Adam
</section> Roach"/> from Mozilla contributed significantly to subsequent
versions.</t>
</section> </section>
</back> </back>
</rfc> </rfc>
 End of changes. 306 change blocks. 
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