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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" ipr="trust200902" docName="draft-shin-avtcore-rtp-multi-opus-04" category="info" submissionType="IETF" tocInclude="true" sortRefs="true" symRefs="true" version="3">
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  <front>
    <title abbrev="OPUS-MULTISTREAM">RTP/SDP Signaling for Opus Multistream Operation</title>
    <seriesInfo name="Internet-Draft" value="draft-shin-avtcore-rtp-multi-opus-04"/>
    <author initials="S." surname="Shin" fullname="Sun Shin">
      <organization>NVIDIA</organization>
      <address>
        <email>sushin@nvidia.com</email>
      </address>
    </author>
    <date year="2026" month="October" day="14"/>
    <area>Web and Internet Transport</area>
    <workgroup>Audio/Video Transport Core Maintenance</workgroup>
    <keyword>Internet-Draft</keyword>
    <abstract>
      <?line 32?>

<t>This document specifies RTP/SDP signaling for Opus multistream (multichannel) operation,
enabling negotiation of layouts such as 5.1 and 7.1 in real-time communications.
It defines SDP format parameters for describing multistream configurations and specifies
Offer/Answer procedures for interoperable negotiation of multichannel Opus sessions.
This document does not define new Opus codec behavior. Rather, it standardizes
SDP signaling for Opus multistream operation while continuing to use the Opus RTP payload
format defined in <xref target="RFC7587"/> and the channel mapping semantics defined in <xref target="RFC7845"/>.
The signaling procedures specified in this document are intended to provide
a standardized alternative to deployment-specific approaches that have historically
been referred to as "multiopus".</t>
    </abstract>
    <note removeInRFC="true">
      <name>Discussion Venues</name>
      <t>Discussion of this document takes place on the
    Audio/Video Transport Core Maintenance Working Group mailing list (avt@ietf.org),
    which is archived at <eref target="https://mailarchive.ietf.org/arch/browse/avt/"/>.</t>
      <t>Source for this draft and an issue tracker can be found at
    <eref target="https://github.com/xingri/multi-opus"/>.</t>
    </note>
  </front>
  <middle>
    <?line 45?>

<section anchor="introduction">
      <name>Introduction</name>
      <t>Opus (<xref target="RFC6716"/>) supports multistream encoding and channel mapping, enabling representation
of multichannel audio configurations beyond mono and stereo. The RTP payload format for Opus
defined in <xref target="RFC7587"/> specifies signaling and transport procedures for Opus in RTP sessions,
but primarily focuses on mono and stereo operation and does not define SDP signaling
procedures for multistream multichannel configurations.</t>
      <t><xref target="RFC7845"/> defines channel mapping families for Opus carried in the Ogg container format.
These mapping families assign semantic meaning to decoded output channels, such as speaker
positions for surround sound layouts. The channel mappings do not alter the Opus bitstream
or RTP payload format but define how decoded channels are to be interpreted by receivers.</t>
      <t>Several deployed implementations have supported multichannel Opus using implementation-specific
SDP signaling approaches, commonly referred to as "multiopus". While these approaches have
enabled deployment of surround-sound audio in practical systems, they have not been standardized
and may present interoperability challenges across implementations.</t>
      <t>This document specifies interoperable SDP signaling and Offer/Answer procedures for Opus
multistream operation in RTP sessions while continuing to use the Opus RTP payload format
defined in <xref target="RFC7587"/> and the channel mapping semantics defined in <xref target="RFC7845"/>.</t>
      <t>This document is limited to RTP/SDP signaling for Opus multistream configurations already
supported by <xref target="RFC6716"/>, <xref target="RFC7587"/>, and <xref target="RFC7845"/>. It does not define new Opus codec
features, bitstream formats, RTP payload formats, or decoding behavior.</t>
      <t>The MLCODEC Working Group is defining extensions to the Opus codec that may introduce
additional codec capabilities. Such extensions are outside the scope of this document.
Future Opus extensions may reuse the SDP signaling framework defined here, subject to
the requirements and constraints of their respective specifications.</t>
    </section>
    <section anchor="relationship-to-existing-rfcs">
      <name>Relationship to Existing RFCs</name>
      <t>This section summarizes the scope and relationship between <xref target="RFC6716"/> (Opus codec),
<xref target="RFC7587"/> (Opus over RTP), <xref target="RFC7845"/> (Ogg encapsulation of Opus), and this document.</t>
      <t><xref target="RFC6716"/> defines the Opus codec, including support for multistream encoding and
channel mapping.</t>
      <t><xref target="RFC7587"/> defines RTP transport and SDP signaling for Opus in RTP sessions but
primarily addresses mono and stereo operation.</t>
      <t><xref target="RFC7845"/> defines channel mapping families for Opus carried in the Ogg container format.
These mappings provide semantic interpretation of decoded output channels but do not
define RTP usage or SDP signaling procedures.</t>
      <t>This document specifies SDP signaling and Offer/Answer procedures for Opus multistream operation
in RTP sessions while continuing to use the RTP payload format defined in <xref target="RFC7587"/> and
the channel mapping semantics defined in <xref target="RFC7845"/>.</t>
    </section>
    <section anchor="relationship-of-rfcs-and-this-draft">
      <name>Relationship of RFCs and This Draft</name>
      <artwork><![CDATA[
  +----------------+     +-------------------+     +----------------+
  |   RFC 6716     |     |     RFC 7845      |     |   RFC 7587     |
  |  Opus Codec    |     | Ogg Encapsulation |     | Opus over RTP  |
  +----------------+     +-------------------+     +----------------+
           |                       |                        |
           |                       |                        |
           +-----------------------+------------------------+
                                   |
                                   v
                   +------------------------------+
                   |         This Document        |
                   |      Opus Multistream SDP    |
                   |      Signaling Framework     |
                   +------------------------------+
]]></artwork>
    </section>
    <section anchor="summary-of-rfcs-and-this-document">
      <name>Summary of RFCs and This Document</name>
      <artwork><![CDATA[
 +------------+------------------------+-----------------------+-----------------------+
 | RFC/Doc    | Primary Scope          | Channel Mapping       | SDP Signaling         |
 +------------+------------------------+-----------------------+-----------------------+
 | RFC 6716   | Opus codec             | Yes                   | No                    |
 | RFC 7845   | Ogg encapsulation      | Yes (families)        | No                    |
 | RFC 7587   | Opus RTP payload       | Mono/Stereo only      | Yes (mono/stereo)     |
 | This Doc   | Multistream SDP rules  | Reuses RFC 7845       | Yes (multichannel)    |
 +------------+------------------------+-----------------------+-----------------------+
]]></artwork>
    </section>
    <section anchor="overview-and-rationale">
      <name>Overview and Rationale</name>
      <t>Opus supports multistream operation and channel mappings that enable multichannel audio
configurations such as 5.1 and 7.1. While such configurations have been deployed in practice,
interoperable SDP signaling for these configurations has not been standardized for RTP sessions.</t>
      <t>Several deployed implementations (e.g., <xref target="libwebrtc"/> based) have historically used
implementation-specific SDP signaling approaches, commonly referred to as "multiopus",
together with parameters such as num_streams, coupled_streams, and channel_mapping.
Although these approaches have enabled deployment of multichannel Opus,
they are not standardized and may lead to interoperability challenges.</t>
      <t>This document standardizes SDP signaling semantics for Opus multistream operation while
continuing to use the Opus RTP payload format defined in <xref target="RFC7587"/>. Standardizing these semantics
improves interoperability and removes the need for application-specific SDP modifications.</t>
    </section>
    <section anchor="sdp-parameters-for-opus-multistream">
      <name>SDP Parameters for Opus Multistream</name>
      <section anchor="field-descriptions">
        <name>Field Descriptions</name>
        <ul spacing="normal">
          <li>
            <t><tt>a=rtpmap:&lt;pt&gt; opus/&lt;clock-rate&gt;/&lt;channel-count&gt;</tt>
            </t>
            <ul spacing="normal">
              <li>
                <t><tt>&lt;pt&gt;</tt>: Dynamic payload type (e.g., 96).</t>
              </li>
              <li>
                <t><tt>&lt;clock-rate&gt;</tt>: Fixed Opus RTP clock rate of 48000 Hz, as defined in
<xref target="RFC7587"/>. This placeholder represents the literal value <tt>48000</tt>
in actual SDP.</t>
              </li>
              <li>
                <t><tt>&lt;channel-count&gt;</tt>: Total number of output channels resulting from decoding.</t>
              </li>
            </ul>
          </li>
          <li>
            <t><tt>a=fmtp:&lt;pt&gt; num_streams=&lt;N&gt;;coupled_streams=&lt;M&gt;;channel_mapping=&lt;C&gt;</tt>
            </t>
            <ul spacing="normal">
              <li>
                <t><tt>&lt;pt&gt;</tt>: Same payload type as in the corresponding <tt>a=rtpmap</tt> attribute.</t>
              </li>
              <li>
                <t><tt>num_streams</tt>: Total number of Opus streams.</t>
              </li>
              <li>
                <t><tt>coupled_streams</tt>: Number of stereo (coupled) streams.</t>
              </li>
              <li>
                <t><tt>channel_mapping</tt>: Comma-separated list defining the mapping from decoded
channels to semantic output positions according to the channel mapping semantics
defined in <xref target="RFC7845"/>.</t>
              </li>
            </ul>
          </li>
        </ul>
      </section>
      <section anchor="mapping-families-and-extensibility">
        <name>Mapping Families and Extensibility</name>
        <t>Channel mapping families define the semantic interpretation of decoded output channels
and do not alter the Opus bitstream or RTP payload format.</t>
        <t>This document normatively specifies SDP signaling for Mapping Family 1
(channel-based speaker layouts) as defined in <xref target="RFC7845"/>.</t>
        <t>The signaling framework is designed to be extensible to additional mapping families,
including Ambisonics or externally defined mappings, without requiring changes to
the underlying Opus RTP payload format.</t>
        <t>An endpoint that receives an offer specifying a mapping family it does not support
MUST reject the corresponding payload type during Offer/Answer negotiation.</t>
      </section>
      <section anchor="field-presence-and-constraints">
        <name>Field Presence and Constraints</name>
        <t>The "channel_mapping" parameter MUST be present when the number of
output channels exceeds two.</t>
        <t>For configurations with one or two output channels, the channel_mapping parameter
MAY be omitted. If omitted, the receiver MUST interpret the configuration according
to Mapping Family 0 semantics defined in <xref target="RFC7845"/>.</t>
        <t>The values of "num_streams", "coupled_streams", and each entry in "channel_mapping"
MUST be integers in the range 0 to 255.</t>
        <t>For Mapping Family 1, the number of output channels MUST NOT exceed eight.</t>
      </section>
      <section anchor="sdp-examples">
        <name>SDP Examples</name>
        <section anchor="audio">
          <name>5.1 Audio</name>
          <t>The following example signals six output channels using four Opus streams,
two of which are coupled streams.</t>
          <t><tt>sdp
a=rtpmap:111 opus/48000/6
a=fmtp:111 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
</tt></t>
        </section>
        <section anchor="audio-1">
          <name>7.1 Audio</name>
          <t>The following example signals eight output channels using five Opus streams,
three of which are coupled streams.</t>
          <t><tt>sdp
a=rtpmap:111 opus/48000/8
a=fmtp:111 num_streams=5;coupled_streams=3;channel_mapping=0,6,1,2,3,4,5,7
</tt></t>
        </section>
      </section>
    </section>
    <section anchor="offeranswer-procedures">
      <name>Offer/Answer Procedures</name>
      <t>An offerer willing to negotiate multichannel Opus MAY include one or more
Opus multistream payload configurations with the appropriate format parameters.
For backward compatibility, an offerer SHOULD also include a stereo alternative
using <tt>opus/&lt;clock-rate&gt;/2</tt> as defined in <xref target="RFC7587"/>.</t>
      <t>An answerer MAY accept an offered multistream configuration if it can parse
the offered parameters and decode the resulting multichannel audio without loss
of the negotiated channel semantics.</t>
      <t>If the answerer supports the offered multistream Opus configuration, it MUST select
the corresponding payload type and include the selected parameters in the answer.</t>
      <t>If the offered configuration is not supported, the answerer MAY select a stereo
Opus payload type or reject the media section according to <xref target="RFC3264"/>.</t>
      <t>An answerer SHOULD NOT silently down-convert decoded output to stereo without
providing an indication of that behavior to the application.</t>
      <section anchor="examples">
        <name>Examples</name>
        <section anchor="offer-51-audio-6-channels">
          <name>Offer: 5.1 Audio (6 channels)</name>
          <t>The following offer advertises a 5.1 multistream Opus configuration using
payload type 111 and a stereo Opus fallback using payload type 112 for
backward compatibility.</t>
          <t><tt>sdp
m=audio 9 UDP/TLS/RTP/SAVPF 111 112
a=mid:audio
a=rtpmap:111 opus/48000/6
a=fmtp:111 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
a=rtpmap:112 opus/48000/2
a=sendrecv
</tt></t>
        </section>
        <section anchor="answer-accept-51-audio">
          <name>Answer: Accept 5.1 Audio</name>
          <t>The answer selects payload type 111 and echoes the negotiated multistream
parameters, indicating support for the offered 5.1-channel layout.</t>
          <t><tt>sdp
m=audio 9 UDP/TLS/RTP/SAVPF 111
a=mid:audio
a=rtpmap:111 opus/48000/6
a=fmtp:111 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
a=sendrecv
</tt></t>
        </section>
      </section>
    </section>
    <section anchor="security-considerations">
      <name>Security Considerations</name>
      <t>The SDP signaling procedures defined in this document do not introduce security considerations
beyond those already described in <xref target="RFC7587"/>.</t>
      <t>Implementations SHOULD ensure that SDP parsing and RTP payload handling are robust against
malformed or malicious input. Implementations SHOULD validate multistream configuration parameters,
including channel mappings, stream counts, and coupled-stream counts, before use.</t>
      <t>Applications using multichannel Opus streams should also consider the privacy implications of
transmitting spatial audio information, which may reveal environmental or application context
beyond that conveyed by mono or stereo audio.</t>
      <t>Transport-level security mechanisms such as DTLS-SRTP are RECOMMENDED to protect RTP streams
from unauthorized access or modification.</t>
    </section>
    <section anchor="iana-considerations">
      <name>IANA Considerations</name>
      <t>This document does not request registration of a new RTP payload format or media subtype.</t>
      <t>The SDP parameters defined by this specification are intended for use with the existing
Opus RTP payload format defined in <xref target="RFC7587"/>.</t>
      <t>No IANA action is required by this document.</t>
    </section>
    <section anchor="acknowledgements">
      <name>Acknowledgements</name>
      <t>The author would like to thank Philipp Hancke for his detailed review and constructive
feedback during the AVTCORE adoption discussion. His comments helped shape the direction
of this document from standardizing a deployment-specific "multiopus" signaling approach
toward specifying interoperable SDP signaling for Opus multistream operation while
continuing to use the Opus RTP payload format defined in <xref target="RFC7587"/>.</t>
      <t>The author would also like to thank Hyunsik Yang for his careful review of the document and
for identifying editorial issues and textual improvements during the adoption process.</t>
      <t>The author further thanks Erik Språng, Timothy Terriberry, Mo Zanaty, the AVTCORE chairs,
and other working group participants for their review, comments, and support.</t>
    </section>
  </middle>
  <back>
    <references anchor="sec-informative-references">
      <name>Informative References</name>
      <reference anchor="libwebrtc" target="https://webrtc-review.googlesource.com/c/src/+/129768">
        <front>
          <title>Opus multistream mapping</title>
          <author>
            <organization>WebRTC</organization>
          </author>
          <date>n.d.</date>
        </front>
      </reference>
      <reference anchor="RFC7587">
        <front>
          <title>RTP Payload Format for the Opus Speech and Audio Codec</title>
          <author fullname="J. Spittka" initials="J." surname="Spittka"/>
          <author fullname="K. Vos" initials="K." surname="Vos"/>
          <author fullname="JM. Valin" initials="JM." surname="Valin"/>
          <date month="June" year="2015"/>
          <abstract>
            <t>This document defines the Real-time Transport Protocol (RTP) payload format for packetization of Opus-encoded speech and audio data necessary to integrate the codec in the most compatible way. It also provides an applicability statement for the use of Opus over RTP. Further, it describes media type registrations for the RTP payload format.</t>
          </abstract>
        </front>
        <seriesInfo name="RFC" value="7587"/>
        <seriesInfo name="DOI" value="10.17487/RFC7587"/>
      </reference>
      <reference anchor="RFC7845">
        <front>
          <title>Ogg Encapsulation for the Opus Audio Codec</title>
          <author fullname="T. Terriberry" initials="T." surname="Terriberry"/>
          <author fullname="R. Lee" initials="R." surname="Lee"/>
          <author fullname="R. Giles" initials="R." surname="Giles"/>
          <date month="April" year="2016"/>
          <abstract>
            <t>This document defines the Ogg encapsulation for the Opus interactive speech and audio codec. This allows data encoded in the Opus format to be stored in an Ogg logical bitstream.</t>
          </abstract>
        </front>
        <seriesInfo name="RFC" value="7845"/>
        <seriesInfo name="DOI" value="10.17487/RFC7845"/>
      </reference>
      <reference anchor="RFC6716">
        <front>
          <title>Definition of the Opus Audio Codec</title>
          <author fullname="JM. Valin" initials="JM." surname="Valin"/>
          <author fullname="K. Vos" initials="K." surname="Vos"/>
          <author fullname="T. Terriberry" initials="T." surname="Terriberry"/>
          <date month="September" year="2012"/>
          <abstract>
            <t>This document defines the Opus interactive speech and audio codec. Opus is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even live, distributed music performances. It scales from low bitrate narrowband speech at 6 kbit/s to very high quality stereo music at 510 kbit/s. Opus uses both Linear Prediction (LP) and the Modified Discrete Cosine Transform (MDCT) to achieve good compression of both speech and music. [STANDARDS-TRACK]</t>
          </abstract>
        </front>
        <seriesInfo name="RFC" value="6716"/>
        <seriesInfo name="DOI" value="10.17487/RFC6716"/>
      </reference>
      <reference anchor="RFC3264">
        <front>
          <title>An Offer/Answer Model with Session Description Protocol (SDP)</title>
          <author fullname="J. Rosenberg" initials="J." surname="Rosenberg"/>
          <author fullname="H. Schulzrinne" initials="H." surname="Schulzrinne"/>
          <date month="July" year="2002"/>
          <abstract>
            <t>This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK]</t>
          </abstract>
        </front>
        <seriesInfo name="RFC" value="3264"/>
        <seriesInfo name="DOI" value="10.17487/RFC3264"/>
      </reference>
    </references>
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