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| <!DOCTYPE rfc SYSTEM "rfc2629.dtd" [ |
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| |
| ]> |
| |
| <rfc category="info" ipr="trust200902" docName="draft-spittka-payload-rtp-opus-03"> |
| <?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='30' 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 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> |