doc/draft-spittka-payload-rtp-opus.xml

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  <rfc category="info" ipr="trust200902" docName="draft-spittka-payload-rtp-opus-02">
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  <front>
    <title abbrev="RTP Payload Format for Opus Codec">
      RTP Payload Format for Opus Speech and Audio Codec
    </title>

    <author fullname="Julian Spittka" initials="J." surname="Spittka">
      <address>
        <email>jspittka@gmail.com</email>
      </address>
    </author>

    <author initials='K.' surname='Vos' fullname='Koen Vos'>
      <organization>Skype Technologies S.A.</organization>
      <address>
        <postal>
          <street>3210 Porter Drive</street>
          <code>94304</code>
          <city>Palo Alto</city>
          <region>CA</region>
          <country>USA</country>
        </postal>
        <email>koenvos74@gmail.com</email>
      </address>
    </author>

    <author initials="JM" surname="Valin" fullname="Jean-Marc Valin">
      <organization>Mozilla</organization>
      <address>
        <postal>
          <street>650 Castro Street</street>
          <city>Mountain View</city>
          <region>CA</region>
          <code>94041</code>
          <country>USA</country>
        </postal>
        <email>jmvalin@jmvalin.ca</email>
      </address>
    </author>

    <date day='22' month='November' year='2012' />

    <abstract>
      <t>
        This document defines the Real-time Transport Protocol (RTP) payload
        format for packetization of Opus encoded
        speech and audio data that is essential to integrate the codec in the
        most compatible way. Further, media type registrations
        are described for the RTP payload format.
      </t>
    </abstract>
  </front>

  <middle>
    <section title='Introduction'>
      <t>
        The Opus codec is a speech and audio codec developed within the
        IETF Internet Wideband Audio Codec working group (codec). The codec
        has a very low algorithmic delay and it
        is highly scalable in terms of audio bandwidth, bitrate, and
        complexity. Further, it provides different modes to efficiently encode speech signals
        as well as music signals, thus, making it the codec of choice for
        various applications using the Internet or similar networks.
      </t>
      <t>
        This document defines the Real-time Transport Protocol (RTP)
        <xref target="RFC3550"/> payload format for packetization
        of Opus encoded speech and audio data that is essential to
        integrate the Opus codec in the
        most compatible way. Further, media type registrations are described for
        the RTP payload format. More information on the Opus
        codec can be obtained from <xref target="RFC6716"/>.
      </t>
    </section>

    <section title='Conventions, Definitions and Acronyms used in this document'>
      <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
      "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
      document are to be interpreted as described in <xref target="RFC2119"/>.</t>
      <t>
      <list style='hanging'>
          <t hangText="CBR:"> Constant bitrate</t>
          <t hangText="CPU:"> Central Processing Unit</t>
          <t hangText="DTX:"> Discontinuous transmission</t>
          <t hangText="FEC:"> Forward error correction</t>
	      <t hangText="IP:"> Internet Protocol</t>
	      <t hangText="samples:"> Speech or audio samples (usually per channel)</t>
	      <t hangText="SDP:"> Session Description Protocol</t>
          <t hangText="VBR:"> Variable bitrate</t>
      </list>
      </t>
      <section title='Audio Bandwidth'>
	<t>
	  Throughout this document, we refer to the following definitions:
	</t>
          <texttable anchor='bandwidth_definitions'>
	    <ttcol align='center'>Abbreviation</ttcol>
            <ttcol align='center'>Name</ttcol>
            <ttcol align='center'>Bandwidth</ttcol>
            <ttcol align='center'>Sampling</ttcol>
            <c>nb</c>
            <c>Narrowband</c>
            <c>0 - 4000</c>
            <c>8000</c>

            <c>mb</c>
            <c>Mediumband</c>
            <c>0 - 6000</c>
            <c>12000</c>

            <c>wb</c>
            <c>Wideband</c>
            <c>0 - 8000</c>
            <c>16000</c>

            <c>swb</c>
            <c>Super-wideband</c>
            <c>0 - 12000</c>
            <c>24000</c>

            <c>fb</c>
            <c>Fullband</c>
            <c>0 - 20000</c>
            <c>48000</c>

            <postamble>
              Audio bandwidth naming
            </postamble>
          </texttable>
      </section>
    </section>

    <section title='Opus Codec'>
      <t>
        The Opus <xref target="RFC6716"/> speech and audio codec has been developed to encode speech
        signals as well as audio signals. Two different modes, a voice mode
        or an audio mode, may be chosen to allow the most efficient coding
        dependent on the type of input signal, the sampling frequency of the
        input signal, and the specific application.
      </t>

      <t>
        The voice mode allows efficient encoding of voice signals at lower bit
        rates while the audio mode is optimized for audio signals at medium and
        higher bitrates.
      </t>

      <t>
        The Opus speech and audio codec is highly scalable in terms of audio
        bandwidth, bitrate, and complexity. Further, Opus allows
        transmitting stereo signals.
      </t>

      <section title='Network Bandwidth'>
          <t>
	    Opus supports all bitrates from 6&nbsp;kb/s to 510&nbsp;kb/s.
	    The bitrate can be changed dynamically within that range.
	    All
	    other parameters being
	    equal, higher bitrate results in higher quality.
	  </t>
	  <section title='Recommended Bitrate' anchor='bitrate_by_bandwidth'>
	  <t>
	    For a frame size of
	    20&nbsp;ms, these
	    are the bitrate "sweet spots" for Opus in various configurations:

          <list style="symbols">
	    <t>8-12 kb/s for NB speech,</t>
	    <t>16-20 kb/s for WB speech,</t>
	    <t>28-40 kb/s for FB speech,</t>
	    <t>48-64 kb/s for FB mono music, and</t>
	    <t>64-128 kb/s for FB stereo music.</t>
	  </list>
	</t>
      </section>
        <section title='Variable versus Constant Bit Rate'  anchor='variable-vs-constant-bitrate'>
          <t>
	    For the same average bitrate, variable bitrate (VBR) can achieve higher quality
	    than constant bitrate (CBR). For the majority of voice transmission application, VBR
	    is the best choice. One potential reason for choosing CBR is the potential
	    information leak that <spanx style='emph'>may</spanx> occur when encrypting the
	    compressed stream. See <xref target="RFC6562"/> for guidelines on when VBR is
	    appropriate for encrypted audio communications. In the case where an existing
	    VBR stream needs to be converted to CBR for security reasons, then the Opus padding
	    mechanism described in <xref target="RFC6716"/> is the RECOMMENDED way to achieve padding
	    because the RTP padding bit is unencrypted.</t>

	    <t>
            The bitrate can be adjusted at any point in time. To avoid congestion,
            the average bitrate SHOULD be adjusted to the available
            network capacity. If no target bitrate is specified, the bitrates specified in
            <xref target='bitrate_by_bandwidth'/> are RECOMMENDED.
          </t>

        </section>

        <section title='Discontinuous Transmission (DTX)'>

          <t>
            The Opus codec may, as described in <xref target='variable-vs-constant-bitrate'/>,
            be operated with an adaptive bitrate. In that case, the bitrate
            will automatically be reduced for certain input signals like periods
            of silence. During continuous transmission the bitrate will be
            reduced, when the input signal allows to do so, but the transmission
            to the receiver itself will never be interrupted. Therefore, the
            received signal will maintain the same high level of quality over the
            full duration of a transmission while minimizing the average bit
            rate over time.
          </t>

          <t>
            In cases where the bitrate of Opus needs to be reduced even
            further or in cases where only constant bitrate is available,
            the Opus encoder may be set to use discontinuous
            transmission (DTX), where parts of the encoded signal that
            correspond to periods of silence in the input speech or audio signal
            are not transmitted to the receiver.
          </t>

          <t>
            On the receiving side, the non-transmitted parts will be handled by a
            frame loss concealment unit in the Opus decoder which generates a
            comfort noise signal to replace the non transmitted parts of the
            speech or audio signal.
          </t>

          <t>
            The DTX mode of Opus will have a slightly lower speech or audio
            quality than the continuous mode. Therefore, it is RECOMMENDED to
            use Opus in the continuous mode unless restraints on network
            capacity are severe. The DTX mode can be engaged for operation
            in both adaptive or constant bitrate.
          </t>

        </section>

        </section>

      <section title='Complexity'>

        <t>
          Complexity can be scaled to optimize for CPU resources in real-time, mostly as
          a trade-off between audio quality and bitrate. Also, different modes of Opus have different complexity.
        </t>

      </section>

      <section title="Forward Error Correction (FEC)">

        <t>
          The voice mode of Opus allows for "in-band" forward error correction (FEC)
          data to be embedded into the bit stream of Opus. This FEC scheme adds
          redundant information about the previous packet (n-1) to the current
          output packet n. For
          each frame, the encoder decides whether to use FEC based on (1) an
          externally-provided estimate of the channel's packet loss rate; (2) an
          externally-provided estimate of the channel's capacity; (3) the
          sensitivity of the audio or speech signal to packet loss; (4) whether
          the receiving decoder has indicated it can take advantage of "in-band"
          FEC information. The decision to send "in-band" FEC information is
          entirely controlled by the encoder and therefore no special precautions
          for the payload have to be taken.
        </t>

        <t>
          On the receiving side, the decoder can take advantage of this
          additional information when, in case of a packet loss, the next packet
          is available.  In order to use the FEC data, the jitter buffer needs
          to provide access to payloads with the FEC data.  The decoder API function
          has a flag to indicate that a FEC frame rather than a regular frame should
          be decoded.  If no FEC data is available for the current frame, the decoder
          will consider the frame lost and invokes the frame loss concealment.
        </t>

        <t>
          If the FEC scheme is not implemented on the receiving side, FEC
          SHOULD NOT be used, as it leads to an inefficient usage of network
          resources. Decoder support for FEC SHOULD be indicated at the time a
          session is set up.
        </t>

      </section>

      <section title='Stereo Operation'>

        <t>
          Opus allows for transmission of stereo audio signals. This operation
          is signaled in-band in the Opus payload and no special arrangement
          is required in the payload format. Any implementation of the Opus
          decoder MUST be capable of receiving stereo signals, although it MAY
	  decode those signals as mono.
        </t>
        <t>
          If a decoder can not take advantage of the benefits of a stereo signal
          this SHOULD be indicated at the time a session is set up. In that case
          the sending side SHOULD NOT send stereo signals as it leads to an
          inefficient usage of the network.
        </t>

      </section>

    </section>

    <section title='Opus RTP Payload Format' anchor='opus-rtp-payload-format'>
      <t>The payload format for Opus consists of the RTP header and Opus payload
      data.</t>
      <section title='RTP Header Usage'>
        <t>The format of the RTP header is specified in <xref target="RFC3550"/>. The Opus
        payload format uses the fields of the RTP header consistent with this
        specification.</t>

        <t>The payload length of Opus is a multiple number of octets and
        therefore no padding is required. The payload MAY be padded by an
        integer number of octets according to <xref target="RFC3550"/>.</t>

        <t>The marker bit (M) of the RTP header is used in accordance with
	Section 4.1 of <xref target="RFC3551"/>.</t>

        <t>The RTP payload type for Opus has not been assigned statically and is
        expected to be assigned dynamically.</t>

        <t>The receiving side MUST be prepared to receive duplicates of RTP
        packets. Only one of those payloads MUST be provided to the Opus decoder
        for decoding and others MUST be discarded.</t>

        <t>Opus supports 5 different audio bandwidths which may be adjusted during
        the duration of a call. The RTP timestamp clock frequency is defined as
        the highest supported sampling frequency of Opus, i.e. 48000 Hz, for all
        modes and sampling rates of Opus. The unit
        for the timestamp is samples per single (mono) channel. The RTP timestamp corresponds to the
        sample time of the first encoded sample in the encoded frame. For sampling
        rates lower than 48000 Hz the number of samples has to be multiplied with
        a multiplier according to <xref target="fs-upsample-factors"/> to determine
        the RTP timestamp.</t>

        <texttable anchor='fs-upsample-factors' title="Timestamp multiplier">
          <ttcol align='center'>fs (Hz)</ttcol>
          <ttcol align='center'>Multiplier</ttcol>
          <c>8000</c>
          <c>6</c>
          <c>12000</c>
          <c>4</c>
          <c>16000</c>
          <c>3</c>
          <c>24000</c>
          <c>2</c>
          <c>48000</c>
          <c>1</c>
        </texttable>
      </section>

      <section title='Payload Structure'>
        <t>
          The Opus encoder can be set to output encoded frames representing 2.5, 5, 10, 20,
          40, or 60 ms of speech or audio data. Further, an arbitrary number of frames can be
          combined into a packet. The maximum packet length is limited to the amount of encoded
          data representing 120 ms of speech or audio data. The packetization of encoded data
          is purely done by the Opus encoder and therefore only one packet output from the Opus
          encoder MUST be used as a payload.
        </t>

        <t><xref target='payload-structure'/> shows the structure combined with the RTP header.</t>

        <figure anchor="payload-structure"
                title="Payload Structure with RTP header">
          <artwork>
            <![CDATA[
+----------+--------------+
|RTP Header| Opus Payload |
+----------+--------------+
           ]]>
          </artwork>
        </figure>

        <t>
          <xref target='opus-packetization'/> shows supported frame sizes for different modes
          and sampling rates of Opus and how the timestamp needs to be incremented for
          packetization.
        </t>

        <texttable anchor='opus-packetization'>
            <ttcol align='center'>Mode</ttcol>
            <ttcol align='center'>fs</ttcol>
            <ttcol align='center'>2.5</ttcol>
            <ttcol align='center'>5</ttcol>
            <ttcol align='center'>10</ttcol>
            <ttcol align='center'>20</ttcol>
            <ttcol align='center'>40</ttcol>
            <ttcol align='center'>60</ttcol>
            <c>ts incr</c>
            <c>all</c>
            <c>120</c>
            <c>240</c>
            <c>480</c>
            <c>960</c>
            <c>1920</c>
            <c>2880</c>
            <c>voice</c>
            <c>nb/mb/wb/swb/fb</c>
            <c></c>
            <c></c>
            <c>x</c>
            <c>x</c>
            <c>x</c>
            <c>x</c>
            <c>audio</c>
            <c>nb/wb/swb/fb</c>
            <c>x</c>
            <c>x</c>
            <c>x</c>
            <c>x</c>
            <c></c>
            <c></c>
            <postamble>
              Mode specifies the Opus mode of operation; fs specifies the audio sampling
              frequency in Hertz (Hz); 2.5, 5, 10, 20, 40, and 60 represent the duration of
              encoded speech or audio data in a packet; ts incr specifies the
              value the timestamp needs to be incremented for the representing packet size.
              For multiple frames in a packet these values have to be multiplied with the
              respective number of frames.
            </postamble>
          </texttable>

      </section>

    </section>

    <section title='Congestion Control'>

      <t>The adaptive nature of the Opus codec allows for an efficient
      congestion control.</t>

      <t>The target bitrate of Opus can be adjusted at any point in time and
      thus allowing for an efficient congestion control. Furthermore, the amount
      of encoded speech or audio data encoded in a
      single packet can be used for congestion control since the transmission
      rate is inversely proportional to these frame sizes. A lower packet
      transmission rate reduces the amount of header overhead but at the same
      time increases latency and error sensitivity and should be done with care.</t>

      <t>It is RECOMMENDED that congestion control is applied during the
      transmission of Opus encoded data.</t>
    </section>

    <section title='IANA Considerations'>
      <t>One media subtype (audio/opus) has been defined and registered as
      described in the following section.</t>

      <section title='Opus Media Type Registration'>
        <t>Media type registration is done according to <xref
        target="RFC4288"/> and <xref target="RFC4855"/>.<vspace
        blankLines='1'/></t>

          <t>Type name: audio<vspace blankLines='1'/></t>
          <t>Subtype name: opus<vspace blankLines='1'/></t>

          <t>Required parameters:</t>
          <t><list style="hanging">
            <t hangText="rate:"> RTP timestamp clock rate is incremented with
            48000 Hz clock rate for all modes of Opus and all sampling
            frequencies. For audio sampling rates other than 48000 Hz the rate
            has to be adjusted to 48000 Hz according to <xref target="fs-upsample-factors"/>.
          </t>
          </list></t>

          <t>Optional parameters:</t>

          <t><list style="hanging">
            <t hangText="maxplaybackrate:">
              a hint about the maximum output sampling rate that the receiver is
              capable of rendering in Hz.
              The decoder MUST be capable of decoding
              any audio bandwidth but due to hardware limitations only signals
              up to the specified sampling rate can be played back. Sending signals
              with higher audio bandwidth results in higher than necessary network
              usage and encoding complexity, so an encoder SHOULD NOT encode
              frequencies above the audio bandwidth specified by maxplaybackrate.
              This parameter can take any value between 8000 and 48000, although
              commonly the value will match one of the Opus bandwidths 
              (<xref target="bandwidth_definitions"/>).
              By default, the receiver is assumed to have no limitations, i.e. 48000.
              <vspace blankLines='1'/>
            </t>

            <t hangText="sprop-maxcapturerate:">
              a hint about the maximum input sampling rate that the sender is likely to produce.
              This is not a guarantee that the sender will never send any higher bandwidth
              (e.g. it could send a pre-recorded prompt that uses a higher bandwidth), but it
              indicates to the receiver that frequencies above this maximum can safely be discarded.
              This parameter is useful to avoid wasting receiver resources by operating the audio
              processing pipeline (e.g. echo cancellation) at a higher rate than necessary.
              This parameter can take any value between 8000 and 48000, although
              commonly the value will match one of the Opus bandwidths 
              (<xref target="bandwidth_definitions"/>).
              By default, the sender is assumed to have no limitations, i.e. 48000.
              <vspace blankLines='1'/>
            </t>

            <t hangText="maxptime:"> the decoder's maximum length of time in
            milliseconds rounded up to the next full integer value represented
            by the media in a packet that can be
            encapsulated in a received packet according to Section 6 of
            <xref target="RFC4566"/>. Possible values are 3, 5, 10, 20, 40,
            and 60 or an arbitrary multiple of Opus frame sizes rounded up to
            the next full integer value up to a maximum value of 120 as
            defined in <xref target='opus-rtp-payload-format'/>. If no value is
              specified, 120 is assumed as default. This value is a recommendation
              by the decoding side to ensure the best
              performance for the decoder. The decoder MUST be
              capable of accepting any allowed packet sizes to
              ensure maximum compatibility.
              <vspace blankLines='1'/></t>

            <t hangText="ptime:"> the decoder's recommended length of time in
            milliseconds rounded up to the next full integer value represented
            by the media in a packet according to
            Section 6 of <xref target="RFC4566"/>. Possible values are
            3, 5, 10, 20, 40, or 60 or an arbitrary multiple of Opus frame sizes
            rounded up to the next full integer value up to a maximum
            value of 120 as defined in <xref
            target='opus-rtp-payload-format'/>. If no value is
              specified, 20 is assumed as default. If ptime is greater than
              maxptime, ptime MUST be ignored. This parameter MAY be changed
              during a session. This value is a recommendation by the decoding
              side to ensure the best
              performance for the decoder. The decoder MUST be
              capable of accepting any allowed packet sizes to
              ensure maximum compatibility.
              <vspace blankLines='1'/></t>

            <t hangText="minptime:"> the decoder's minimum length of time in
            milliseconds rounded up to the next full integer value represented
            by the media in a packet that SHOULD
            be encapsulated in a received packet according to Section 6 of <xref
            target="RFC4566"/>. Possible values are 3, 5, 10, 20, 40, and 60
            or an arbitrary multiple of Opus frame sizes rounded up to the next
            full integer value up to a maximum value of 120
            as defined in <xref target='opus-rtp-payload-format'/>. If no value is
              specified, 3 is assumed as default. This value is a recommendation
              by the decoding side to ensure the best
              performance for the decoder. The decoder MUST be
              capable to accept any allowed packet sizes to
              ensure maximum compatibility.
              <vspace blankLines='1'/></t>

            <t hangText="maxaveragebitrate:"> specifies the maximum average
	    receive bitrate of a session in bits per second (b/s). The actual
            value of the bitrate may vary as it is dependent on the
            characteristics of the media in a packet. Note that the maximum
            average bitrate MAY be modified dynamically during a session. Any
            positive integer is allowed but values outside the range between
            6000 and 510000 SHOULD be ignored. If no value is specified, the
            maximum value specified in <xref target='bitrate_by_bandwidth'/>
            for the corresponding mode of Opus and corresponding maxplaybackrate:
            will be the default.<vspace blankLines='1'/></t>

            <t hangText="stereo:">
              specifies whether the decoder prefers receiving stereo or mono signals.
              Possible values are 1 and 0 where 1 specifies that stereo signals are preferred
              and 0 specifies that only mono signals are preferred.
              Independent of the stereo parameter every receiver MUST be able to receive and
              decode stereo signals but sending stereo signals to a receiver that signaled a
              preference for mono signals may result in higher than necessary network
              utilisation and encoding complexity. If no value is specified, mono
              is assumed (stereo=0).<vspace blankLines='1'/>
            </t>

            <t hangText="sprop-stereo:">
              specifies whether the sender is likely to produce stereo audio.
              Possible values are 1 and 0 where 1 specifies that stereo signals are likely to
	      be sent, and 0 speficies that the sender will likely only send mono.
	      This is not a guarantee that the sender will never send stereo audio
	      (e.g. it could send a pre-recorded prompt that uses stereo), but it
	      indicates to the receiver that the received signal can be safely downmixed to mono.
	      This parameter is useful to avoid wasting receiver resources by operating the audio
	      processing pipeline (e.g. echo cancellation) in stereo when not necessary.
              If no value is specified, mono
              is assumed (sprop-stereo=0).<vspace blankLines='1'/>
            </t>

            <t hangText="cbr:">
              specifies if the decoder prefers the use of a constant bitrate versus
              variable bitrate. Possible values are 1 and 0 where 1 specifies constant
              bitrate and 0 specifies variable bitrate. If no value is specified, cbr
              is assumed to be 0. Note that the maximum average bitrate may still be
              changed, e.g. to adapt to changing network conditions.<vspace blankLines='1'/>
            </t>

            <t hangText="useinbandfec:"> specifies that the decoder has the capability to
            use the Opus in-band FEC. Possible values are 1 and 0. It is RECOMMENDED to provide
            0 in case FEC cannot be used on the receiving side. If no
            value is specified, useinbandfec is assumed to be 1.
            This parameter is only a preference and the receiver MUST be able to process
            packets that have FEC information, even if it means the FEC part is discarded.
            <vspace blankLines='1'/></t>

            <t hangText="usedtx:"> specifies if the decoder prefers the use of
            DTX. Possible values are 1 and 0. If no value is specified, usedtx
            is assumed to be 0.<vspace blankLines='1'/></t>
          </list></t>

          <t>Encoding considerations:<vspace blankLines='1'/></t>
          <t><list style="hanging">
            <t>Opus media type is framed and consists of binary data according
            to Section 4.8 in <xref target="RFC4288"/>.</t>
          </list></t>

          <t>Security considerations: </t>
          <t><list style="hanging">
            <t>See <xref target='security-considerations'/> of this document.</t>
          </list></t>

          <t>Interoperability considerations: none<vspace blankLines='1'/></t>
          <t>Published specification: none<vspace blankLines='1'/></t>

          <t>Applications that use this media type: </t>
          <t><list style="hanging">
            <t>Any application that requires the transport of
            speech or audio data may use this media type. Some examples are,
            but not limited to, audio and video conferencing, Voice over IP,
            media streaming.</t>
          </list></t>

          <t>Person & email address to contact for further information:</t>
          <t><list style="hanging">
            <t>SILK Support silksupport@skype.net</t>
            <t>Jean-Marc Valin jmvalin@jmvalin.ca</t>
          </list></t>

          <t>Intended usage: COMMON<vspace blankLines='1'/></t>

          <t>Restrictions on usage:<vspace blankLines='1'/></t>

          <t><list style="hanging">
            <t>For transfer over RTP, the RTP payload format (<xref
            target='opus-rtp-payload-format'/> of this document) SHALL be
            used.</t>
          </list></t>

          <t>Author:</t>
          <t><list style="hanging">
            <t>Julian Spittka julian.spittka@skype.net<vspace blankLines='1'/></t>
            <t>Koen Vos koen.vos@skype.net<vspace blankLines='1'/></t>
            <t>Jean-Marc Valin jmvalin@jmvalin.ca<vspace blankLines='1'/></t>
          </list></t>

          <t> Change controller: TBD</t>
      </section>

      <section title='Mapping to SDP Parameters'>
        <t>The information described in the media type specification has a
        specific mapping to fields in the Session Description Protocol (SDP)
        <xref target="RFC4566"/>, which is commonly used to describe RTP
        sessions. When SDP is used to specify sessions employing the Opus codec,
        the mapping is as follows:</t>

        <t>
          <list style="symbols">
            <t>The media type ("audio") goes in SDP "m=" as the media name.</t>

            <t>The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
            name. The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of
	    channels MUST be 2.</t>

            <t>The OPTIONAL media type parameters "ptime" and "maxptime" are
            mapped to "a=ptime" and "a=maxptime" attributes, respectively, in the
            SDP.</t>

            <t>The OPTIONAL media type parameters "maxaveragebitrate",
            "minptime", "stereo", "cbr", "useinbandfec", and "usedtx", when
            present, MUST be included in the "a=fmtp" attribute in the SDP,
            expressed as a media type string in the form of a
            semicolon-separated list of parameter=value pairs (e.g.,
            maxaveragebitrate=20000). They MUST NOT be specified in an
            SSRC-specific "fmtp" source-level attribute (as defined in
            Section&nbsp;6.3 of&nbsp;<xref target="RFC5576"/>).</t>

            <t>The OPTIONAL media type parameters "sprop-maxcapturerate",
            and "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by
            copying them directly from the media type parameter string as part
            of the semicolon-separated list of parameter=value pairs (e.g.,
            sprop-stereo=1). These same OPTIONAL media type parameters MAY also
            be specified using an SSRC-specific "fmtp" source-level attribute
            as described in Section&nbsp;6.3 of&nbsp;<xref target="RFC5576"/>.
            They MAY be specified in both places, in which case the parameter
            in the source-level attribute overrides the one found on the
            "a=fmtp" line. The value of any parameter which is not specified in
            a source-level source attribute MUST be taken from the "a=fmtp"
            line, if it is present there.</t>

          </list>
        </t>

        <t>Below are some examples of SDP session descriptions for Opus:</t>

        <t>Example 1: Standard mono session with 48000 Hz clock rate</t>
          <figure>
            <artwork>
              <![CDATA[
    m=audio 54312 RTP/AVP 101
    a=rtpmap:101 opus/48000/2
              ]]>
            </artwork>
          </figure>


        <t>Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
        recommended packet size of 40 ms, maximum average bitrate of 20000 bps,
        prefers to receive stereo but only plans to send mono, FEC is allowed,
        DTX is not allowed</t>

        <figure>
          <artwork>
            <![CDATA[
    m=audio 54312 RTP/AVP 101
    a=rtpmap:101 opus/48000/2
    a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000;
    maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0
    a=ptime:40
    a=maxptime:40
            ]]>
          </artwork>
        </figure>

        <t>Example 3: Two-way full-band stereo preferred</t>

        <figure>
          <artwork>
            <![CDATA[
    m=audio 54312 RTP/AVP 101
    a=rtpmap:101 opus/48000/2
    a=fmtp:101 stereo=1; sprop-stereo=1
            ]]>
          </artwork>
        </figure>


      <section title='Offer-Answer Model Considerations for Opus'>

          <t>When using the offer-answer procedure described in <xref
          target="RFC3264"/> to negotiate the use of Opus, the following
          considerations apply:</t>

          <t><list style="symbols">

            <t>Opus supports several clock rates. For signaling purposes only
            the highest, i.e. 48000, is used. The actual clock rate of the
            corresponding media is signaled inside the payload and is not
            subject to this payload format description. The decoder MUST be
            capable to decode every received clock rate. An example
            is shown below:

            <figure>
              <artwork>
                <![CDATA[
    m=audio 54312 RTP/AVP 100
    a=rtpmap:100 opus/48000/2
                ]]>
              </artwork>
            </figure>
            </t>

            <t>The "ptime" and "maxptime" parameters are unidirectional
            receive-only parameters and typically will not compromise
            interoperability; however, dependent on the set values of the
            parameters the performance of the application may suffer.  <xref
            target="RFC3264"/> defines the SDP offer-answer handling of the
            "ptime" parameter. The "maxptime" parameter MUST be handled in the
            same way.</t>

            <t>
              The "minptime" parameter is a unidirectional
              receive-only parameters and typically will not compromise
              interoperability; however, dependent on the set values of the
              parameter the performance of the application may suffer and should be
              set with care.
            </t>

            <t>
              The "maxplaybackrate" parameter is a unidirectional receive-only
              parameter that reflects limitations of the local receiver. The sender
              of the other side SHOULD NOT send with an audio bandwidth higher than
              "maxplaybackrate" as this would lead to inefficient use of network resources.
              The "maxplaybackrate" parameter does not
	      affect interoperability. Also, this parameter SHOULD NOT be used
	      to adjust the audio bandwidth as a function of the bitrates, as this
	      is the responsibility of the Opus encoder implementation.
            </t>

            <t>The "maxaveragebitrate" parameter is a unidirectional receive-only
            parameter that reflects limitations of the local receiver. The sender
            of the other side MUST NOT send with an average bitrate higher than
            "maxaveragebitrate" as it might overload the network and/or
            receiver. The "maxaveragebitrate" parameter typically will not
            compromise interoperability; however, dependent on the set value of
            the parameter the performance of the application may suffer and should
            be set with care.</t>

            <t>The "sprop-maxcaptureerate" and "sprop-stereo" parameters are
            unidirectional sender-only parameters that reflect limitations of
            the sender side.
            They allow the receiver to set up a reduced-complexity audio
            processing pipeline if the  sender is not planning to use the full
            range of Opus's capabilities.
            Neither "sprop-maxcaptureerate" nor "sprop-stereo" affect
            interoperability and the receiver MUST be capable of receiving any signal.
            </t>

            <t>
              The "stereo" parameter is a unidirectional receive-only
              parameter.
            </t>

            <t>
              The "cbr" parameter is a unidirectional receive-only
              parameter.
            </t>

            <t>The "useinbandfec" parameter is a unidirectional receive-only
            parameter.</t>

            <t>The "usedtx" parameter is a unidirectional receive-only
            parameter.</t>

            <t>Any unknown parameter in an offer MUST be ignored by the receiver
            and MUST be removed from the answer.</t>

          </list></t>
      </section>

      <section title='Declarative SDP Considerations for Opus'>

        <t>For declarative use of SDP such as in Session Announcement Protocol
        (SAP), <xref target="RFC2974"/>, and RTSP, <xref target="RFC2326"/>, for
        Opus, the following needs to be considered:</t>

        <t><list style="symbols">

          <t>The values for "maxptime", "ptime", "minptime", "maxplaybackrate", and
          "maxaveragebitrate" should be selected carefully to ensure that a
          reasonable performance can be achieved for the participants of a session.</t>

          <t>
            The values for "maxptime", "ptime", and "minptime" of the payload
            format configuration are recommendations by the decoding side to ensure
            the best performance for the decoder. The decoder MUST be
            capable to accept any allowed packet sizes to
            ensure maximum compatibility.
          </t>

          <t>All other parameters of the payload format configuration are declarative
          and a participant MUST use the configurations that are provided for
          the session. More than one configuration may be provided if necessary
          by declaring multiple RTP payload types; however, the number of types
          should be kept small.</t>
        </list></t>
      </section>
    </section>
  </section>

    <section title='Security Considerations' anchor='security-considerations'>

      <t>All RTP packets using the payload format defined in this specification
      are subject to the general security considerations discussed in the RTP
      specification <xref target="RFC3550"/> and any profile from
      e.g. <xref target="RFC3711"/> or <xref target="RFC3551"/>.</t>

      <t>This payload format transports Opus encoded speech or audio data,
      hence, security issues include confidentiality, integrity protection, and
      authentication of the speech or audio itself. The Opus payload format does
      not have any built-in security mechanisms. Any suitable external
      mechanisms, such as SRTP <xref target="RFC3711"/>, MAY be used.</t>

      <t>This payload format and the Opus encoding do not exhibit any
      significant non-uniformity in the receiver-end computational load and thus
      are unlikely to pose a denial-of-service threat due to the receipt of
      pathological datagrams.</t>
    </section>

    <section title='Acknowledgements'>
    <t>TBD</t>
    </section>
  </middle>

  <back>
    <references title="Normative References">
      &rfc2119;
      &rfc3550;
      &rfc3711;
      &rfc3551;
      &rfc4288;
      &rfc4855;
      &rfc4566;
      &rfc3264;
      &rfc2974;
      &rfc2326;
      &rfc5576;
      &rfc6562;
      &rfc6716;
    </references>

  </back>
</rfc>