ref: 65471dd567beb6b1156dc292858c0a28ca55ca3e
dir: /doc/draft-ietf-payload-rtp-opus.xml/
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<rfc category="std" ipr="trust200902" docName="draft-ietf-payload-rtp-opus-01">
<?xml-stylesheet type='text/xsl' href='rfc2629.xslt' ?>
<?rfc strict="yes" ?>
<?rfc toc="yes" ?>
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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>[email protected]</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>[email protected]</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>[email protected]</email>
</address>
</author>
<date day='2' month='August' year='2013' />
<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 kb/s to 510 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 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 in
milliseconds of encoded speech or audio data for speech and audio mode
(Mode) and sampling rates (fs) of Opus and how the timestamp needs to
be incremented for packetization (ts incr). If the Opus encoder
outputs multiple encoded frames into a single packet the timestamps
have to be added up according to the combined frames.
</t>
<texttable anchor='opus-packetization' title="Supported Opus frame
sizes and timestamp increments">
<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>
</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
take advantage of the Opus in-band FEC. Possible values are 1 and 0. It is RECOMMENDED to provide
0 in case FEC cannot be utilized on the receiving side. If no
value is specified, useinbandfec is assumed to be 0.
This parameter is only a preference and the receiver MUST be able to process
packets that include 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 [email protected]</t>
<t>Jean-Marc Valin [email protected]</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 [email protected]<vspace blankLines='1'/></t>
<t>Koen Vos [email protected]<vspace blankLines='1'/></t>
<t>Jean-Marc Valin [email protected]<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",
"maxplaybackrate", "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 6.3 of <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 6.3 of <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-maxcapturerate" 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-maxcapturerate" 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>