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diff --git a/doc/draft-ietf-payload-rtp-opus.xml b/doc/draft-ietf-payload-rtp-opus.xml new file mode 100644 index 0000000..02440d9 --- /dev/null +++ b/doc/draft-ietf-payload-rtp-opus.xml @@ -0,0 +1,932 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE rfc SYSTEM "rfc2629.dtd" [ +<!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'> +<!ENTITY rfc3550 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml'> +<!ENTITY rfc3711 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3711.xml'> +<!ENTITY rfc3551 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3551.xml'> +<!ENTITY rfc4288 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4288.xml'> +<!ENTITY rfc4855 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4855.xml'> +<!ENTITY rfc4566 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4566.xml'> +<!ENTITY rfc3264 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3264.xml'> +<!ENTITY rfc2974 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2974.xml'> +<!ENTITY rfc2326 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2326.xml'> +<!ENTITY rfc3555 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3555.xml'> +<!ENTITY rfc5576 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5576.xml'> +<!ENTITY rfc6562 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6562.xml'> +<!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'> +<!ENTITY nbsp " "> + ]> + + <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" ?> +<?rfc tocdepth="3" ?> +<?rfc tocappendix='no' ?> +<?rfc tocindent='yes' ?> +<?rfc symrefs="yes" ?> +<?rfc sortrefs="yes" ?> +<?rfc compact="no" ?> +<?rfc subcompact="yes" ?> +<?rfc iprnotified="yes" ?> + + <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='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 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 jspittka@gmail.com<vspace blankLines='1'/></t> + <t>Koen Vos koenvos74@gmail.com<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", + "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> |