rfc8853xml2.original.xml   rfc8853.xml 
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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" stream="IETF"/>
<author fullname="Bo Burman" initials="B." surname="Burman"> <author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization> <organization showOnFrontPage="true">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 showOnFrontPage="true">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 showOnFrontPage="true">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 showOnFrontPage="true">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="01" year="2021"/>
<date day="5" month="March" year="2019"/> <keyword>Conference</keyword>
<keyword>multi-party</keyword>
<abstract> <keyword>middlebox</keyword>
<t>In some application scenarios it may be desirable to send multiple <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 pn="section-abstract">
<t indent="0" pn="section-abstract-1">In some application scenarios, it ma
y 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>
<boilerplate>
<section anchor="status-of-memo" numbered="false" removeInRFC="false" toc=
"exclude" pn="section-boilerplate.1">
<name slugifiedName="name-status-of-this-memo">Status of This Memo</name
>
<t indent="0" pn="section-boilerplate.1-1">
This is an Internet Standards Track document.
</t>
<t indent="0" pn="section-boilerplate.1-2">
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by
the Internet Engineering Steering Group (IESG). Further
information on Internet Standards is available in Section 2 of
RFC 7841.
</t>
<t indent="0" pn="section-boilerplate.1-3">
Information about the current status of this document, any
errata, and how to provide feedback on it may be obtained at
<eref target="https://www.rfc-editor.org/info/rfc8853" brackets="non
e"/>.
</t>
</section>
<section anchor="copyright" numbered="false" removeInRFC="false" toc="excl
ude" pn="section-boilerplate.2">
<name slugifiedName="name-copyright-notice">Copyright Notice</name>
<t indent="0" pn="section-boilerplate.2-1">
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
</t>
<t indent="0" pn="section-boilerplate.2-2">
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(<eref target="https://trustee.ietf.org/license-info" brackets="none
"/>) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with
respect to this document. Code Components extracted from this
document must include Simplified BSD License text as described in
Section 4.e of the Trust Legal Provisions and are provided without
warranty as described in the Simplified BSD License.
</t>
</section>
</boilerplate>
<toc>
<section anchor="toc" numbered="false" removeInRFC="false" toc="exclude" p
n="section-toc.1">
<name slugifiedName="name-table-of-contents">Table of Contents</name>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="section-to
c.1-1">
<li pn="section-toc.1-1.1">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.1.1"><xref der
ivedContent="1" format="counter" sectionFormat="of" target="section-1"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-introduction">
Introduction</xref></t>
</li>
<li pn="section-toc.1-1.2">
<t indent="0" pn="section-toc.1-1.2.1"><xref derivedContent="2" form
at="counter" sectionFormat="of" target="section-2"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-definitions">Definitions</xref></t
>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.2.2">
<li pn="section-toc.1-1.2.2.1">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.2.2.1.1"><
xref derivedContent="2.1" format="counter" sectionFormat="of" target="section-2.
1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-te
rminology">Terminology</xref></t>
</li>
<li pn="section-toc.1-1.2.2.2">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.2.2.2.1"><
xref derivedContent="2.2" format="counter" sectionFormat="of" target="section-2.
2"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-re
quirements-language">Requirements Language</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.3">
<t indent="0" pn="section-toc.1-1.3.1"><xref derivedContent="3" form
at="counter" sectionFormat="of" target="section-3"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-use-cases">Use Cases</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.3.2">
<li pn="section-toc.1-1.3.2.1">
<t indent="0" pn="section-toc.1-1.3.2.1.1"><xref derivedContent=
"3.1" format="counter" sectionFormat="of" target="section-3.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-reaching-a-diverse-set
-of-r">Reaching a Diverse Set of Receivers</xref></t>
</li>
<li pn="section-toc.1-1.3.2.2">
<t indent="0" pn="section-toc.1-1.3.2.2.1"><xref derivedContent=
"3.2" format="counter" sectionFormat="of" target="section-3.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-application-specific-m
edia-">Application-Specific Media Source Handling</xref></t>
</li>
<li pn="section-toc.1-1.3.2.3">
<t indent="0" pn="section-toc.1-1.3.2.3.1"><xref derivedContent=
"3.3" format="counter" sectionFormat="of" target="section-3.3"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-receiver-media-source-
prefe">Receiver Media-Source Preferences</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.4">
<t indent="0" pn="section-toc.1-1.4.1"><xref derivedContent="4" form
at="counter" sectionFormat="of" target="section-4"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-overview">Overview</xref></t>
</li>
<li pn="section-toc.1-1.5">
<t indent="0" pn="section-toc.1-1.5.1"><xref derivedContent="5" form
at="counter" sectionFormat="of" target="section-5"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-detailed-description">Detailed Des
cription</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.5.2">
<li pn="section-toc.1-1.5.2.1">
<t indent="0" pn="section-toc.1-1.5.2.1.1"><xref derivedContent=
"5.1" format="counter" sectionFormat="of" target="section-5.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-simulcast-attribute">S
imulcast Attribute</xref></t>
</li>
<li pn="section-toc.1-1.5.2.2">
<t indent="0" pn="section-toc.1-1.5.2.2.1"><xref derivedContent=
"5.2" format="counter" sectionFormat="of" target="section-5.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-simulcast-capability">
Simulcast Capability</xref></t>
</li>
<li pn="section-toc.1-1.5.2.3">
<t indent="0" pn="section-toc.1-1.5.2.3.1"><xref derivedContent=
"5.3" format="counter" sectionFormat="of" target="section-5.3"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-offer-answer-use">Offe
r/Answer Use</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="se
ction-toc.1-1.5.2.3.2">
<li pn="section-toc.1-1.5.2.3.2.1">
<t indent="0" pn="section-toc.1-1.5.2.3.2.1.1"><xref derived
Content="5.3.1" format="counter" sectionFormat="of" target="section-5.3.1"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-generating
-the-initial-sdp-">Generating the Initial SDP Offer</xref></t>
</li>
<li pn="section-toc.1-1.5.2.3.2.2">
<t indent="0" pn="section-toc.1-1.5.2.3.2.2.1"><xref derived
Content="5.3.2" format="counter" sectionFormat="of" target="section-5.3.2"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-creating-t
he-sdp-answer">Creating the SDP Answer</xref></t>
</li>
<li pn="section-toc.1-1.5.2.3.2.3">
<t indent="0" pn="section-toc.1-1.5.2.3.2.3.1"><xref derived
Content="5.3.3" format="counter" sectionFormat="of" target="section-5.3.3"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-offerer-pr
ocessing-the-sdp-">Offerer Processing the SDP Answer</xref></t>
</li>
<li pn="section-toc.1-1.5.2.3.2.4">
<t indent="0" pn="section-toc.1-1.5.2.3.2.4.1"><xref derived
Content="5.3.4" format="counter" sectionFormat="of" target="section-5.3.4"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-modifying-
the-session">Modifying the Session</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.5.2.4">
<t indent="0" pn="section-toc.1-1.5.2.4.1"><xref derivedContent=
"5.4" format="counter" sectionFormat="of" target="section-5.4"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-use-with-declarative-s
dp">Use with Declarative SDP</xref></t>
</li>
<li pn="section-toc.1-1.5.2.5">
<t indent="0" pn="section-toc.1-1.5.2.5.1"><xref derivedContent=
"5.5" format="counter" sectionFormat="of" target="section-5.5"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-relating-simulcast-str
eams">Relating Simulcast Streams</xref></t>
</li>
<li pn="section-toc.1-1.5.2.6">
<t indent="0" pn="section-toc.1-1.5.2.6.1"><xref derivedContent=
"5.6" format="counter" sectionFormat="of" target="section-5.6"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-signaling-examples">Si
gnaling Examples</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="se
ction-toc.1-1.5.2.6.2">
<li pn="section-toc.1-1.5.2.6.2.1">
<t indent="0" pn="section-toc.1-1.5.2.6.2.1.1"><xref derived
Content="5.6.1" format="counter" sectionFormat="of" target="section-5.6.1"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-single-sou
rce-client">Single-Source Client</xref></t>
</li>
<li pn="section-toc.1-1.5.2.6.2.2">
<t indent="0" pn="section-toc.1-1.5.2.6.2.2.1"><xref derived
Content="5.6.2" format="counter" sectionFormat="of" target="section-5.6.2"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-multisourc
e-client">Multisource Client</xref></t>
</li>
<li pn="section-toc.1-1.5.2.6.2.3">
<t indent="0" pn="section-toc.1-1.5.2.6.2.3.1"><xref derived
Content="5.6.3" format="counter" sectionFormat="of" target="section-5.6.3"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-simulcast-
and-redundancy">Simulcast and Redundancy</xref></t>
</li>
</ul>
</li>
</ul>
</li>
<li pn="section-toc.1-1.6">
<t indent="0" pn="section-toc.1-1.6.1"><xref derivedContent="6" form
at="counter" sectionFormat="of" target="section-6"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-rtp-aspects">RTP Aspects</xref></t
>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.6.2">
<li pn="section-toc.1-1.6.2.1">
<t indent="0" pn="section-toc.1-1.6.2.1.1"><xref derivedContent=
"6.1" format="counter" sectionFormat="of" target="section-6.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-outgoing-from-endpoint
-with">Outgoing from Endpoint with Media Source</xref></t>
</li>
<li pn="section-toc.1-1.6.2.2">
<t indent="0" pn="section-toc.1-1.6.2.2.1"><xref derivedContent=
"6.2" format="counter" sectionFormat="of" target="section-6.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-rtp-middlebox-to-recei
ver">RTP Middlebox to Receiver</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="se
ction-toc.1-1.6.2.2.2">
<li pn="section-toc.1-1.6.2.2.2.1">
<t indent="0" pn="section-toc.1-1.6.2.2.2.1.1"><xref derived
Content="6.2.1" format="counter" sectionFormat="of" target="section-6.2.1"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-media-swit
ching-mixer">Media-Switching Mixer</xref></t>
</li>
<li pn="section-toc.1-1.6.2.2.2.2">
<t indent="0" pn="section-toc.1-1.6.2.2.2.2.1"><xref derived
Content="6.2.2" format="counter" sectionFormat="of" target="section-6.2.2"/>.  <
xref derivedContent="" format="title" sectionFormat="of" target="name-selective-
forwarding-middle">Selective Forwarding Middlebox</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.6.2.3">
<t indent="0" pn="section-toc.1-1.6.2.3.1"><xref derivedContent=
"6.3" format="counter" sectionFormat="of" target="section-6.3"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-rtp-middlebox-to-rtp-m
iddle">RTP Middlebox to RTP Middlebox</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.7">
<t indent="0" pn="section-toc.1-1.7.1"><xref derivedContent="7" form
at="counter" sectionFormat="of" target="section-7"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-network-aspects">Network Aspects</
xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.7.2">
<li pn="section-toc.1-1.7.2.1">
<t indent="0" pn="section-toc.1-1.7.2.1.1"><xref derivedContent=
"7.1" format="counter" sectionFormat="of" target="section-7.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-bitrate-adaptation">Bi
trate Adaptation</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.8">
<t indent="0" pn="section-toc.1-1.8.1"><xref derivedContent="8" form
at="counter" sectionFormat="of" target="section-8"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-limitation">Limitation</xref></t>
</li>
<li pn="section-toc.1-1.9">
<t indent="0" pn="section-toc.1-1.9.1"><xref derivedContent="9" form
at="counter" sectionFormat="of" target="section-9"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-iana-considerations">IANA Consider
ations</xref></t>
</li>
<li pn="section-toc.1-1.10">
<t indent="0" pn="section-toc.1-1.10.1"><xref derivedContent="10" fo
rmat="counter" sectionFormat="of" target="section-10"/>. <xref derivedContent=""
format="title" sectionFormat="of" target="name-security-considerations">Securit
y Considerations</xref></t>
</li>
<li pn="section-toc.1-1.11">
<t indent="0" pn="section-toc.1-1.11.1"><xref derivedContent="11" fo
rmat="counter" sectionFormat="of" target="section-11"/>. <xref derivedContent=""
format="title" sectionFormat="of" target="name-references">References</xref></t
>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.11.2">
<li pn="section-toc.1-1.11.2.1">
<t indent="0" pn="section-toc.1-1.11.2.1.1"><xref derivedContent
="11.1" format="counter" sectionFormat="of" target="section-11.1"/>.  <xref deri
vedContent="" format="title" sectionFormat="of" target="name-normative-reference
s">Normative References</xref></t>
</li>
<li pn="section-toc.1-1.11.2.2">
<t indent="0" pn="section-toc.1-1.11.2.2.1"><xref derivedContent
="11.2" format="counter" sectionFormat="of" target="section-11.2"/>.  <xref deri
vedContent="" format="title" sectionFormat="of" target="name-informative-referen
ces">Informative References</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.12">
<t indent="0" pn="section-toc.1-1.12.1"><xref derivedContent="Append
ix A" format="default" sectionFormat="of" target="section-appendix.a"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-requirements">R
equirements</xref></t>
</li>
<li pn="section-toc.1-1.13">
<t indent="0" pn="section-toc.1-1.13.1"><xref derivedContent="" form
at="none" sectionFormat="of" target="section-appendix.b"/><xref derivedContent="
" format="title" sectionFormat="of" target="name-acknowledgements">Acknowledgeme
nts</xref></t>
</li>
<li pn="section-toc.1-1.14">
<t indent="0" pn="section-toc.1-1.14.1"><xref derivedContent="" form
at="none" sectionFormat="of" target="section-appendix.c"/><xref derivedContent="
" format="title" sectionFormat="of" target="name-contributors">Contributors</xre
f></t>
</li>
<li pn="section-toc.1-1.15">
<t indent="0" pn="section-toc.1-1.15.1"><xref derivedContent="" form
at="none" sectionFormat="of" target="section-appendix.d"/><xref derivedContent="
" format="title" sectionFormat="of" target="name-authors-addresses">Authors' Add
resses</xref></t>
</li>
</ul>
</section>
</toc>
</front> </front>
<middle> <middle>
<section anchor="sec-intro" title="Introduction"> <section anchor="sec-intro" numbered="true" toc="include" removeInRFC="false
<t>Most of today's multiparty video conference solutions make use of " pn="section-1">
<name slugifiedName="name-introduction">Introduction</name>
<t indent="0" pn="section-1-1">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 indent="0" pn="section-1-2">Simulcast is defined in this memo as the ac
<t>Simulcast is defined in this memo as the act of simultaneously t 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" sectionFormat="of" derivedContent="RFC3550">RTP</xref> tow
ards 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 indent="0" pn="section-1-3">The intended scope of the defined mechanism
<t>The intended scope of the defined mechanism is to support negotiation 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 indent="0" pn="section-1-4">This document describes a few scenarios tha
<t>This document describes a few scenarios that motivate the use of t 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="include" removeInRFC=
<section anchor="sec-definitions" title="Definitions"> "false" pn="section-2">
<t/> <name slugifiedName="name-definitions">Definitions</name>
<section numbered="true" toc="include" removeInRFC="false" pn="section-2.1
<section title="Terminology"> ">
<t>This document makes use of the terminology defined in <xref <name slugifiedName="name-terminology">Terminology</name>
target="RFC7656">RTP Taxonomy</xref>, and <xref target="RFC7667">RTP <t indent="0" pn="section-2.1-1">This document makes use of the terminol
Topologies</xref>. The following terms are especially noted or here ogy defined in <xref target="RFC7656" format="default" sectionFormat="of" derive
defined:<list style="hanging"> dContent="RFC7656">"A Taxonomy of Semantics and
<t hangText="RTP Mixer:">An RTP middle node, defined in <xref Mechanisms for Real-Time
target="RFC7667"/> (Section 3.6 to 3.9).</t> Transport Protocol (RTP) Sources"</xref> and <xref target="RFC7667" forma
t="default" sectionFormat="of" derivedContent="RFC7667">"RTP Topologies"</xref>.
<t hangText="RTP Session:">An association among a group of The following terms are
participants communicating with RTP, as defined in <xref especially noted or here defined:</t>
target="RFC3550"/> and amended by <xref target="RFC7656"/>.</t> <dl newline="false" spacing="normal" indent="3" pn="section-2.1-2">
<dt pn="section-2.1-2.1">RTP mixer:</dt>
<t hangText="RTP Stream:">A stream of RTP packets containing media <dd pn="section-2.1-2.2">An RTP middlebox, in the wide sense of the te
data, as defined in <xref target="RFC7656"/>.</t> rm, encompassing
Sections <xref target="RFC7667" section="3.6" sectionFormat="bare" for
<t hangText="RTP Switch:">A common short term for the terms mat="default" derivedLink="https://rfc-editor.org/rfc/rfc7667#section-3.6" deriv
edContent="RFC7667"/>
to <xref target="RFC7667" section="3.9" sectionFormat="bare" format="d
efault" derivedLink="https://rfc-editor.org/rfc/rfc7667#section-3.9" derivedCont
ent="RFC7667"/> of
<xref target="RFC7667" format="default" sectionFormat="of" derivedCont
ent="RFC7667"/>.</dd>
<dt pn="section-2.1-2.3">RTP session:</dt>
<dd pn="section-2.1-2.4">An association among a group of
participants communicating with RTP, as defined in <xref target="RFC
3550" format="default" sectionFormat="of" derivedContent="RFC3550"/> and amended
by <xref target="RFC7656" format="default" sectionFormat="of" derivedContent="R
FC7656"/>.</dd>
<dt pn="section-2.1-2.5">RTP stream:</dt>
<dd pn="section-2.1-2.6">A stream of RTP packets containing media
data, as defined in <xref target="RFC7656" format="default" sectionF
ormat="of" derivedContent="RFC7656"/>.</dd>
<dt pn="section-2.1-2.7">RTP switch:</dt>
<dd pn="section-2.1-2.8">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" sectionFormat="of" derivedContent="RFC7667"/>.</dd
<t hangText="Simulcast Stream:">One encoded stream or dependent >
<dt pn="section-2.1-2.9">Simulcast stream:</dt>
<dd pn="section-2.1-2.10">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" sectionFormat="of " derivedContent="RFC7656"/>. For example, HD 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 pn="section-2.1-2.11">Simulcast format:</dt>
<t hangText="Simulcast Format:">Different formats of a simulcast <dd pn="section-2.1-2.12">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" sectionFormat=" of" derivedContent="RFC3264">offer/answer</xref>, 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="include" removeInRFC="false" pn="section-2.2
<section title="Requirements Language"> ">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", <name slugifiedName="name-requirements-language">Requirements Language</
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and name>
"OPTIONAL" in this document are to be interpreted as described in BCP <t indent="0" pn="section-2.2-1">
14 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and only The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
when, they appear in all capitals, as shown here.</t> "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL NOT</bcp14>
", "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
"<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 14 <xref target="RFC2119" format="default" sectionFormat="o
f" derivedContent="RFC2119"/> <xref target="RFC8174" format="default" sectionFor
mat="of" derivedContent="RFC8174"/>
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="include" removeInRFC="f
<section anchor="sec-use-cases" title="Use Cases"> alse" pn="section-3">
<t>The use cases of simulcast described in this document relate to a <name slugifiedName="name-use-cases">Use Cases</name>
multi-party communication session where one or more central nodes are <t indent="0" pn="section-3-1">The use cases of simulcast described in thi
s document relate to a
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 indent="0" pn="section-3-2">There are two principal approaches for an R
<t>There are two principal approaches for an RTP Mixer to provide this TP 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" bare="false" empty="false" indent="3" pn="section-3-3
">
<li pn="section-3-3.1">Transcoding (decoding and re-encoding) received R
TP 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 pn="section-3-3.2">Switching a subset of all received RTP streams or
<t>Switching a subset of all received RTP streams or sub-streams to substreams 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 indent="0" pn="section-3-4">The use of simulcast relates to the latter
<t>The use of simulcast relates to the latter approach, where it is more 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="include" remo
<section anchor="sec-diverse-receivers" veInRFC="false" pn="section-3.1">
title="Reaching a Diverse Set of Receivers"> <name slugifiedName="name-reaching-a-diverse-set-of-r">Reaching a Divers
<t>The media sources provided by a sending participant potentially e Set of Receivers</name>
<t indent="0" pn="section-3.1-1">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" indent="3" pn="section-3.1-2">
<dt pn="section-3.1-2.1">Codec:</dt>
<dd pn="section-3.1-2.2">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 pn="section-3.1-2.3">Sampling:</dt>
<t hangText="Sampling:">This relates to how the media source is <dd pn="section-3.1-2.4">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 pn="section-3.1-2.5">Bitrate:</dt>
<t hangText="Bitrate:">This relates to the number of bits sent per <dd pn="section-3.1-2.6">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 indent="0" pn="section-3.1-3">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 indent="0" pn="section-3.1-4">This requires that the receiving partic
<t>This requires that the receiving participants can be categorized in ipants 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 indent="0" pn="section-3.1-5">The maximum number of simulcasted RTP s
<t>The maximum number of simulcasted RTP streams that can be sent is treams 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="include" r
<section anchor="sec-application-specific" emoveInRFC="false" pn="section-3.2">
title="Application Specific Media Source Handling"> <name slugifiedName="name-application-specific-media-">Application-Speci
<t>The application logic that controls the communication session may fic Media Source Handling</name>
<t indent="0" pn="section-3.2-1">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 indent="0" pn="section-3.2-2">It is also common that a currently acti
<t>It is also common that a currently active speaker participant is ve 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" form at="default" sectionFormat="of" derivedContent="Section 3.1"/>)
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="include" r
<section anchor="sec-receiver-preferences" emoveInRFC="false" pn="section-3.3">
title="Receiver Media Source Preferences"> <name slugifiedName="name-receiver-media-source-prefe">Receiver Media-So
<t>The application logic that controls the communication session may urce Preferences</name>
<t indent="0" pn="section-3.3-1">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" sectionFormat="of" derivedContent="Section 3.1"/>.
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="include" removeInRFC="fa
<section anchor="sec-overview" title="Overview"> lse" pn="section-4">
<t>This memo defines <xref target="RFC4566">SDP</xref> signaling that <name slugifiedName="name-overview">Overview</name>
<t indent="0" pn="section-4-1">This memo defines <xref target="RFC4566" fo
rmat="default" sectionFormat="of" derivedContent="RFC4566">SDP</xref> signaling
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" sectionFormat="of" derivedContent="Appendix A
"/>.</t>
<t>The RID mechanism, as defined in <xref <t indent="0" pn="section-4-2">The Restriction Identifier (RID) mechanism,
target="I-D.ietf-mmusic-rid"/>, enables an SDP offerer or answerer to as defined in <xref target="RFC8851" format="default" sectionFormat="of" derive
dContent="RFC8851"/>, 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 indent="0" pn="section-4-3">A new SDP media-level attribute, "a=simulca
<t>A new SDP media level attribute "a=simulcast" is defined. The st", is defined. The
attribute describes, independently for send and receive directions, the attribute describes, independently for "send" and "receive" directions, th
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" sectionFormat="of" derivedConte
nt="RFC8851"/>.</t>
<figure align="left"> <sourcecode type="sdp" markers="false" pn="section-4-4">
<artwork align="left"><![CDATA[a=simulcast:send 1;2,3 recv 4 a=simulcast:send 1;2,3 recv 4
]]></artwork> </sourcecode>
</figure> <t indent="0" pn="section-4-5">If this line is included in an SDP offer, t
he "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 indent="0" pn="section-4-6">A more complete example SDP-offer media des
<t>A more complete example SDP offer media description is provided cription is provided
below:</t> in <xref target="fig-md-offer" format="default" sectionFormat="of" derived
Content="Figure 1"/>.</t>
<figure align="center" anchor="fig-md-offer" <figure anchor="fig-md-offer" align="left" suppress-title="false" pn="figu
title="Example Simulcast Media Description in Offer"> re-1">
<artwork align="left"><![CDATA[ <name slugifiedName="name-example-simulcast-media-des">Example Simulcast
Media Description in Offer</name>
<sourcecode type="sdp" markers="false" pn="section-4-7.1">
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>
<t indent="0" pn="section-4-8">The SDP media description in <xref target="
<t>The above SDP media description can be interpreted at a high level to fig-md-offer" format="default" sectionFormat="of" derivedContent="Figure 1"/> ca
say that the offerer is capable of sending two simulcast RTP streams, n 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 indent="0" pn="section-4-9">The receiver of this SDP offer can generate
<t>The receiver of this SDP offer can generate an SDP answer that 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>
<sourcecode type="sdp" markers="false" pn="section-4-10">
<figure align="left"> a=simulcast:recv 1;2 send 4
<artwork align="left"><![CDATA[a=simulcast:recv 1;2 send 4 </sourcecode>
]]></artwork> <t indent="0" pn="section-4-11">With this SDP answer, the answerer indicat
</figure> es 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" sectionFormat="of" derivedContent="Figure 2"/>.</t>
<figure align="center" anchor="fig-md-answer" <figure anchor="fig-md-answer" align="left" suppress-title="false" pn="fig
title="Example Simulcast Media Description in Answer"> ure-2">
<artwork align="left"><![CDATA[ <name slugifiedName="name-example-simulcast-media-desc">Example Simulcas
t Media Description in Answer</name>
<sourcecode type="sdp" markers="false" pn="section-4-12.1">
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>
<t indent="0" pn="section-4-13">It is assumed that a single SDP media desc
<t>It is assumed that a single SDP media description is used to describe ription 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" sectionFormat="of" derivedContent=
and without <xref "RFC7656"/> and will work in a WebRTC context, 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" sectionFormat="of" derivedContent="RFC8843"/>.</t>
<t indent="0" pn="section-4-14">To summarize, the "a=simulcast" line descr
<t>To summarize, the "a=simulcast" line describes send and receive ibes "send"- and
direction simulcast streams separately. Each direction can in turn "receive"-direction simulcast streams separately. Each direction can in
describe one or more simulcast streams, separated by semicolon. The turn describe one or more simulcast streams, separated by semicolons. 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" sectionFormat="of" derivedContent="RFC8851"/>. Each simulcast stream can b
a list of alternative rid-id, with each alternative separated by comma e offered as
(not in the examples above). A detailed specification can be found in a list of alternative rid-ids, with each alternative separated by a comma
<xref target="sec-details"/> and more detailed examples are outlined in as shown in the example offer in <xref target="fig-md-offer" format="defau
<xref target="sec-ex"/>.</t> lt" sectionFormat="of" derivedContent="Figure 1"/>. A detailed specification can
be found in
<xref target="sec-details" format="default" sectionFormat="of" derivedCont
ent="Section 5"/>, and more detailed examples are outlined in
<xref target="sec-ex" format="default" sectionFormat="of" derivedContent="
Section 5.6"/>.</t>
</section> </section>
<section anchor="sec-details" numbered="true" toc="include" removeInRFC="fal
<section anchor="sec-details" title="Detailed Description"> se" pn="section-5">
<t>This section further details the overview <xref <name slugifiedName="name-detailed-description">Detailed Description</name
target="sec-overview">above</xref>. First, formal syntax is <xref >
target="sec-attr">provided</xref>, followed by the rest of the SDP <t indent="0" pn="section-5-1">This section provides further details to th
attribute definition in <xref target="sec-cap"/>. <xref e overview in <xref target="sec-overview" format="default" sectionFormat="of" de
target="sec-relating">Relating Simulcast Streams </xref> provides the rivedContent="Section 4"/>. First, formal syntax is <xref target="sec-attr" form
definition of the RTP/RTCP mechanisms used. The section is concluded at="default" sectionFormat="of" derivedContent="Section 5.1">provided</xref>, fo
llowed by the rest of the SDP
attribute definition in <xref target="sec-cap" format="default" sectionFor
mat="of" derivedContent="Section 5.2"/>. <xref target="sec-relating" format="def
ault" sectionFormat="of" derivedContent="Section 5.5">"Relating Simulcast Stream
s"</xref> provides 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="include" removeInRFC="fals
<section anchor="sec-attr" title="Simulcast Attribute"> e" pn="section-5.1">
<t>This document defines a new SDP media-level "a=simulcast" <name slugifiedName="name-simulcast-attribute">Simulcast Attribute</name
attribute, with value according to the following <xref >
target="RFC5234">ABNF</xref> syntax and its update for <xref <t indent="0" pn="section-5.1-1">This document defines a new SDP media-l
target="RFC7405">Case-Sensitive String Support in ABNF</xref>:</t> evel "a=simulcast"
attribute, with value according to the syntax in <xref target="fig-abnf"
<figure align="center" anchor="fig-abnf" format="default" sectionFormat="of" derivedContent="Figure 3"/>, which uses <xr
title="ABNF for Simulcast Value"> ef target="RFC5234" format="default" sectionFormat="of" derivedContent="RFC5234"
<artwork align="center"><![CDATA[ >ABNF</xref> and its update, <xref target="RFC7405" format="default" sectionForm
at="of" derivedContent="RFC7405">"Case-Sensitive String Support in ABNF"</xref>:
</t>
<figure anchor="fig-abnf" align="left" suppress-title="false" pn="figure
-3">
<name slugifiedName="name-abnf-for-simulcast-value">ABNF for Simulcast
Value</name>
<sourcecode type="abnf" markers="false" pn="section-5.1-2.1">
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>
<t indent="0" pn="section-5.1-3">The "a=simulcast" attribute has a param
<t><list style="empty"> eter in the form of one or
<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
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" sectionFormat="of" derivedContent="RFC8851">"RTP Payload Format Restrictions"
</xref>.</t>
<t>In the list of simulcast streams, each simulcast stream is <t indent="0" pn="section-5.1-4">In the list of simulcast streams, each
separated by a semicolon (";"). Each simulcast stream can in turn be simulcast stream is
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" sectionFormat="of" der
ivedContent="RFC7728">paused</xref>, indicated 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="include" removeInRFC="false
<section anchor="sec-cap" title="Simulcast Capability"> " pn="section-5.2">
<t>Simulcast capability is expressed through a new media level <xref <name slugifiedName="name-simulcast-capability">Simulcast Capability</na
target="sec-attr">SDP attribute, "a=simulcast"</xref>. The use of this me>
<t indent="0" pn="section-5.2-1">Simulcast capability is expressed throu
gh a new media-level <xref target="sec-attr" format="default" sectionFormat="of"
derivedContent="Section 5.1">SDP attribute, "a=simulcast"</xref>. The use of th
is
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 indent="0" pn="section-5.2-2">There are separate and independent sets
<t>There are separate and independent sets of simulcast streams in of simulcast streams in the
send and receive directions. When listing multiple directions, each "send" and "receive" directions. When listing multiple directions, each
direction MUST NOT occur more than once on the same line.</t> direction <bcp14>MUST NOT</bcp14> occur more than once on the same line.
</t>
<t>Simulcast streams using undefined rid-id MUST NOT be used as valid <t indent="0" pn="section-5.2-3">Simulcast streams using undefined rid-i
simulcast streams by an RTP stream receiver. The direction for an ds <bcp14>MUST NOT</bcp14> be used as valid
rid-id MUST be aligned with the direction specified for the simulcast streams by an RTP stream receiver. The direction for a
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 indent="0" pn="section-5.2-4">The listed number of simulcast streams
<t>The listed number of simulcast streams for a direction sets a limit 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 indent="0" pn="section-5.2-5">rid-ids that have explicit <xref target
<t>rid-id that have explicit <xref ="RFC5583" format="default" sectionFormat="of" derivedContent="RFC5583">dependen
target="RFC5583">dependencies</xref> <xref cies</xref> <xref target="RFC8851" format="default" sectionFormat="of" derivedCo
target="I-D.ietf-mmusic-rid"/> to other rid-id (even in the same media ntent="RFC8851"/> to other rid-ids (even in the same media
description) MAY be used.</t> description) <bcp14>MAY</bcp14> be used.</t>
<t indent="0" pn="section-5.2-6">Use of more than a single, alternative
<t>Use of more than a single, alternative simulcast format for a 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>A simulcast stream can use a codec defined such that the same RTP </t>
SSRC can change RTP payload type multiple times during a session, <t indent="0" pn="section-5.2-7">A simulcast stream can use a codec defi
possibly even on a per-packet basis. A typical example can be a speech ned such that the same RTP
codec that makes use of <xref target="RFC3389">Comfort Noise</xref> synchronization source (SSRC) can change RTP payload type multiple
and/or <xref target="RFC4733">DTMF</xref> formats.</t> times during a session, possibly even on a per-packet basis. A typical
example is a speech codec that makes use of formats for <xref target="RF
<t>If <xref target="RFC7728">RTP stream pause/resume</xref> is C3389" format="default" sectionFormat="of" derivedContent="RFC3389">Comfort Nois
supported, any rid-id MAY be prefixed by a "~" character to indicate e</xref> and/or <xref target="RFC4733" format="default" sectionFormat="of" deriv
that the corresponding simulcast stream is initially paused already edContent="RFC4733">dual-tone multifrequency
from start of the RTP session. In this case, support for RTP stream (DTMF)</xref>.</t>
pause/resume MUST also be included under the same "m=" line where <t indent="0" pn="section-5.2-8">If <xref target="RFC7728" format="defau
lt" sectionFormat="of" derivedContent="RFC7728">RTP stream
pause/resume</xref> is supported, any rid-id <bcp14>MAY</bcp14> be
prefixed by a "~" character to indicate that the corresponding
simulcast stream is paused already from the start of the RTP
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" forma
using the "*" wildcard format for "a=rtcp-fb".</t> t="default" sectionFormat="of" derivedContent="RFC7728"/> -- e.g., by using the
"*" wildcard format for
<t>An initially paused simulcast stream in "send" direction for the "a=rtcp-fb".</t>
endpoint sending the SDP MUST be considered equivalent to an <t indent="0" pn="section-5.2-9">An initially paused simulcast stream in
unsolicited locally paused stream, and be handled accordingly. the "send" direction for the
endpoint sending the SDP <bcp14>MUST</bcp14> be considered equivalent to
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="def
ault" sectionFormat="of" derivedContent="RFC8843">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="RFC88
52" format="default" sectionFormat="of" derivedContent="RFC8852">RtpStreamId RTC
P SDES
item</xref>.</t>
<t indent="0" pn="section-5.2-10">If the endpoint sending the SDP includ
es 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" sect
target="RFC7728">Section 5.6 of </xref>) since the RTP receiver's SSRC ionFormat="of" section="5.6" format="default" derivedLink="https://rfc-editor.or
in send direction is sometimes not yet known.</t> g/rfc/rfc7728#section-5.6" derivedContent="RFC7728"/>); this is because the RTP
receiver's SSRC
<t>Use of the <xref target="RFC2198">redundant audio data</xref> in the "send" direction is sometimes not yet known.</t>
format could be seen as a form of simulcast for loss protection <t indent="0" pn="section-5.2-11">Use of the redundant audio data format
purposes, but is not considered conflicting with the mechanisms <xref target="RFC2198" format="default" sectionFormat="of" derivedContent="RFC2
described in this memo and MAY therefore be used as any other format. 198"/>
In this case the "red" format, rather than the carried formats, SHOULD could be seen as a form of simulcast for loss-protection
purposes, but it is not considered conflicting with the mechanisms
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 indent="0" pn="section-5.2-12">The media formats and corresponding ch
<t>The media formats and corresponding characteristics of simulcast aracteristics 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" sectionFormat="of" der
MUST NOT occur more than once per "a=simulcast" line. Note that this ivedContent="RFC7104"/> instead of simulcast. 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="include" removeInR
<section anchor="sec-offer-answer" title="Offer/Answer Use"> FC="false" pn="section-5.3">
<t><list style="empty"> <name slugifiedName="name-offer-answer-use">Offer/Answer Use</name>
<t>Note: The inclusion of "a=simulcast" or the use of simulcast <dl indent="3" newline="false" spacing="normal" pn="section-5.3-1">
<dt pn="section-5.3-1.1">Note:</dt>
<dd pn="section-5.3-1.2">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="include" removeInRFC="false" pn="section-5
<t>An offerer wanting to use simulcast for a media description SHALL .3.1">
<name slugifiedName="name-generating-the-initial-sdp-">Generating the
Initial SDP Offer</name>
<t indent="0" pn="section-5.3.1-1">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="include" removeInRFC="false" pn="section-5
<section title="Creating the SDP Answer"> .3.2">
<t>An answerer that does not understand the concept of simulcast <name slugifiedName="name-creating-the-sdp-answer">Creating the SDP An
swer</name>
<t indent="0" pn="section-5.3.2-1">An answerer that does not understan
d 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" sectionFormat="of" derivedContent="RFC3264"/>. Sinc
e 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 indent="0" pn="section-5.3.2-2">An answerer that does understand th
<t>An answerer that does understand the attribute and that wants to e attribute and wants to
support simulcast in an indicated direction SHALL reverse support simulcast in an indicated direction <bcp14>SHALL</bcp14> rever
directionality of the unidirectional direction parameters; "send" se
becomes "recv" and vice versa, and include it in the answer.</t> directionality of the unidirectional direction parameters -- "send"
becomes "recv" and vice versa -- and include it in the answer.</t>
<t>An answerer that receives an offer with simulcast containing an <t indent="0" pn="section-5.3.2-3">An answerer that receives an offer
"a=simulcast" attribute listing alternative rid-id MAY keep all the with simulcast containing an
alternative rid-id in the answer, but it MAY also choose to remove "a=simulcast" attribute listing alternative rid-ids <bcp14>MAY</bcp14>
any non-desirable alternative rid-id in the answer. The answerer keep all the
MUST NOT add any alternative rid-id in send direction in the answer alternative rid-ids in the answer, but it <bcp14>MAY</bcp14> also choo
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 indent="0" pn="section-5.3.2-4">An answerer that receives an offer
<t>An answerer that receives an offer with simulcast that lists a 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
<t>An answerer that receives an offer without RTP stream reams.</t>
pause/resume capability MUST NOT mark any simulcast streams as <t indent="0" pn="section-5.3.2-5">An answerer that receives an offer
without RTP stream
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 indent="0" pn="section-5.3.2-6">An RTP stream answerer capable of p
<t>An RTP stream pause/resume capable answerer that receives an ause/resume that receives an
offer with RTP stream pause/resume capability MAY mark any rid-id offer with RTP stream pause/resume capability <bcp14>MAY</bcp14> mark
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 indent="0" pn="section-5.3.2-7">An answerer that receives indicatio
<t>An answerer that receives indication in an offer of an rid-id n in an offer of a rid-id
being initially paused SHOULD mark that rid-id as initially paused being initially paused <bcp14>SHOULD</bcp14> mark that rid-id as initi
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="include" removeInRFC="false" pn="section-5
<section title="Offerer Processing the SDP Answer"> .3.3">
<t>An offerer that receives an answer without "a=simulcast" MUST NOT <name slugifiedName="name-offerer-processing-the-sdp-">Offerer Process
ing the SDP Answer</name>
<t indent="0" pn="section-5.3.3-1">An offerer that receives an answer
without "a=simulcast" <bcp14>MUST 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 indent="0" pn="section-5.3.3-2">An offerer that receives an answer
<t>An offerer that receives an answer where some rid-id alternatives 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 indent="0" pn="section-5.3.3-3">An offerer that receives an answer
<t>An offerer that receives an answer where some of the rid-id are where some of the rid-ids are
removed compared to the offer MAY release the corresponding removed compared to the offer <bcp14>MAY</bcp14> release the correspon
resources (codec, transport, etc) in its receive direction and MUST ding
NOT send any RTP packets corresponding to the removed rid-id.</t> resources (codec, transport, etc) in its "receive" direction and <bcp1
4>MUST NOT</bcp14> send any RTP packets corresponding to the removed rid-ids.</t
<t>An offerer that offered some of its rid-id as initially paused >
and that receives an answer that does not indicate RTP stream <t indent="0" pn="section-5.3.3-4">An offerer that offered some of its
pause/resume capability, MUST NOT initially pause any simulcast rid-ids as initially paused
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 indent="0" pn="section-5.3.3-5">An offerer with RTP stream pause/re
<t>An offerer with RTP stream pause/resume capability that receives sume 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="include" removeInRFC="false" pn="section-5
<section title="Modifying the Session"> .3.4">
<t>Offers inside an existing session follow the same rules as for <name slugifiedName="name-modifying-the-session">Modifying the Session
initial SDP offer, with these additions:<list style="numbers"> </name>
<t>rid-id marked as initially paused in the offerer's send <t indent="0" pn="section-5.3.4-1">Offers inside an existing session f
direction SHALL reflect the offerer's opinion of the current ollow the same rules as for
initial SDP offer, with these additions:</t>
<ol spacing="normal" type="1" indent="adaptive" start="1" pn="section-
5.3.4-2">
<li pn="section-5.3.4-2.1" derivedCounter="1.">rid-ids marked as ini
tially 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" sec
precedence in case of any conflict or ambiguity.</t> tionFormat="of" derivedContent="RFC7728"/> in 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 pn="section-5.3.4-2.2" derivedCounter="2.">rid-ids marked as ini
tially 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" sectionFormat="of" derived
, this is identical to the conditions at an initial offer.</t> Content="RFC7728">RTP stream pause/resume</xref> 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 indent="0" pn="section-5.3.4-3">Creation of SDP answers and process
ing 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 indent="0" pn="section-5.3.4-4">Session modification restrictions i
<t>Session modification restrictions in section 6.5 of <xref n <xref target="RFC8851" sectionFormat="of" section="6.5" format="default" deriv
target="I-D.ietf-mmusic-rid">RTP payload format restrictions</xref> edLink="https://rfc-editor.org/rfc/rfc8851#section-6.5" derivedContent="RFC8851"
>"RTP Payload Format
Restrictions"</xref>
also apply.</t> also apply.</t>
</section> </section>
</section> </section>
<section numbered="true" toc="include" removeInRFC="false" pn="section-5.4
<section title="Use with Declarative SDP"> ">
<t>This document does not define the use of "a=simulcast" in <name slugifiedName="name-use-with-declarative-sdp">Use with Declarative
declarative SDP, partly motivated by use of the <xref SDP</name>
target="I-D.ietf-mmusic-rid">simulcast format identification</xref> <t indent="0" pn="section-5.4-1">This document does not define the use o
not being defined for use in declarative SDP. If concrete use cases f "a=simulcast" in
declarative SDP, partly because use of the <xref target="RFC8851" format
="default" sectionFormat="of" derivedContent="RFC8851">simulcast format identifi
cation</xref>
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="include" removeInRFC="
<section anchor="sec-relating" title="Relating Simulcast Streams"> false" pn="section-5.5">
<t>Simulcast RTP streams MUST be related on RTP level through <xref <name slugifiedName="name-relating-simulcast-streams">Relating Simulcast
target="I-D.ietf-avtext-rid">RtpStreamId</xref>, as specified in the Streams</name>
SDP <xref target="sec-cap">"a=simulcast" attribute </xref> parameters. <t indent="0" pn="section-5.5-1">Simulcast RTP streams <bcp14>MUST</bcp1
4> be related on the RTP
level through <xref target="RFC8852" format="default" sectionFormat="of"
derivedContent="RFC8852">RtpStreamId</xref>, as specified in the
SDP <xref target="sec-cap" format="default" sectionFormat="of" derivedCo
ntent="Section 5.2">"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 " sectionFormat="of" derivedContent="RFC8843">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" format="de
(based on RTP timestamps), but MAY change format (and rid-id) on a fault" sectionFormat="of" derivedContent="RFC7941">RTCP
per-RTP packet basis. This corresponds to the existing (non-simulcast) source description items</xref> for both MID
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" indent="3" pn="section-5.5-2">
<dt pn="section-5.5-2.1">NOTE:</dt>
<dd pn="section-5.5-2.2">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 indent="0" pn="section-5.5-3">RTP streams <bcp14>MUST</bcp14> only us
e 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="include" removeInRFC="false"
<section anchor="sec-ex" title="Signaling Examples"> pn="section-5.6">
<t>These examples describe a client to video conference service, using <name slugifiedName="name-signaling-examples">Signaling Examples</name>
<t indent="0" pn="section-5.6-1">These examples describe a client-to-vid
eo-conference service, using
a centralized media topology with an RTP mixer.</t> a centralized media topology with an RTP mixer.</t>
<figure anchor="fig-mixer-four-party" align="left" suppress-title="false
<figure align="center" anchor="fig-mixer-four-party" " pn="figure-4">
title="Four-party Mixer-based Conference"> <name slugifiedName="name-four-party-mixer-based-conf">Four-Party Mixe
<artwork align="center"><![CDATA[ r-Based Conference</name>
<artwork align="center" name="" type="" alt="" pn="section-5.6-2.1">
+---+ +-----------+ +---+ +---+ +-----------+ +---+
| A |<---->| |<----&gt;| B | | A |<----&gt;| |&lt;----&gt;| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| F |<---->| |<---->| J | | F |&lt;----&gt;| |&lt;----&gt;| J |
+---+ +-----------+ +---+]]></artwork> +---+ +-----------+ +---+</artwork>
</figure> </figure>
---+ +-----------+ <span class="insert">+---+&lt;/artwork&gt;</span>
<section anchor="sec-ex-single-source" title="Single-Source Client"> <section anchor="sec-ex-single-source" numbered="true" toc="include" rem
<t>Alice is calling in to the mixer with a simulcast-enabled client oveInRFC="false" pn="section-5.6.1">
<name slugifiedName="name-single-source-client">Single-Source Client</
name>
<t indent="0" pn="section-5.6.1-1">Alice is calling in to the mixer wi
th 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" sectionFormat="of" derivedCont
"pt" "a=rid" parameter is used, effectively achieving a 1:1 mapping ent="RFC6236">"imageattr"</xref>. In this example, 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
<figure align="center" anchor="fig-up-offer" er:</t>
title="Single-Source Simulcast Offer"> <figure anchor="fig-up-offer" align="left" suppress-title="false" pn="
<artwork align="left"><![CDATA[ figure-5">
<name slugifiedName="name-single-source-simulcast-off">Single-Source
Simulcast Offer</name>
<sourcecode type="sdp" markers="false" pn="section-5.6.1-2.1">
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>
<t indent="0" pn="section-5.6.1-3">The only thing in the SDP that indi
<t>The only thing in the SDP that indicates simulcast capability is cates 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 indent="0" pn="section-5.6.1-4">The answer from the server indicate
<t>The Answer from the server indicates that it too is simulcast s 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>
<figure anchor="fig-up-answer" align="left" suppress-title="false" pn=
<figure align="center" anchor="fig-up-answer" "figure-6">
title="Single-Source Simulcast Answer"> <name slugifiedName="name-single-source-simulcast-ans">Single-Source
<artwork align="left"><![CDATA[ Simulcast Answer</name>
<sourcecode type="sdp" markers="false" pn="section-5.6.1-5.1">
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>
<t indent="0" pn="section-5.6.1-6">Since the server is the simulcast m
<t>Since the server is the simulcast media receiver, it reverses the edia 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="include" remo
<section anchor="sec-ex-multi-source" title="Multi-Source Client"> veInRFC="false" pn="section-5.6.2">
<t>Fred is calling in to the same conference as in the example above <name slugifiedName="name-multisource-client">Multisource Client</name
>
<t indent="0" pn="section-5.6.2-1">Fred is calling in to the same conf
erence 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 indent="0" pn="section-5.6.2-2">The first two simulcast streams for
<t>The first two simulcast streams for the first media source use the first media source use
different codecs, <xref target="RFC6190">H264-SVC</xref> and <xref different codecs, <xref target="RFC6190" format="default" sectionForma
target="RFC6184">H264</xref>. These two simulcast streams also have t="of" derivedContent="RFC6190">H264-SVC</xref> and <xref target="RFC6184" forma
a temporal dependency. Two different video codecs, <xref t="default" sectionFormat="of" derivedContent="RFC6184">H264</xref>. These two s
target="RFC7741">VP8</xref> and H264, are offered as alternatives imulcast streams also have
a temporal dependency. Two different video codecs, <xref target="RFC77
41" format="default" sectionFormat="of" derivedContent="RFC7741">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 indent="0" pn="section-5.6.2-3">The second media source is offered
<t>The second media source is offered with three different simulcast 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" sectionFormat="of
here, all but the highest fidelity simulcast stream are initially " derivedContent="RFC4588">RTP retransmission</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 indent="0" pn="section-5.6.2-4">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" format="default" secti
title="Fred's Multi-Source Simulcast Offer"> onFormat="of" derivedContent="RFC8843">BUNDLE</xref> and <xref target="RFC8851"
<artwork><![CDATA[ format="default" sectionFormat="of" derivedContent="RFC8851">"a=rid"</xref>, it
is recommended to use the RTP
header extension for the <xref target="RFC7941" format="default" sectio
nFormat="of" derivedContent="RFC7941">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" sectionFormat="of" derive
dContent="RFC8852">rtp-stream-id</xref>.</t>
<figure anchor="fig-ms-offer" align="left" suppress-title="false" pn="
figure-7">
<name slugifiedName="name-freds-multisource-simulcast">Fred's Multis
ource Simulcast Offer</name>
<sourcecode type="sdp" markers="false" pn="section-5.6.2-5.1">
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 1063
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>
</section> </section>
<section numbered="true" toc="include" removeInRFC="false" pn="section-5
.6.3">
<name slugifiedName="name-simulcast-and-redundancy">Simulcast and Redu
ndancy</name>
<t indent="0" pn="section-5.6.3-1">The example in this section looks a
t applying simulcast with
audio and video redundancy formats.
<section title="Simulcast and Redundancy"> The audio media description uses codec and bitrate restrictions,
<t>The example in this section looks at applying simulcast with combined with the <xref target="RFC2198" format="default" sectionFormat="
audio and video redundancy formats. The audio media description uses of" derivedContent="RFC2198">RTP
codec and bitrate restrictions, combining it with <xref payload for redundant audio data</xref> for enhanced packet-loss
target="RFC2198">RTP Payload for Redundant Audio Data</xref> for resilience. The video media description applies both resolution and
enhanced packet loss resilience. The video media description applies bitrate restrictions, combined with Forward Error Correction (FEC)
both resolution and bitrate restrictions, combining it with FEC in in the form of <xref target="RFC8627" format="default" sectionFormat="of"
the form of <xref derivedContent="RFC8627">flexible
target="I-D.ietf-payload-flexible-fec-scheme">Flexible FEC</xref> FEC</xref> and <xref target="RFC4588" format="default" sectionFormat="of"
and <xref target="RFC4588">RTP Retransmission</xref>.</t> derivedContent="RFC4588">RTP
retransmission</xref>.</t>
<t>The audio source is offered to be sent as two simulcast streams. <t indent="0" pn="section-5.6.3-2">
The first simulcast stream is encoded with Opus, restricted to 50 The audio source is offered to be sent as two simulcast
kbps (rid-id=5), and the second simulcast stream is encoded either streams. The first simulcast stream is encoded with Opus,
with G.711 (rid-id=7) or with G.711 combined with LPC for redundancy restricted to 64 kbps (rid-id=1), and the second simulcast stream
(rid-id=6). In this example, stand-alone LPC is not offered as an (rid-id=2) is encoded with either G.711, or G.711 combined with
possible payload type for the second simulcast stream's RID, which linear predictive coding (LPC) for redundancy and explicit comfort
could e.g. be motivated by not providing sufficient quality.</t> noise (CN). Both simulcast streams include telephone-event
capability. In this example, stand-alone LPC is not offered as a
<t>The video source is offered to be sent as two simulcast streams, possible payload type for the second simulcast stream's RID, which
could be motivated by, for example, not providing sufficient
quality.
</t>
<t indent="0" pn="section-5.6.3-3">The video source is offered to be s
ent 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>
<figure anchor="fig-sim-red" align="left" suppress-title="false" pn="f
<figure anchor="fig-sim-red" igure-8">
title="Simulcast and Redundancy Example"> <name slugifiedName="name-simulcast-and-redundancy-ex">Simulcast and
<artwork><![CDATA[v=0 Redundancy Example</name>
<sourcecode type="sdp" markers="false" pn="section-5.6.3-4.1">
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 1134
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>
<t/>
</section> </section>
</section> </section>
</section> </section>
<section anchor="sec-rtp-aspects" numbered="true" toc="include" removeInRFC=
<section anchor="sec-rtp-aspects" title="RTP Aspects"> "false" pn="section-6">
<t>This section discusses what the different entities in a simulcast <name slugifiedName="name-rtp-aspects">RTP Aspects</name>
media path can expect to happen on RTP level. This is explored from <t indent="0" pn="section-6-1">This section discusses what the different e
ntities in a simulcast
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="include" removeInRFC="false" pn="section-6.1
<section title="Outgoing from Endpoint with Media Source"> ">
<t>The most straightforward simulcast case is the RTP streams being <name slugifiedName="name-outgoing-from-endpoint-with">Outgoing from End
point with Media Source</name>
<t indent="0" pn="section-6.1-1">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" sectionFormat="of" derivedContent="Section 5.5"/>) 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" sectionFormat="of" derivedContent="RFC8843">BUNDLE</xref> is used,
also the MID SDES item will be sent similarly to the RtpStreamId.</t> the MID SDES item will also be
sent, similarly to the RtpStreamId.</t>
<t>Each RTP stream might not be continuously transmitted due to any of <t indent="0" pn="section-6.1-2">Each RTP stream might not be continuous
the following reasons; temporarily paused using <xref ly transmitted due to any of
target="RFC7728">Pause/Resume</xref>, sender side application logic the following reasons: temporarily paused using <xref target="RFC7728" f
ormat="default" sectionFormat="of" derivedContent="RFC7728">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="include" removeInRFC="false" pn="section-6.2
<section title="RTP Middlebox to Receiver"> ">
<t>RTP streams in a multi-party RTP session can be used in multiple <name slugifiedName="name-rtp-middlebox-to-receiver">RTP Middlebox to Re
different ways, when the session utilizes simulcast at least on the ceiver</name>
media source to middlebox legs. This is to a large degree due to the <t indent="0" pn="section-6.2-1">RTP streams in a multiparty RTP session
can be used in multiple
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
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" sectionFormat="of"
derivedContent="Section 6.3"/>).</t>
<t>The selection of which simulcast streams to forward towards the <t indent="0" pn="section-6.2-2">The selection of which simulcast stream
receiver, is application specific. However, in conferencing s to forward towards the
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> " sectionFormat="of" derivedContent="RFC6464">RTP header
extension for client-to-mixer audio level indication</xref>.</t>
<t>The possibilities for stream switching are media type specific, but <t indent="0" pn="section-6.2-3">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" sectionFormat="of" derivedContent="RFC5104">Full Intr
a 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 indent="0" pn="section-6.2-4">Inclusion of the RtpStreamId SDES item
<t>Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox for an SSRC in the
to receiver direction should only occur when use of RtpStreamId has middlebox-to-receiver direction should only occur when use of
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="include" removeInRFC="false" pn="section-6
<section title="Media-Switching Mixer"> .2.1">
<t>This section discusses the behavior in cases where the RTP <name slugifiedName="name-media-switching-mixer">Media-Switching Mixer
middlebox behaves like the Media-Switching Mixer (Section 3.6.2) in </name>
<xref target="RFC7667">RTP Topologies</xref>. The fundamental aspect <t indent="0" pn="section-6.2.1-1">This section discusses the behavior
in cases where the RTP
middlebox behaves like the media-switching mixer in
RTP topologies (<xref target="RFC7667" sectionFormat="of" section="3.6
.2" format="default" derivedLink="https://rfc-editor.org/rfc/rfc7667#section-3.6
.2" derivedContent="RFC7667"/>). 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 indent="0" pn="section-6.2.1-2">The above results in the RTP stream
<t>The above results in that the RTP stream produced by the mixer is produced by the mixer being
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 indent="0" pn="section-6.2.1-3">This will result in a single RTP st
<t>This will result in a single RTP stream being used for each of ream 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 indent="0" pn="section-6.2.1-4">The mixer may communicate the ident
<t>The mixer may communicate the identity of the originating media ity 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 indent="0" pn="section-6.2.1-5">It is important to note that any MI
<t>It is important to note that any MID SDES item from the D 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 indent="0" pn="section-6.2.1-6">If RtpStreamIds are used in the sce
<t>If RtpStreamIds are used in the scenario described by this nario 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="include" removeInRFC="false" pn="section-6
<section title="Selective Forwarding Middlebox"> .2.2">
<t>This section discusses the behavior in cases where the RTP <name slugifiedName="name-selective-forwarding-middle">Selective Forwa
middlebox behaves like the Selective Forwarding Middlebox (Section rding Middlebox</name>
3.7) in <xref target="RFC7667">RTP Topologies</xref>. Applications <t indent="0" pn="section-6.2.2-1">This section discusses the behavior
for this type of RTP middlebox results in that each originating in cases where the RTP
media source will have a corresponding media source on the leg middlebox behaves like the Selective Forwarding Middlebox in RTP
topologies (<xref target="RFC7667" sectionFormat="of" section="3.7" for
mat="default" derivedLink="https://rfc-editor.org/rfc/rfc7667#section-3.7" deriv
edContent="RFC7667"/>). Applications
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 indent="0" pn="section-6.2.2-2">This usage will result in the indiv
<t>This usage will result in that the individual RTP stream(s) for idual RTP stream(s) for
one media source can switch between being active to paused, based on one media source being able to switch between being active and
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 indent="0" pn="section-6.2.2-3">Any MID SDES items being sent by th
<t>Any MID SDES items being sent by the SFM to the receiver are only e 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 indent="0" pn="section-6.2.2-4">An SFM could expose corresponding R
<t>A SFM could expose corresponding RTP streams for all the media TP 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 an 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" sectionFormat="of"
derivedContent="RFC7728">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="include" removeInRF
<section anchor="sec-rtp-box-box" title="RTP Middlebox to RTP Middlebox"> C="false" pn="section-6.3">
<t>This relates to the transmission of simulcast streams between RTP <name slugifiedName="name-rtp-middlebox-to-rtp-middle">RTP Middlebox to
RTP Middlebox</name>
<t indent="0" pn="section-6.3-1">This relates to the transmission of sim
ulcast 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 indent="0" pn="section-6.3-2">The MIDs used between A and B are the o
nes agreed between these two
<t>The MIDs used between A and B are the ones agreed between these two identities in signaling. The RtpStreamId values will also be provided
identities in signalling. 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="include" removeIn
<section anchor="sec-network-aspects" title="Network Aspects"> RFC="false" pn="section-7">
<t>Simulcast is in this memo defined as the act of sending multiple <name slugifiedName="name-network-aspects">Network Aspects</name>
alternative encoded streams of the same underlying media source. When <t indent="0" pn="section-7-1">Simulcast is in this memo defined as the ac
transmitting multiple independent streams that originate from the same t of sending multiple
source, it could potentially be done in several different ways using alternative encoded streams of the same underlying media
source. Transmitting multiple independent streams that originate from
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="RFC8872" format="d
target="I-D.ietf-avtcore-multiplex-guidelines">Guidelines for efault" sectionFormat="of" derivedContent="RFC8872">"Guidelines for Using the Mu
Multiplexing in RTP</xref>. Discussion and clarification on how to ltiplexing Features 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" sectionFormat="of" derive
<t>The network aspects that are relevant for simulcast are:<list dContent="RFC8108"/>.</t>
style="hanging"> <t indent="0" pn="section-7-2">The network aspects that are relevant for s
<t hangText="Quality of Service:">When using simulcast it might be imulcast are:</t>
<dl newline="false" spacing="normal" indent="3" pn="section-7-3">
<dt pn="section-7-3.1">Quality of Service (QoS):</dt>
<dd pn="section-7-3.2">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>
<dt pn="section-7-3.3">NAT/FW Traversal (Network Address Translator / Fi
<t hangText="NAT/FW Traversal:">Using multiple RTP sessions incurs rewall Traversal):</dt>
more cost for NAT/FW traversal unless they can re-use the same <dd pn="section-7-3.4">Using multiple RTP sessions incurs
transport flow, which can be achieved by <xref more cost for NAT/FW traversal unless they can reuse the same
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Multiplexing transport flow, which can be achieved by <xref target="RFC8843" format
Negotiation Using SDP Port Numbers</xref>.</t> ="default" sectionFormat="of" derivedContent="RFC8843">multiplexing negotiation
</list></t> using SDP port
numbers</xref>.</dd>
<t/> </dl>
<t indent="0" pn="section-7-4"/>
<section title="Bitrate Adaptation"> <section numbered="true" toc="include" removeInRFC="false" pn="section-7.1
<t>Use of multiple simulcast streams can require a significant amount ">
<name slugifiedName="name-bitrate-adaptation">Bitrate Adaptation</name>
<t indent="0" pn="section-7.1-1">Use of multiple simulcast streams can r
equire 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 " sectionFormat="of" derivedContent="Section 5.2"/>). 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="include" removeInRFC="
<section anchor="sec-limitation" title="Limitation"> false" pn="section-8">
<t>The chosen approach has a limitation that relates to the use of a <name slugifiedName="name-limitation">Limitation</name>
<t indent="0" pn="section-8-1">The chosen approach has a limitation that r
elates 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 indent="0" pn="section-8-2">It is not possible to use different simulca
<t>It is not possible to use different simulcast streams on different st 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 indent="0" pn="section-8-3">It is also not possible to separate differe
<t>It is also not possible to separate different simulcast streams into nt 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="include" removeInRFC="false"
<section anchor="sec-iana" title="IANA Considerations"> pn="section-9">
<t>This document requests to register a new media-level SDP attribute, <name slugifiedName="name-iana-considerations">IANA Considerations</name>
<t indent="0" pn="section-9-1">This document registers a new media-level S
DP 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" sectionFormat="of" d
target="I-D.ietf-mmusic-sdp-mux-attributes"/>.<list style="hanging"> erivedContent="RFC4566"/> and <xref target="RFC8859" format="default" sectionFor
<t hangText="Contact name, email:">The IESG (iesg@ietf.org)</t> mat="of" derivedContent="RFC8859"/>.</t>
<dl newline="false" spacing="normal" indent="3" pn="section-9-2">
<t hangText="Attribute name:">simulcast</t> <dt pn="section-9-2.1">Contact name, email:</dt>
<dd pn="section-9-2.2">The IESG (iesg@ietf.org)</dd>
<t hangText="Long-form attribute name:">Simulcast stream <dt pn="section-9-2.3">Attribute name:</dt>
description</t> <dd pn="section-9-2.4">simulcast</dd>
<dt pn="section-9-2.5">Long-form attribute name:</dt>
<t hangText="Charset dependent:">No</t> <dd pn="section-9-2.6">Simulcast stream description</dd>
<dt pn="section-9-2.7">Charset dependent:</dt>
<t hangText="Attribute value:">sc-value; see <xref <dd pn="section-9-2.8">No</dd>
target="sec-attr"/> of RFC XXXX.</t> <dt pn="section-9-2.9">Attribute value:</dt>
<dd pn="section-9-2.10">sc-value; see <xref target="sec-attr" format="de
<t hangText="Purpose:">Signals simulcast capability for a set of RTP fault" sectionFormat="of" derivedContent="Section 5.1"/> of RFC
streams</t> 8853.</dd>
<dt pn="section-9-2.11">Purpose:</dt>
<t hangText="MUX category:">NORMAL</t> <dd pn="section-9-2.12">Signals simulcast capability for a set of RTP
</list>Note to RFC Editor: Please replace "RFC XXXX" with the assigned streams</dd>
number of this RFC.</t> <dt pn="section-9-2.13">Mux category:</dt>
<dd pn="section-9-2.14">NORMAL</dd>
</dl>
</section> </section>
<section anchor="sec-security" numbered="true" toc="include" removeInRFC="fa
<section anchor="sec-security" title="Security Considerations"> lse" pn="section-10">
<t>The simulcast capability, configuration attributes, and parameters <name slugifiedName="name-security-considerations">Security Considerations
</name>
<t indent="0" pn="section-10-1">The simulcast capability, configuration at
tributes, and parameters
are vulnerable to attacks in signaling.</t> are vulnerable to attacks in signaling.</t>
<t indent="0" pn="section-10-2">A false inclusion of the "a=simulcast" att
<t>A false inclusion of the "a=simulcast" attribute may result in ribute 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 indent="0" pn="section-10-3">A hostile removal of the "a=simulcast" att
<t>A hostile removal of the "a=simulcast" attribute will result in ribute will result in
simulcast not being used.</t> simulcast not being used.</t>
<t indent="0" pn="section-10-4">
<t>Neither of the above will likely have any major consequences and can Integrity protection and source authentication of all SDP signaling,
be mitigated by signaling that is at least integrity and source including simulcast attributes, can mitigate the risks of such attacks
authenticated to prevent an attacker to change it.</t> that attempt to alter signaling.
</t>
<t>Security considerations related to the use of "a=rid" and the <t indent="0" pn="section-10-5">Security considerations related to the use
RtpStreamId SDES item is covered in <xref target="I-D.ietf-mmusic-rid"/> of "a=rid" and the
and <xref target="I-D.ietf-avtext-rid"/>. There are no additional RtpStreamId SDES item are covered in <xref target="RFC8851" format="defaul
t" sectionFormat="of" derivedContent="RFC8851"/>
and <xref target="RFC8852" format="default" sectionFormat="of" derivedCont
ent="RFC8852"/>. 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"> <references pn="section-11">
<?rfc include="reference.RFC.2119"?> <name slugifiedName="name-references">References</name>
<references pn="section-11.1">
<?rfc include='reference.RFC.3550'?> <name slugifiedName="name-normative-references">Normative References</na
me>
<?rfc include='reference.RFC.4566'?> <reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2
119" quoteTitle="true" derivedAnchor="RFC2119">
<?rfc include='reference.RFC.5234'?> <front>
<title>Key words for use in RFCs to Indicate Requirement Levels</tit
<?rfc include='reference.RFC.7405'?> le>
<author initials="S." surname="Bradner" fullname="S. Bradner">
<?rfc include='reference.RFC.7728'?> <organization showOnFrontPage="true"/>
</author>
<?rfc include='reference.RFC.8174'?> <date year="1997" month="March"/>
<abstract>
<?rfc include='reference.I-D.ietf-mmusic-rid'?> <t indent="0">In many standards track documents several words are
used to signify the requirements in the specification. These words are often ca
<?rfc include='reference.I-D.ietf-avtext-rid'?> pitalized. This document defines these words as they should be interpreted in IE
TF documents. This document specifies an Internet Best Current Practices for th
<?rfc include='reference.I-D.ietf-mmusic-sdp-mux-attributes'?> e Internet Community, and requests discussion and suggestions for improvements.<
/t>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?> </abstract>
</references> </front>
<seriesInfo name="BCP" value="14"/>
<references title="Informative References"> <seriesInfo name="RFC" value="2119"/>
<?rfc include='reference.RFC.2198'?> <seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>
<?rfc include='reference.RFC.3264'?> <reference anchor="RFC3264" target="https://www.rfc-editor.org/info/rfc3
264" quoteTitle="true" derivedAnchor="RFC3264">
<?rfc include='reference.RFC.3389'?> <front>
<title>An Offer/Answer Model with Session Description Protocol (SDP)
<?rfc include='reference.RFC.4588'?> </title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<?rfc include='reference.RFC.4733'?> <organization showOnFrontPage="true"/>
</author>
<?rfc include='reference.RFC.5104'?> <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne
">
<?rfc include='reference.RFC.5109'?> <organization showOnFrontPage="true"/>
</author>
<?rfc include='reference.RFC.5583'?> <date year="2002" month="June"/>
<abstract>
<?rfc include='reference.RFC.6184'?> <t indent="0">This document defines a mechanism by which two entit
ies can make use of the Session Description Protocol (SDP) to arrive at a common
<?rfc include='reference.RFC.6190'?> view of a multimedia session between them. In the model, one participant offer
s the other a description of the desired session from their perspective, and the
<?rfc include='reference.RFC.6236'?> other participant answers with the desired session from their perspective. Thi
s offer/answer model is most useful in unicast sessions where information from b
<?rfc include='reference.RFC.6464'?> oth participants is needed for the complete view of the session. The offer/answ
er model is used by protocols like the Session Initiation Protocol (SIP). [STAN
<?rfc include='reference.RFC.7104'?> DARDS-TRACK]</t>
</abstract>
<?rfc include='reference.RFC.7656'?> </front>
<seriesInfo name="RFC" value="3264"/>
<?rfc include='reference.RFC.7667'?> <seriesInfo name="DOI" value="10.17487/RFC3264"/>
</reference>
<?rfc include='reference.RFC.7741'?> <reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3
550" quoteTitle="true" derivedAnchor="RFC3550">
<?rfc include='reference.RFC.8108'?> <front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<?rfc include='reference.RFC.8285'?> <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne
">
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?> <organization showOnFrontPage="true"/>
</author>
<?rfc include='reference.I-D.ietf-payload-flexible-fec-scheme'?> <author initials="S." surname="Casner" fullname="S. Casner">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Frederick" fullname="R. Frederick">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson">
<organization showOnFrontPage="true"/>
</author>
<date year="2003" month="July"/>
<abstract>
<t indent="0">This memorandum describes RTP, the real-time transpo
rt protocol. RTP provides end-to-end network transport functions suitable for a
pplications transmitting real-time data, such as audio, video or simulation data
, over multicast or unicast network services. RTP does not address resource res
ervation and does not guarantee quality-of- service for real-time services. The
data transport is augmented by a control protocol (RTCP) to allow monitoring of
the data delivery in a manner scalable to large multicast networks, and to prov
ide minimal control and identification functionality. RTP and RTCP are designed
to be independent of the underlying transport and network layers. The protocol
supports the use of RTP-level translators and mixers. Most of the text in this
memorandum is identical to RFC 1889 which it obsoletes. There are no changes in
the packet formats on the wire, only changes to the rules and algorithms govern
ing how the protocol is used. The biggest change is an enhancement to the scalab
le timer algorithm for calculating when to send RTCP packets in order to minimiz
e transmission in excess of the intended rate when many participants join a sess
ion simultaneously. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="STD" value="64"/>
<seriesInfo name="RFC" value="3550"/>
<seriesInfo name="DOI" value="10.17487/RFC3550"/>
</reference>
<reference anchor="RFC4566" target="https://www.rfc-editor.org/info/rfc4
566" quoteTitle="true" derivedAnchor="RFC4566">
<front>
<title>SDP: Session Description Protocol</title>
<author initials="M." surname="Handley" fullname="M. Handley">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson">
<organization showOnFrontPage="true"/>
</author>
<author initials="C." surname="Perkins" fullname="C. Perkins">
<organization showOnFrontPage="true"/>
</author>
<date year="2006" month="July"/>
<abstract>
<t indent="0">This memo defines the Session Description Protocol (
SDP). SDP is intended for describing multimedia sessions for the purposes of se
ssion announcement, session invitation, and other forms of multimedia session in
itiation. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4566"/>
<seriesInfo name="DOI" value="10.17487/RFC4566"/>
</reference>
<reference anchor="RFC5234" target="https://www.rfc-editor.org/info/rfc5
234" quoteTitle="true" derivedAnchor="RFC5234">
<front>
<title>Augmented BNF for Syntax Specifications: ABNF</title>
<author initials="D." surname="Crocker" fullname="D. Crocker" role="
editor">
<organization showOnFrontPage="true"/>
</author>
<author initials="P." surname="Overell" fullname="P. Overell">
<organization showOnFrontPage="true"/>
</author>
<date year="2008" month="January"/>
<abstract>
<t indent="0">Internet technical specifications often need to defi
ne a formal syntax. Over the years, a modified version of Backus-Naur Form (BNF
), called Augmented BNF (ABNF), has been popular among many Internet specificati
ons. The current specification documents ABNF. It balances compactness and simp
licity with reasonable representational power. The differences between standard
BNF and ABNF involve naming rules, repetition, alternatives, order-independence
, and value ranges. This specification also supplies additional rule definition
s and encoding for a core lexical analyzer of the type common to several Interne
t specifications. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="STD" value="68"/>
<seriesInfo name="RFC" value="5234"/>
<seriesInfo name="DOI" value="10.17487/RFC5234"/>
</reference>
<reference anchor="RFC7405" target="https://www.rfc-editor.org/info/rfc7
405" quoteTitle="true" derivedAnchor="RFC7405">
<front>
<title>Case-Sensitive String Support in ABNF</title>
<author initials="P." surname="Kyzivat" fullname="P. Kyzivat">
<organization showOnFrontPage="true"/>
</author>
<date year="2014" month="December"/>
<abstract>
<t indent="0">This document extends the base definition of ABNF (A
ugmented Backus-Naur Form) to include a way to specify US-ASCII string literals
that are matched in a case-sensitive manner.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7405"/>
<seriesInfo name="DOI" value="10.17487/RFC7405"/>
</reference>
<reference anchor="RFC7728" target="https://www.rfc-editor.org/info/rfc7
728" quoteTitle="true" derivedAnchor="RFC7728">
<front>
<title>RTP Stream Pause and Resume</title>
<author initials="B." surname="Burman" fullname="B. Burman">
<organization showOnFrontPage="true"/>
</author>
<author initials="A." surname="Akram" fullname="A. Akram">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Even" fullname="R. Even">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<date year="2016" month="February"/>
<abstract>
<t indent="0">With the increased popularity of real-time multimedi
a applications, it is desirable to provide good control of resource usage, and u
sers also demand more control over communication sessions. This document descri
bes how a receiver in a multimedia conversation can pause and resume incoming da
ta from a sender by sending real-time feedback messages when using the Real-time
Transport Protocol (RTP) for real- time data transport. This document extends
the Codec Control Message (CCM) RTP Control Protocol (RTCP) feedback package by
explicitly allowing and describing specific use of existing CCMs and adding a gr
oup of new real-time feedback messages used to pause and resume RTP data streams
. This document updates RFC 5104.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7728"/>
<seriesInfo name="DOI" value="10.17487/RFC7728"/>
</reference>
<reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8
174" quoteTitle="true" derivedAnchor="RFC8174">
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</ti
tle>
<author initials="B." surname="Leiba" fullname="B. Leiba">
<organization showOnFrontPage="true"/>
</author>
<date year="2017" month="May"/>
<abstract>
<t indent="0">RFC 2119 specifies common key words that may be used
in protocol specifications. This document aims to reduce the ambiguity by cla
rifying that only UPPERCASE usage of the key words have the defined special mea
nings.</t>
</abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="8174"/>
<seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>
<reference anchor="RFC8843" target="https://www.rfc-editor.org/info/rfc8
843" quoteTitle="true" derivedAnchor="RFC8843">
<front>
<title>Negotiating Media Multiplexing Using the Session Description
Protocol (SDP)</title>
<author initials="C" surname="Holmberg" fullname="Christer Holmberg"
>
<organization showOnFrontPage="true"/>
</author>
<author initials="H" surname="Alvestrand" fullname="Harald Alvestran
d">
<organization showOnFrontPage="true"/>
</author>
<author initials="C" surname="Jennings" fullname="Cullen Jennings">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8843"/>
<seriesInfo name="DOI" value="10.17487/RFC8843"/>
</reference>
<reference anchor="RFC8851" target="https://www.rfc-editor.org/info/rfc8
851" quoteTitle="true" derivedAnchor="RFC8851">
<front>
<title>RTP Payload Format Restrictions</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach" role="
editor">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8851"/>
<seriesInfo name="DOI" value="10.17487/RFC8851"/>
</reference>
<reference anchor="RFC8852" target="https://www.rfc-editor.org/info/rfc8
852" quoteTitle="true" derivedAnchor="RFC8852">
<front>
<title>RTP Stream Identifier Source Description (SDES)</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach"/>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar
"/>
<author initials="P" surname="Thatcher" fullname="Peter Thatcher"/>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8852"/>
<seriesInfo name="DOI" value="10.17487/RFC8852"/>
</reference>
<reference anchor="RFC8859" target="https://www.rfc-editor.org/info/rfc8
859" quoteTitle="true" derivedAnchor="RFC8859">
<front>
<title>A Framework for Session Description Protocol (SDP) Attributes
When Multiplexing</title>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar
">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8859"/>
<seriesInfo name="DOI" value="10.17487/RFC8859"/>
</reference>
</references>
<references pn="section-11.2">
<name slugifiedName="name-informative-references">Informative References
</name>
<reference anchor="RFC2198" target="https://www.rfc-editor.org/info/rfc2
198" quoteTitle="true" derivedAnchor="RFC2198">
<front>
<title>RTP Payload for Redundant Audio Data</title>
<author initials="C." surname="Perkins" fullname="C. Perkins">
<organization showOnFrontPage="true"/>
</author>
<author initials="I." surname="Kouvelas" fullname="I. Kouvelas">
<organization showOnFrontPage="true"/>
</author>
<author initials="O." surname="Hodson" fullname="O. Hodson">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Hardman" fullname="V. Hardman">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Handley" fullname="M. Handley">
<organization showOnFrontPage="true"/>
</author>
<author initials="J.C." surname="Bolot" fullname="J.C. Bolot">
<organization showOnFrontPage="true"/>
</author>
<author initials="A." surname="Vega-Garcia" fullname="A. Vega-Garcia
">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Fosse-Parisis" fullname="S. Fosse-Par
isis">
<organization showOnFrontPage="true"/>
</author>
<date year="1997" month="September"/>
<abstract>
<t indent="0">This document describes a payload format for use wit
h the real-time transport protocol (RTP), version 2, for encoding redundant audi
o data. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="2198"/>
<seriesInfo name="DOI" value="10.17487/RFC2198"/>
</reference>
<reference anchor="RFC3389" target="https://www.rfc-editor.org/info/rfc3
389" quoteTitle="true" derivedAnchor="RFC3389">
<front>
<title>Real-time Transport Protocol (RTP) Payload for Comfort Noise
(CN)</title>
<author initials="R." surname="Zopf" fullname="R. Zopf">
<organization showOnFrontPage="true"/>
</author>
<date year="2002" month="September"/>
</front>
<seriesInfo name="RFC" value="3389"/>
<seriesInfo name="DOI" value="10.17487/RFC3389"/>
</reference>
<reference anchor="RFC4588" target="https://www.rfc-editor.org/info/rfc4
588" quoteTitle="true" derivedAnchor="RFC4588">
<front>
<title>RTP Retransmission Payload Format</title>
<author initials="J." surname="Rey" fullname="J. Rey">
<organization showOnFrontPage="true"/>
</author>
<author initials="D." surname="Leon" fullname="D. Leon">
<organization showOnFrontPage="true"/>
</author>
<author initials="A." surname="Miyazaki" fullname="A. Miyazaki">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Varsa" fullname="V. Varsa">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Hakenberg" fullname="R. Hakenberg">
<organization showOnFrontPage="true"/>
</author>
<date year="2006" month="July"/>
<abstract>
<t indent="0">RTP retransmission is an effective packet loss recov
ery technique for real-time applications with relaxed delay bounds. This docume
nt describes an RTP payload format for performing retransmissions. Retransmitted
RTP packets are sent in a separate stream from the original RTP stream. It is
assumed that feedback from receivers to senders is available. In particular, it
is assumed that Real-time Transport Control Protocol (RTCP) feedback as defined
in the extended RTP profile for RTCP-based feedback (denoted RTP/AVPF) is avail
able in this memo. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4588"/>
<seriesInfo name="DOI" value="10.17487/RFC4588"/>
</reference>
<reference anchor="RFC4733" target="https://www.rfc-editor.org/info/rfc4
733" quoteTitle="true" derivedAnchor="RFC4733">
<front>
<title>RTP Payload for DTMF Digits, Telephony Tones, and Telephony S
ignals</title>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne
">
<organization showOnFrontPage="true"/>
</author>
<author initials="T." surname="Taylor" fullname="T. Taylor">
<organization showOnFrontPage="true"/>
</author>
<date year="2006" month="December"/>
<abstract>
<t indent="0">This memo describes how to carry dual-tone multifreq
uency (DTMF) signalling, other tone signals, and telephony events in RTP packets
. It obsoletes RFC 2833.</t>
<t indent="0">This memo captures and expands upon the basic framew
ork defined in RFC 2833, but retains only the most basic event codes. It sets u
p an IANA registry to which other event code assignments may be added. Companion
documents add event codes to this registry relating to modem, fax, text telepho
ny, and channel-associated signalling events. The remainder of the event codes d
efined in RFC 2833 are conditionally reserved in case other documents revive the
ir use.</t>
<t indent="0">This document provides a number of clarifications to
the original document. However, it specifically differs from RFC 2833 by remov
ing the requirement that all compliant implementations support the DTMF events.
Instead, compliant implementations taking part in out-of-band negotiations of m
edia stream content indicate what events they support. This memo adds three new
procedures to the RFC 2833 framework: subdivision of long events into segments,
reporting of multiple events in a single packet, and the concept and reporting
of state events. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4733"/>
<seriesInfo name="DOI" value="10.17487/RFC4733"/>
</reference>
<reference anchor="RFC5104" target="https://www.rfc-editor.org/info/rfc5
104" quoteTitle="true" derivedAnchor="RFC5104">
<front>
<title>Codec Control Messages in the RTP Audio-Visual Profile with F
eedback (AVPF)</title>
<author initials="S." surname="Wenger" fullname="S. Wenger">
<organization showOnFrontPage="true"/>
</author>
<author initials="U." surname="Chandra" fullname="U. Chandra">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<author initials="B." surname="Burman" fullname="B. Burman">
<organization showOnFrontPage="true"/>
</author>
<date year="2008" month="February"/>
<abstract>
<t indent="0">This document specifies a few extensions to the mess
ages defined in the Audio-Visual Profile with Feedback (AVPF). They are helpful
primarily in conversational multimedia scenarios where centralized multipoint f
unctionalities are in use. However, some are also usable in smaller multicast e
nvironments and point-to-point calls.</t>
<t indent="0">The extensions discussed are messages related to the
ITU-T Rec. H.271 Video Back Channel, Full Intra Request, Temporary Maximum Medi
a Stream Bit Rate, and Temporal-Spatial Trade-off. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5104"/>
<seriesInfo name="DOI" value="10.17487/RFC5104"/>
</reference>
<reference anchor="RFC5109" target="https://www.rfc-editor.org/info/rfc5
109" quoteTitle="true" derivedAnchor="RFC5109">
<front>
<title>RTP Payload Format for Generic Forward Error Correction</titl
e>
<author initials="A." surname="Li" fullname="A. Li" role="editor">
<organization showOnFrontPage="true"/>
</author>
<date year="2007" month="December"/>
<abstract>
<t indent="0">This document specifies a payload format for generic
Forward Error Correction (FEC) for media data encapsulated in RTP. It is based
on the exclusive-or (parity) operation. The payload format described in this d
ocument allows end systems to apply protection using various protection lengths
and levels, in addition to using various protection group sizes to adapt to diff
erent media and channel characteristics. It enables complete recovery of the pro
tected packets or partial recovery of the critical parts of the payload dependin
g on the packet loss situation. This scheme is completely compatible with non-F
EC-capable hosts, so the receivers in a multicast group that do not implement FE
C can still work by simply ignoring the protection data. This specification obs
oletes RFC 2733 and RFC 3009. The FEC specified in this document is not backwar
d compatible with RFC 2733 and RFC 3009. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5109"/>
<seriesInfo name="DOI" value="10.17487/RFC5109"/>
</reference>
<reference anchor="RFC5583" target="https://www.rfc-editor.org/info/rfc5
583" quoteTitle="true" derivedAnchor="RFC5583">
<front>
<title>Signaling Media Decoding Dependency in the Session Descriptio
n Protocol (SDP)</title>
<author initials="T." surname="Schierl" fullname="T. Schierl">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Wenger" fullname="S. Wenger">
<organization showOnFrontPage="true"/>
</author>
<date year="2009" month="July"/>
<abstract>
<t indent="0">This memo defines semantics that allow for signaling
the decoding dependency of different media descriptions with the same media typ
e in the Session Description Protocol (SDP). This is required, for example, if
media data is separated and transported in different network streams as a result
of the use of a layered or multiple descriptive media coding process.</t>
<t indent="0">A new grouping type "DDP" -- decoding dependency --
is defined, to be used in conjunction with RFC 3388 entitled "Grouping of Media
Lines in the Session Description Protocol". In addition, an attribute is specif
ied describing the relationship of the media streams in a "DDP" group indicated
by media identification attribute(s) and media format description(s). [STANDARD
S-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5583"/>
<seriesInfo name="DOI" value="10.17487/RFC5583"/>
</reference>
<reference anchor="RFC6184" target="https://www.rfc-editor.org/info/rfc6
184" quoteTitle="true" derivedAnchor="RFC6184">
<front>
<title>RTP Payload Format for H.264 Video</title>
<author initials="Y.-K." surname="Wang" fullname="Y.-K. Wang">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Even" fullname="R. Even">
<organization showOnFrontPage="true"/>
</author>
<author initials="T." surname="Kristensen" fullname="T. Kristensen">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Jesup" fullname="R. Jesup">
<organization showOnFrontPage="true"/>
</author>
<date year="2011" month="May"/>
<abstract>
<t indent="0">This memo describes an RTP Payload format for the IT
U-T Recommendation H.264 video codec and the technically identical ISO/IEC Inter
national Standard 14496-10 video codec, excluding the Scalable Video Coding (SVC
) extension and the Multiview Video Coding extension, for which the RTP payload
formats are defined elsewhere. The RTP payload format allows for packetization o
f one or more Network Abstraction Layer Units (NALUs), produced by an H.264 vide
o encoder, in each RTP payload. The payload format has wide applicability, as i
t supports applications from simple low bitrate conversational usage, to Interne
t video streaming with interleaved transmission, to high bitrate video-on-demand
.</t>
<t indent="0">This memo obsoletes RFC 3984. Changes from RFC 3984
are summarized in Section 14. Issues on backward compatibility to RFC 3984 are
discussed in Section 15. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="6184"/>
<seriesInfo name="DOI" value="10.17487/RFC6184"/>
</reference>
<reference anchor="RFC6190" target="https://www.rfc-editor.org/info/rfc6
190" quoteTitle="true" derivedAnchor="RFC6190">
<front>
<title>RTP Payload Format for Scalable Video Coding</title>
<author initials="S." surname="Wenger" fullname="S. Wenger">
<organization showOnFrontPage="true"/>
</author>
<author initials="Y.-K." surname="Wang" fullname="Y.-K. Wang">
<organization showOnFrontPage="true"/>
</author>
<author initials="T." surname="Schierl" fullname="T. Schierl">
<organization showOnFrontPage="true"/>
</author>
<author initials="A." surname="Eleftheriadis" fullname="A. Eleftheri
adis">
<organization showOnFrontPage="true"/>
</author>
<date year="2011" month="May"/>
<abstract>
<t indent="0">This memo describes an RTP payload format for Scalab
le Video Coding (SVC) as defined in Annex G of ITU-T Recommendation H.264, which
is technically identical to Amendment 3 of ISO/IEC International Standard 14496
-10. The RTP payload format allows for packetization of one or more Network Abs
traction Layer (NAL) units in each RTP packet payload, as well as fragmentation
of a NAL unit in multiple RTP packets. Furthermore, it supports transmission of
an SVC stream over a single as well as multiple RTP sessions. The payload forma
t defines a new media subtype name "H264-SVC", but is still backward compatible
to RFC 6184 since the base layer, when encapsulated in its own RTP stream, must
use the H.264 media subtype name ("H264") and the packetization method specified
in RFC 6184. The payload format has wide applicability in videoconferencing, I
nternet video streaming, and high-bitrate entertainment-quality video, among oth
ers. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="6190"/>
<seriesInfo name="DOI" value="10.17487/RFC6190"/>
</reference>
<reference anchor="RFC6236" target="https://www.rfc-editor.org/info/rfc6
236" quoteTitle="true" derivedAnchor="RFC6236">
<front>
<title>Negotiation of Generic Image Attributes in the Session Descri
ption Protocol (SDP)</title>
<author initials="I." surname="Johansson" fullname="I. Johansson">
<organization showOnFrontPage="true"/>
</author>
<author initials="K." surname="Jung" fullname="K. Jung">
<organization showOnFrontPage="true"/>
</author>
<date year="2011" month="May"/>
<abstract>
<t indent="0">This document proposes a new generic session setup a
ttribute to make it possible to negotiate different image attributes such as ima
ge size. A possible use case is to make it possible for a \%low-end \%hand- hel
d terminal to display video without the need to rescale the image, something tha
t may consume large amounts of memory and processing power. The document also h
elps to maintain an optimal bitrate for video as only the image size that is des
ired by the receiver is transmitted. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="6236"/>
<seriesInfo name="DOI" value="10.17487/RFC6236"/>
</reference>
<reference anchor="RFC6464" target="https://www.rfc-editor.org/info/rfc6
464" quoteTitle="true" derivedAnchor="RFC6464">
<front>
<title>A Real-time Transport Protocol (RTP) Header Extension for Cli
ent-to-Mixer Audio Level Indication</title>
<author initials="J." surname="Lennox" fullname="J. Lennox" role="ed
itor">
<organization showOnFrontPage="true"/>
</author>
<author initials="E." surname="Ivov" fullname="E. Ivov">
<organization showOnFrontPage="true"/>
</author>
<author initials="E." surname="Marocco" fullname="E. Marocco">
<organization showOnFrontPage="true"/>
</author>
<date year="2011" month="December"/>
<abstract>
<t indent="0">This document defines a mechanism by which packets o
f Real-time Transport Protocol (RTP) audio streams can indicate, in an RTP heade
r extension, the audio level of the audio sample carried in the RTP packet. In
large conferences, this can reduce the load on an audio mixer or other middlebox
that wants to forward only a few of the loudest audio streams, without requirin
g it to decode and measure every stream that is received. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="6464"/>
<seriesInfo name="DOI" value="10.17487/RFC6464"/>
</reference>
<reference anchor="RFC7104" target="https://www.rfc-editor.org/info/rfc7
104" quoteTitle="true" derivedAnchor="RFC7104">
<front>
<title>Duplication Grouping Semantics in the Session Description Pro
tocol</title>
<author initials="A." surname="Begen" fullname="A. Begen">
<organization showOnFrontPage="true"/>
</author>
<author initials="Y." surname="Cai" fullname="Y. Cai">
<organization showOnFrontPage="true"/>
</author>
<author initials="H." surname="Ou" fullname="H. Ou">
<organization showOnFrontPage="true"/>
</author>
<date year="2014" month="January"/>
<abstract>
<t indent="0">Packet loss is undesirable for real-time multimedia
sessions, but it can occur due to congestion or other unplanned network outages.
This is especially true for IP multicast networks, where packet loss patterns
can vary greatly between receivers. One technique that can be used to recover f
rom packet loss without incurring unbounded delay for all the receivers is to du
plicate the packets and send them in separate redundant streams. This document
defines the semantics for grouping redundant streams in the Session Description
Protocol (SDP). The semantics defined in this document are to be used with the S
DP Grouping Framework. Grouping semantics at the Synchronization Source (SSRC)
level are also defined in this document for RTP streams using SSRC multiplexing.
</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7104"/>
<seriesInfo name="DOI" value="10.17487/RFC7104"/>
</reference>
<reference anchor="RFC7656" target="https://www.rfc-editor.org/info/rfc7
656" quoteTitle="true" derivedAnchor="RFC7656">
<front>
<title>A Taxonomy of Semantics and Mechanisms for Real-Time Transpor
t Protocol (RTP) Sources</title>
<author initials="J." surname="Lennox" fullname="J. Lennox">
<organization showOnFrontPage="true"/>
</author>
<author initials="K." surname="Gross" fullname="K. Gross">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Nandakumar" fullname="S. Nandakumar">
<organization showOnFrontPage="true"/>
</author>
<author initials="G." surname="Salgueiro" fullname="G. Salgueiro">
<organization showOnFrontPage="true"/>
</author>
<author initials="B." surname="Burman" fullname="B. Burman" role="ed
itor">
<organization showOnFrontPage="true"/>
</author>
<date year="2015" month="November"/>
<abstract>
<t indent="0">The terminology about, and associations among, Real-
time Transport Protocol (RTP) sources can be complex and somewhat opaque. This
document describes a number of existing and proposed properties and relationship
s among RTP sources and defines common terminology for discussing protocol entit
ies and their relationships.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7656"/>
<seriesInfo name="DOI" value="10.17487/RFC7656"/>
</reference>
<reference anchor="RFC7667" target="https://www.rfc-editor.org/info/rfc7
667" quoteTitle="true" derivedAnchor="RFC7667">
<front>
<title>RTP Topologies</title>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Wenger" fullname="S. Wenger">
<organization showOnFrontPage="true"/>
</author>
<date year="2015" month="November"/>
<abstract>
<t indent="0">This document discusses point-to-point and multi-end
point topologies used in environments based on the Real-time Transport Protocol
(RTP). In particular, centralized topologies commonly employed in the video conf
erencing industry are mapped to the RTP terminology.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7667"/>
<seriesInfo name="DOI" value="10.17487/RFC7667"/>
</reference>
<reference anchor="RFC7741" target="https://www.rfc-editor.org/info/rfc7
741" quoteTitle="true" derivedAnchor="RFC7741">
<front>
<title>RTP Payload Format for VP8 Video</title>
<author initials="P." surname="Westin" fullname="P. Westin">
<organization showOnFrontPage="true"/>
</author>
<author initials="H." surname="Lundin" fullname="H. Lundin">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Glover" fullname="M. Glover">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Uberti" fullname="J. Uberti">
<organization showOnFrontPage="true"/>
</author>
<author initials="F." surname="Galligan" fullname="F. Galligan">
<organization showOnFrontPage="true"/>
</author>
<date year="2016" month="March"/>
<abstract>
<t indent="0">This memo describes an RTP payload format for the VP
8 video codec. The payload format has wide applicability, as it supports applica
tions from low-bitrate peer-to-peer usage to high-bitrate video conferences.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7741"/>
<seriesInfo name="DOI" value="10.17487/RFC7741"/>
</reference>
<reference anchor="RFC7941" target="https://www.rfc-editor.org/info/rfc7
941" quoteTitle="true" derivedAnchor="RFC7941">
<front>
<title>RTP Header Extension for the RTP Control Protocol (RTCP) Sour
ce Description Items</title>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<author initials="B." surname="Burman" fullname="B. Burman">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Even" fullname="R. Even">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Zanaty" fullname="M. Zanaty">
<organization showOnFrontPage="true"/>
</author>
<date year="2016" month="August"/>
<abstract>
<t indent="0">Source Description (SDES) items are normally transpo
rted in the RTP Control Protocol (RTCP). In some cases, it can be beneficial to
speed up the delivery of these items. The main case is when a new synchronizat
ion source (SSRC) joins an RTP session and the receivers need this source's iden
tity, relation to other sources, or its synchronization context, all of which ma
y be fully or partially identified using SDES items. To enable this optimizatio
n, this document specifies a new RTP header extension that can carry SDES items.
</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7941"/>
<seriesInfo name="DOI" value="10.17487/RFC7941"/>
</reference>
<reference anchor="RFC8108" target="https://www.rfc-editor.org/info/rfc8
108" quoteTitle="true" derivedAnchor="RFC8108">
<front>
<title>Sending Multiple RTP Streams in a Single RTP Session</title>
<author initials="J." surname="Lennox" fullname="J. Lennox">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<author initials="Q." surname="Wu" fullname="Q. Wu">
<organization showOnFrontPage="true"/>
</author>
<author initials="C." surname="Perkins" fullname="C. Perkins">
<organization showOnFrontPage="true"/>
</author>
<date year="2017" month="March"/>
<abstract>
<t indent="0">This memo expands and clarifies the behavior of Real
-time Transport Protocol (RTP) endpoints that use multiple synchronization sourc
es (SSRCs). This occurs, for example, when an endpoint sends multiple RTP strea
ms in a single RTP session. This memo updates RFC 3550 with regard to handling
multiple SSRCs per endpoint in RTP sessions, with a particular focus on RTP Cont
rol Protocol (RTCP) behavior. It also updates RFC 4585 to change and clarify th
e calculation of the timeout of SSRCs and the inclusion of feedback messages.</t
>
</abstract>
</front>
<seriesInfo name="RFC" value="8108"/>
<seriesInfo name="DOI" value="10.17487/RFC8108"/>
</reference>
<reference anchor="RFC8627" target="https://www.rfc-editor.org/info/rfc8
627" quoteTitle="true" derivedAnchor="RFC8627">
<front>
<title>RTP Payload Format for Flexible Forward Error Correction (FEC
)</title>
<author initials="M." surname="Zanaty" fullname="M. Zanaty">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Singh" fullname="V. Singh">
<organization showOnFrontPage="true"/>
</author>
<author initials="A." surname="Begen" fullname="A. Begen">
<organization showOnFrontPage="true"/>
</author>
<author initials="G." surname="Mandyam" fullname="G. Mandyam">
<organization showOnFrontPage="true"/>
</author>
<date year="2019" month="July"/>
<abstract>
<t indent="0">This document defines new RTP payload formats for th
e Forward Error Correction (FEC) packets that are generated by the non-interleav
ed and interleaved parity codes from source media encapsulated in RTP. These par
ity codes are systematic codes (Flexible FEC, or "FLEX F
EC"), where a number of FEC repair packets are generated from a set of source pa
ckets from one or more source RTP streams. These FEC repair packets are sent in
a redundancy RTP stream separate from the source RTP stream(s) that carries the
source packets. RTP source packets that were lost in transmission can be recon
structed using the source and repair packets that were received. The non-interl
eaved and interleaved parity codes that are defined in this specification offer
a good protection against random and bursty packet losses, respectively, at a co
st of complexity. The RTP payload formats that are defined in this document add
ress scalability issues experienced with the earlier specifications and offer se
veral improvements. Due to these changes, the new payload formats are not backw
ard compatible with earlier specifications; however, endpoints that do not imple
ment this specification can still work by simply ignoring the FEC repair packets
.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="8627"/>
<seriesInfo name="DOI" value="10.17487/RFC8627"/>
</reference>
<reference anchor="RFC8872" target="https://www.rfc-editor.org/info/rfc8
872" quoteTitle="true" derivedAnchor="RFC8872">
<front>
<title>Guidelines for Using the Multiplexing Features of RTP to Supp
ort Multiple Media Streams</title>
<author initials="M" surname="Westerlund" fullname="Magnus Westerlun
d">
<organization showOnFrontPage="true"/>
</author>
<author initials="B" surname="Burman" fullname="Bo Burman">
<organization showOnFrontPage="true"/>
</author>
<author initials="C" surname="Perkins" fullname="Colin Perkins">
<organization showOnFrontPage="true"/>
</author>
<author initials="H" surname="Alvestrand" fullname="Harald Alvestran
d">
<organization showOnFrontPage="true"/>
</author>
<author initials="R" surname="Even" fullname="Roni Even">
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8872"/>
<seriesInfo name="DOI" value="10.17487/RFC8872"/>
</reference>
</references>
</references> </references>
<section anchor="sec-requirements" numbered="true" toc="include" removeInRFC
<section anchor="sec-requirements" title="Requirements"> ="false" pn="section-appendix.a">
<t>The following requirements are met by the defined solution to support <name slugifiedName="name-requirements">Requirements</name>
the <xref target="sec-use-cases">use cases</xref>:<list style="hanging"> <t indent="0" pn="section-appendix.a-1">The following requirements are met
<t anchor="req-1" hangText="REQ-1:">Identification:<list by the defined solution to support
style="hanging"> the <xref target="sec-use-cases" format="default" sectionFormat="of" deriv
<t anchor="req-1.1" hangText="REQ-1.1:">It must be possible to edContent="Section 3">use cases</xref>:</t>
<dl newline="false" spacing="normal" indent="3" pn="section-appendix.a-2">
<dt pn="section-appendix.a-2.1">REQ-1:</dt>
<dd anchor="req-1" pn="section-appendix.a-2.2">
<t indent="0" pn="section-appendix.a-2.2.1">Identification:</t>
<dl newline="false" spacing="normal" indent="3" pn="section-appendix.a
-2.2.2">
<dt pn="section-appendix.a-2.2.2.1">REQ-1.1:</dt>
<dd anchor="req-1.1" pn="section-appendix.a-2.2.2.2">It must be poss
ible to
identify a set of simulcasted RTP streams as originating from identify a set of simulcasted RTP streams as originating from
the same media source in SDP signaling.</t> the same media source in SDP signaling.</dd>
<dt pn="section-appendix.a-2.2.2.3">REQ-1.2:</dt>
<t anchor="req-1.2" hangText="REQ-1.2:">An RTP endpoint must be <dd anchor="req-1.2" pn="section-appendix.a-2.2.2.4">An RTP endpoint
capable of identifying the simulcast stream a received RTP 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 pn="section-appendix.a-2.3">REQ-2:</dt>
must work when using:<list style="hanging"> <dd anchor="req-2" pn="section-appendix.a-2.4">
<t anchor="req-2.1" hangText="REQ-2.1:">Legacy SDP with separate <t indent="0" pn="section-appendix.a-2.4.1">Transport usage. The solut
media transports per SDP media description.</t> ion
must work when using:</t>
<t anchor="req-2.2" hangText="REQ-2.2:"><xref <dl newline="false" spacing="normal" indent="3" pn="section-appendix.a
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Bundled</xref> -2.4.2">
SDP media descriptions.</t> <dt pn="section-appendix.a-2.4.2.1">REQ-2.1:</dt>
</list></t> <dd anchor="req-2.1" pn="section-appendix.a-2.4.2.2">Legacy SDP with
separate
<t anchor="req-3" hangText="REQ-3:">Capability negotiation. It must media transports per SDP media description.</dd>
be possible that:<list style="hanging"> <dt pn="section-appendix.a-2.4.2.3">REQ-2.2:</dt>
<t anchor="req-3.1" hangText="REQ-3.1:">Sender can express <dd anchor="req-2.2" pn="section-appendix.a-2.4.2.4">
capability of sending simulcast.</t> <xref target="RFC8843" format="default" sectionFormat="of" derived
Content="RFC8843">Bundled</xref>
<t anchor="req-3.2" hangText="REQ-3.2:">Receiver can express SDP media descriptions.</dd>
capability of receiving simulcast.</t> </dl>
</dd>
<t anchor="req-3.3" hangText="REQ-3.3:">Sender can express <dt pn="section-appendix.a-2.5">REQ-3:</dt>
maximum number of simulcast streams that can be provided.</t> <dd anchor="req-3" pn="section-appendix.a-2.6">
<t indent="0" pn="section-appendix.a-2.6.1">Capability negotiation. Th
<t anchor="req-3.4" hangText="REQ-3.4:">Receiver can express e
maximum number of simulcast streams that can be received.</t> following must be possible:</t>
<dl newline="false" spacing="normal" indent="3" pn="section-appendix.a
<t anchor="req-3.5" hangText="REQ-3.5:">Sender can detail the -2.6.2">
<dt pn="section-appendix.a-2.6.2.1">REQ-3.1:</dt>
<dd anchor="req-3.1" pn="section-appendix.a-2.6.2.2">The sender can
express
capability of sending simulcast.</dd>
<dt pn="section-appendix.a-2.6.2.3">REQ-3.2:</dt>
<dd anchor="req-3.2" pn="section-appendix.a-2.6.2.4">The receiver ca
n express
capability of receiving simulcast.</dd>
<dt pn="section-appendix.a-2.6.2.5">REQ-3.3:</dt>
<dd anchor="req-3.3" pn="section-appendix.a-2.6.2.6">The sender can
express
the maximum number of simulcast streams that can be
provided.</dd>
<dt pn="section-appendix.a-2.6.2.7">REQ-3.4:</dt>
<dd anchor="req-3.4" pn="section-appendix.a-2.6.2.8">The receiver ca
n express the
maximum number of simulcast streams that can be received.</dd>
<dt pn="section-appendix.a-2.6.2.9">REQ-3.5:</dt>
<dd anchor="req-3.5" pn="section-appendix.a-2.6.2.10">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 pn="section-appendix.a-2.6.2.11">REQ-3.6:</dt>
<t anchor="req-3.6" hangText="REQ-3.6:">Receiver can detail the <dd anchor="req-3.6" pn="section-appendix.a-2.6.2.12">The receiver c
an 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 pn="section-appendix.a-2.7">REQ-4:</dt>
<dd anchor="req-4" pn="section-appendix.a-2.8">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 pn="section-appendix.a-2.9">REQ-5:</dt>
<t anchor="req-5" hangText="REQ-5:">Compatibility. It must be <dd anchor="req-5" pn="section-appendix.a-2.10">
<t indent="0" pn="section-appendix.a-2.10.1">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" indent="3" pn="section-appendix.a
target="RFC4588">RTP Retransmission</xref>.</t> -2.10.2">
<dt pn="section-appendix.a-2.10.2.1">REQ-5.1:</dt>
<t anchor="req-5.2" hangText="REQ-5.2:"><xref <dd anchor="req-5.1" pn="section-appendix.a-2.10.2.2">
target="RFC5109">RTP Forward Error Correction</xref>.</t> <xref target="RFC4588" format="default" sectionFormat="of" derived
Content="RFC4588">RTP retransmission</xref>.</dd>
<t anchor="req-5.3" hangText="REQ-5.3:">Related payload types <dt pn="section-appendix.a-2.10.2.3">REQ-5.2:</dt>
such as audio Comfort Noise and/or DTMF.</t> <dd anchor="req-5.2" pn="section-appendix.a-2.10.2.4">
<xref target="RFC5109" format="default" sectionFormat="of" derived
<t hangText="REQ-5.4:">A single simulcast stream can consist of Content="RFC5109">RTP Forward Error Correction</xref>.</dd>
<dt pn="section-appendix.a-2.10.2.5">REQ-5.3:</dt>
<dd anchor="req-5.3" pn="section-appendix.a-2.10.2.6">Related payloa
d types
such as audio Comfort Noise and/or DTMF.</dd>
<dt pn="section-appendix.a-2.10.2.7">REQ-5.4:</dt>
<dd pn="section-appendix.a-2.10.2.8">A single simulcast stream can c
onsist 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 pn="section-appendix.a-2.11">REQ-6:</dt>
must be possible to use in:<list style="hanging"> <dd anchor="req-6" pn="section-appendix.a-2.12">
<t anchor="req-6.1" hangText="REQ-6.1:">Interworking with <t indent="0" pn="section-appendix.a-2.12.1">Interoperability. The sol
non-simulcast legacy clients using a single media source per ution
media type.</t> must be possible to use in:</t>
<dl newline="false" spacing="normal" indent="3" pn="section-appendix.a
<t anchor="req-6.2" hangText="REQ-6.2:">WebRTC environment with -2.12.2">
a single media source per SDP media description.</t> <dt pn="section-appendix.a-2.12.2.1">REQ-6.1:</dt>
</list></t> <dd anchor="req-6.1" pn="section-appendix.a-2.12.2.2">Interworking w
</list></t> ith
nonsimulcast legacy clients using a single media source per
media type.</dd>
<dt pn="section-appendix.a-2.12.2.3">REQ-6.2:</dt>
<dd anchor="req-6.2" pn="section-appendix.a-2.12.2.4">WebRTC environ
ment with
a single media source per SDP media description.</dd>
</dl>
</dd>
</dl>
</section> </section>
<section anchor="sec-ack" numbered="false" toc="include" removeInRFC="false"
<section title="Changes From Earlier Versions"> pn="section-appendix.b">
<t>NOTE TO RFC EDITOR: Please remove this section prior to <name slugifiedName="name-acknowledgements">Acknowledgements</name>
publication.</t> <t indent="0" pn="section-appendix.b-1">The authors would like to thank <c
ontact fullname="Bernard Aboba"/>, <contact fullname="Thomas Belling"/>, <contac
<section title="Modifications Between WG Version -13 and -14"> t fullname="Roni Even"/>, <contact fullname="Adam Roach"/>, <contact fullname="I
<t><list style="symbols"> ñaki Baz Castillo"/>,
<t>c= and t= line order corrected in SDP examples</t> <contact fullname="Paul Kyzivat"/>, and <contact fullname="Arun Arun
</list></t> achalam"/> for the feedback they provided during the development of
</section> this document.</t>
</section>
<section title="Modifications Between WG Version -12 and -13"> <section anchor="sec-contributors" numbered="false" toc="include" removeInRF
<t><list style="symbols"> C="false" pn="section-appendix.c">
<t>Examples corrected to follow RID ABNF</t> <name slugifiedName="name-contributors">Contributors</name>
<t indent="0" pn="section-appendix.c-1"><contact fullname="Morgan Lindqvis
<t>Example <xref target="fig-ms-offer"/> now comments on priority t"/> and <contact fullname="Fredrik Jansson"/>, both from Ericsson, have c
for second media source.</t> ontributed with important material
to the first draft versions of this document. <contact fullname="Robert
<t>Clarified a SHOULD limitation.</t> Hanton"/> and <contact fullname="Cullen Jennings"/> from Cisco, <contact ful
lname="Peter Thatcher"/> from Google, and <contact fullname="Adam Roach"/>
<t>Added urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id in from Mozilla contributed significantly to subsequent
examples with RTX.</t> versions.</t>
</section>
<t>ABNF now uses RFC 7405 to indicate case sensitivity</t> <section anchor="authors-addresses" numbered="false" removeInRFC="false" toc
="include" pn="section-appendix.d">
<t>Various minor editorials and nits.</t> <name slugifiedName="name-authors-addresses">Authors' Addresses</name>
</list></t> <author fullname="Bo Burman" initials="B." surname="Burman">
</section> <organization showOnFrontPage="true">Ericsson</organization>
<address>
<section title="Modifications Between WG Version -11 and -12"> <postal>
<t><list style="symbols"> <street>Gronlandsgatan 31</street>
<t>Modified Normative statement regarding RTP stream duplication <city>SE-164 60 Stockholm</city>
in Section 5.2.</t> <region/>
<code/>
<t>Clarified assumption about use of congestion control by <country>Sweden</country>
applications.</t> </postal>
<phone/>
<t>Changed to use RFC 8174 boilerplate instead of RFC 2119.</t> <email>bo.burman@ericsson.com</email>
<uri/>
<t>Clarified explanation of syntax for simulcast attribute in </address>
Section 4.</t> </author>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<t>Editorial clarification in Section 5.2 and 5.3.2.</t> <organization showOnFrontPage="true">Ericsson</organization>
<address>
<t>Various minor editorials and nits.</t> <postal>
</list></t> <street>Torshamnsgatan 23</street>
</section> <city>SE-164 83 Stockholm</city>
<country>Sweden</country>
<section title="Modifications Between WG Version -10 and -11"> </postal>
<t><list style="symbols"> <email>magnus.westerlund@ericsson.com</email>
<t>Added new SDP example section on Simulcast and Redundancy, </address>
including both RED (RFC2198), RTP RTX (RFC4588), and FEC </author>
(draft-ietf-payload-flexible-fec-scheme).</t> <author fullname="Suhas Nandakumar" initials="S." surname="Nandakumar">
<organization showOnFrontPage="true">Cisco</organization>
<t>Removed restriction that "related" payload formats in an RTP <address>
stream (such as CN and DTMF) must not have their own rid-id, since <postal>
there is no reason to forbid this and corresponding clarification <street>170 West Tasman Drive</street>
is made in draft-ietf-mmusic-rid.</t> <city>San Jose</city>
<region>CA</region>
<t>Removed any mention of source-specific signaling and the <code>95134</code>
reference to RFC5576, since draft-ietf-mmusic-rid is not defined <country>United States of America</country>
for source-specific signaling.</t> </postal>
<phone/>
<t>Changed some SDP examples to use a=rid restrictions instead of <email>snandaku@cisco.com</email>
a=imageattr.</t> <uri/>
</address>
<t>Changed reference from the obsoleted RFC 5285 to RFC 8285.</t> </author>
</list></t> <author fullname="Mo Zanaty" initials="M." surname="Zanaty">
</section> <organization showOnFrontPage="true">Cisco</organization>
<address>
<section title="Modifications Between WG Version -09 and -10"> <postal>
<t><list style="symbols"> <street>170 West Tasman Drive</street>
<t>Amended overview section with a bit more explanation on the <city>San Jose</city>
examples, and added an rid-id alternative for one of the <region>CA</region>
streams.</t> <code>95134</code>
<country>United States of America</country>
<t>Removed SCID also from the Terminology section, which was </postal>
forgotten in -09 when changing SCID to rid-id.</t> <phone/>
</list></t> <email>mzanaty@cisco.com</email>
</section> <uri/>
</address>
<section title="Modifications Between WG Version -08 and -09"> </author>
<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
n -00">
<t><list style="symbols">
<t>Added this appendix.</t>
</list></t>
</section>
</section> </section>
</back> </back>
</rfc> </rfc>
 End of changes. 257 change blocks. 
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