961 lines
42 KiB
XML
961 lines
42 KiB
XML
<?xml version="1.0" encoding="UTF-8"?>
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<!DOCTYPE rfc SYSTEM "rfc2629.dtd" [
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<!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'>
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<!ENTITY rfc3389 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3389.xml'>
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<!ENTITY rfc3550 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml'>
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<!ENTITY rfc3711 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3711.xml'>
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<!ENTITY rfc3551 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3551.xml'>
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<!ENTITY rfc6838 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6838.xml'>
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<!ENTITY rfc4855 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4855.xml'>
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<!ENTITY rfc4566 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4566.xml'>
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<!ENTITY rfc4585 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4585.xml'>
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<!ENTITY rfc3264 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3264.xml'>
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<!ENTITY rfc2974 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2974.xml'>
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<!ENTITY rfc2326 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2326.xml'>
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<!ENTITY rfc3555 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3555.xml'>
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<!ENTITY rfc5124 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5124.xml'>
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<!ENTITY rfc5405 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5405.xml'>
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<!ENTITY rfc5576 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5576.xml'>
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<!ENTITY rfc6562 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6562.xml'>
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<!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'>
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<!ENTITY rfc7202 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.7202.xml'>
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<!ENTITY nbsp " ">
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]>
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<rfc category="std" ipr="trust200902" docName="draft-ietf-payload-rtp-opus-11">
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<?xml-stylesheet type='text/xsl' href='rfc2629.xslt' ?>
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<?rfc strict="yes" ?>
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<?rfc toc="yes" ?>
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<?rfc tocdepth="3" ?>
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<?rfc tocappendix='no' ?>
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<?rfc tocindent='yes' ?>
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<?rfc symrefs="yes" ?>
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<?rfc sortrefs="yes" ?>
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<?rfc compact="no" ?>
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<?rfc subcompact="yes" ?>
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<?rfc iprnotified="yes" ?>
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<front>
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<title abbrev="RTP Payload Format for Opus">
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RTP Payload Format for the Opus Speech and Audio Codec
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</title>
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<author fullname="Julian Spittka" initials="J." surname="Spittka">
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<address>
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<email>jspittka@gmail.com</email>
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</address>
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</author>
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<author initials='K.' surname='Vos' fullname='Koen Vos'>
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<organization>vocTone</organization>
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<address>
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<postal>
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<street></street>
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<code></code>
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<city></city>
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<region></region>
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<country></country>
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</postal>
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<email>koenvos74@gmail.com</email>
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</address>
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</author>
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<author initials="JM" surname="Valin" fullname="Jean-Marc Valin">
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<organization>Mozilla</organization>
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<address>
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<postal>
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<street>331 E. Evelyn Avenue</street>
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<city>Mountain View</city>
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<region>CA</region>
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<code>94041</code>
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<country>USA</country>
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</postal>
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<email>jmvalin@jmvalin.ca</email>
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</address>
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</author>
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<date day='14' month='April' year='2015' />
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<abstract>
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<t>
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This document defines the Real-time Transport Protocol (RTP) payload
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format for packetization of Opus encoded
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speech and audio data necessary to integrate the codec in the
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most compatible way. It also provides an applicability statement
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for the use of Opus over RTP. Further, it describes media type registrations
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for the RTP payload format.
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</t>
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</abstract>
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</front>
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<middle>
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<section title='Introduction'>
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<t>
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Opus <xref target="RFC6716"/> is a speech and audio codec developed within the
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IETF Internet Wideband Audio Codec working group. The codec
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has a very low algorithmic delay and it
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is highly scalable in terms of audio bandwidth, bitrate, and
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complexity. Further, it provides different modes to efficiently encode speech signals
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as well as music signals, thus making it the codec of choice for
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various applications using the Internet or similar networks.
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</t>
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<t>
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This document defines the Real-time Transport Protocol (RTP)
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<xref target="RFC3550"/> payload format for packetization
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of Opus encoded speech and audio data necessary to
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integrate Opus in the
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most compatible way. It also provides an applicability statement
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for the use of Opus over RTP.
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Further, it describes media type registrations for
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the RTP payload format.
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</t>
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</section>
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|
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<section title='Conventions, Definitions and Acronyms used in this document'>
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<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
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"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
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document are to be interpreted as described in <xref target="RFC2119"/>.</t>
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<t>
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<list style='hanging'>
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<t hangText="audio bandwidth:"> The range of audio frequecies being coded</t>
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<t hangText="CBR:"> Constant bitrate</t>
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<t hangText="CPU:"> Central Processing Unit</t>
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<t hangText="DTX:"> Discontinuous transmission</t>
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<t hangText="FEC:"> Forward error correction</t>
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<t hangText="IP:"> Internet Protocol</t>
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<t hangText="samples:"> Speech or audio samples (per channel)</t>
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<t hangText="SDP:"> Session Description Protocol</t>
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<t hangText="VBR:"> Variable bitrate</t>
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</list>
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</t>
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<t>
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Throughout this document, we refer to the following definitions:
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</t>
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<texttable anchor='bandwidth_definitions'>
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<ttcol align='center'>Abbreviation</ttcol>
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<ttcol align='center'>Name</ttcol>
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<ttcol align='center'>Audio Bandwidth (Hz)</ttcol>
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<ttcol align='center'>Sampling Rate (Hz)</ttcol>
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<c>NB</c>
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<c>Narrowband</c>
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<c>0 - 4000</c>
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<c>8000</c>
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<c>MB</c>
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<c>Mediumband</c>
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<c>0 - 6000</c>
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<c>12000</c>
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<c>WB</c>
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<c>Wideband</c>
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<c>0 - 8000</c>
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<c>16000</c>
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<c>SWB</c>
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<c>Super-wideband</c>
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<c>0 - 12000</c>
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<c>24000</c>
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<c>FB</c>
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<c>Fullband</c>
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<c>0 - 20000</c>
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<c>48000</c>
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<postamble>
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Audio bandwidth naming
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</postamble>
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</texttable>
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</section>
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<section title='Opus Codec'>
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<t>
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Opus encodes speech
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signals as well as general audio signals. Two different modes can be
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chosen, a voice mode or an audio mode, to allow the most efficient coding
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depending on the type of the input signal, the sampling frequency of the
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input signal, and the intended application.
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</t>
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<t>
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The voice mode allows efficient encoding of voice signals at lower bit
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rates while the audio mode is optimized for general audio signals at medium and
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higher bitrates.
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</t>
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<t>
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Opus is highly scalable in terms of audio
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bandwidth, bitrate, and complexity. Further, Opus allows
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transmitting stereo signals with in-band signaling in the bit-stream.
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</t>
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<section title='Network Bandwidth'>
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<t>
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Opus supports bitrates from 6 kb/s to 510 kb/s.
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The bitrate can be changed dynamically within that range.
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All
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other parameters being
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equal, higher bitrates result in higher audio quality.
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</t>
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<section title='Recommended Bitrate' anchor='bitrate_by_bandwidth'>
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<t>
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For a frame size of
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20 ms, these
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are the bitrate "sweet spots" for Opus in various configurations:
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<list style="symbols">
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<t>8-12 kb/s for NB speech,</t>
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<t>16-20 kb/s for WB speech,</t>
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<t>28-40 kb/s for FB speech,</t>
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<t>48-64 kb/s for FB mono music, and</t>
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<t>64-128 kb/s for FB stereo music.</t>
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</list>
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</t>
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</section>
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<section title='Variable versus Constant Bitrate' anchor='variable-vs-constant-bitrate'>
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<t>
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For the same average bitrate, variable bitrate (VBR) can achieve higher audio quality
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|
than constant bitrate (CBR). For the majority of voice transmission applications, VBR
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is the best choice. One reason for choosing CBR is the potential
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information leak that <spanx style='emph'>might</spanx> occur when encrypting the
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compressed stream. See <xref target="RFC6562"/> for guidelines on when VBR is
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appropriate for encrypted audio communications. In the case where an existing
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VBR stream needs to be converted to CBR for security reasons, then the Opus padding
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mechanism described in <xref target="RFC6716"/> is the RECOMMENDED way to achieve padding
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because the RTP padding bit is unencrypted.</t>
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<t>
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The bitrate can be adjusted at any point in time. To avoid congestion,
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the average bitrate SHOULD NOT exceed the available
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network bandwidth. If no target bitrate is specified, the bitrates specified in
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<xref target='bitrate_by_bandwidth'/> are RECOMMENDED.
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</t>
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|
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</section>
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|
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|
<section title='Discontinuous Transmission (DTX)'>
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|
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<t>
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Opus can, as described in <xref target='variable-vs-constant-bitrate'/>,
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be operated with a variable bitrate. In that case, the encoder will
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automatically reduce the bitrate for certain input signals, like periods
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of silence. When using continuous transmission, it will reduce the
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bitrate when the characteristics of the input signal permit, but
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will never interrupt the transmission to the receiver. Therefore, the
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received signal will maintain the same high level of audio quality over the
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full duration of a transmission while minimizing the average bit
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rate over time.
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|
</t>
|
|
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<t>
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|
In cases where the bitrate of Opus needs to be reduced even
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further or in cases where only constant bitrate is available,
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|
the Opus encoder can use discontinuous
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|
transmission (DTX), where parts of the encoded signal that
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|
correspond to periods of silence in the input speech or audio signal
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|
are not transmitted to the receiver. A receiver can distinguish
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|
between DTX and packet loss by looking for gaps in the sequence
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|
number, as described by Section 4.1
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of <xref target="RFC3551"/>.
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</t>
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|
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<t>
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|
On the receiving side, the non-transmitted parts will be handled by a
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frame loss concealment unit in the Opus decoder which generates a
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|
comfort noise signal to replace the non transmitted parts of the
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speech or audio signal. Use of <xref target="RFC3389"/> Comfort
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|
Noise (CN) with Opus is discouraged.
|
|
The transmitter MUST drop whole frames only,
|
|
based on the size of the last transmitted frame,
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|
to ensure successive RTP timestamps differ by a multiple of 120 and
|
|
to allow the receiver to use whole frames for concealment.
|
|
</t>
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|
|
|
<t>
|
|
DTX can be used with both variable and constant bitrate.
|
|
It will have a slightly lower speech or audio
|
|
quality than continuous transmission. Therefore, using continuous
|
|
transmission is RECOMMENDED unless constraints on available network bandwidth
|
|
are severe.
|
|
</t>
|
|
|
|
</section>
|
|
|
|
</section>
|
|
|
|
<section title='Complexity'>
|
|
|
|
<t>
|
|
Complexity of the encoder 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 embedding "in-band" forward error correction (FEC)
|
|
data into the Opus bit stream. This FEC scheme adds
|
|
redundant information about the previous packet (N-1) to the current
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|
output packet N. For
|
|
each frame, the encoder decides whether to use FEC based on (1) an
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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
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|
entirely controlled by the encoder and therefore no special precautions
|
|
for the payload have to be taken.
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|
</t>
|
|
|
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<t>
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|
On the receiving side, the decoder can take advantage of this
|
|
additional information when it loses a packet and the next packet
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|
is available. In order to use the FEC data, the jitter buffer needs
|
|
to provide access to payloads with the FEC data.
|
|
Instead of performing loss concealment for a missing packet, the
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receiver can then configure its decoder to decode the FEC data from the next packet.
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</t>
|
|
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<t>
|
|
Any compliant Opus decoder is capable of ignoring
|
|
FEC information when it is not needed, so encoding with FEC cannot cause
|
|
interoperability problems.
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|
However, if FEC cannot be used on the receiving side, then 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
|
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session is set up.
|
|
</t>
|
|
|
|
</section>
|
|
|
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<section title='Stereo Operation'>
|
|
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<t>
|
|
Opus allows for transmission of stereo audio signals. This operation
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|
is signaled in-band in the Opus bit-stream and no special arrangement
|
|
is needed in the payload format. An
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|
Opus decoder is capable of handling a stereo encoding, but an
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|
application might only be capable of consuming a single audio
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channel.
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</t>
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<t>
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If a decoder cannot take advantage of the benefits of a stereo signal
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this SHOULD be indicated at the time a session is set up. In that case
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|
the sending side SHOULD NOT send stereo signals as it leads to an
|
|
inefficient usage of network resources.
|
|
</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'>
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|
<t>The format of the RTP header is specified in <xref target="RFC3550"/>.
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|
The use of the fields of the RTP header by the Opus payload format is
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|
consistent with that specification.</t>
|
|
|
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<t>The payload length of Opus is an integer number of octets and
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|
therefore no padding is necessary. The payload MAY be padded by an
|
|
integer number of octets according to <xref target="RFC3550"/>,
|
|
although the Opus internal padding is preferred.</t>
|
|
|
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<t>The timestamp, sequence number, and marker bit (M) of the RTP header
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|
are used in accordance with Section 4.1
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|
of <xref target="RFC3551"/>.</t>
|
|
|
|
<t>The RTP payload type for Opus is to be assigned dynamically.</t>
|
|
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<t>The receiving side MUST be prepared to receive duplicate RTP
|
|
packets. The receiver MUST provide at most one of those payloads to the
|
|
Opus decoder for decoding, and MUST discard the others.</t>
|
|
|
|
<t>Opus supports 5 different audio bandwidths, which can be adjusted during
|
|
a stream.
|
|
The RTP timestamp is incremented with a 48000 Hz clock rate
|
|
for all modes of Opus and all sampling rates.
|
|
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 data encoded with sampling rates other than 48000 Hz,
|
|
the sampling rate has to be adjusted to 48000 Hz.</t>
|
|
|
|
</section>
|
|
|
|
<section title='Payload Structure'>
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|
<t>
|
|
The Opus encoder can 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, up to a maximum packet duration representing
|
|
120 ms of speech or audio data. The grouping of one or more Opus
|
|
frames into a single Opus packet is defined in Section 3 of
|
|
<xref target="RFC6716"/>. An RTP payload MUST contain exactly one
|
|
Opus packet as defined by that document.
|
|
</t>
|
|
|
|
<t><xref target='payload-structure'/> shows the structure combined with the RTP header.</t>
|
|
|
|
<figure anchor="payload-structure"
|
|
title="Packet structure with RTP header">
|
|
<artwork align="center">
|
|
<![CDATA[
|
|
+----------+--------------+
|
|
|RTP Header| Opus Payload |
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|
+----------+--------------+
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|
]]>
|
|
</artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
<xref target='opus-packetization'/> shows supported frame sizes in
|
|
milliseconds of encoded speech or audio data for the speech and audio modes
|
|
(Mode) and sampling rates (fs) of Opus and shows how the timestamp is
|
|
incremented for packetization (ts incr). If the Opus encoder
|
|
outputs multiple encoded frames into a single packet, the timestamp
|
|
increment is the sum of the increments for the individual frames.
|
|
</t>
|
|
|
|
<texttable anchor='opus-packetization' title="Supported Opus frame
|
|
sizes and timestamp increments marked with an o. Unsupported marked with an x.">
|
|
<ttcol align='center'>Mode</ttcol>
|
|
<ttcol align='center'>fs</ttcol>
|
|
<ttcol align='center'>2.5</ttcol>
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|
<ttcol align='center'>5</ttcol>
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|
<ttcol align='center'>10</ttcol>
|
|
<ttcol align='center'>20</ttcol>
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|
<ttcol align='center'>40</ttcol>
|
|
<ttcol align='center'>60</ttcol>
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|
<c>ts incr</c>
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|
<c>all</c>
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|
<c>120</c>
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|
<c>240</c>
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|
<c>480</c>
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|
<c>960</c>
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|
<c>1920</c>
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|
<c>2880</c>
|
|
<c>voice</c>
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|
<c>NB/MB/WB/SWB/FB</c>
|
|
<c>x</c>
|
|
<c>x</c>
|
|
<c>o</c>
|
|
<c>o</c>
|
|
<c>o</c>
|
|
<c>o</c>
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|
<c>audio</c>
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|
<c>NB/WB/SWB/FB</c>
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|
<c>o</c>
|
|
<c>o</c>
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|
<c>o</c>
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|
<c>o</c>
|
|
<c>x</c>
|
|
<c>x</c>
|
|
</texttable>
|
|
|
|
</section>
|
|
|
|
</section>
|
|
|
|
<section title='Congestion Control'>
|
|
|
|
<t>The target bitrate of Opus can be adjusted at any point in time, thus
|
|
allowing 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 the packet duration. A lower packet
|
|
transmission rate reduces the amount of header overhead, but at the same
|
|
time increases latency and loss sensitivity, so it ought to be used with
|
|
care.</t>
|
|
|
|
<t>Since UDP does not provide congestion control, applications that use
|
|
RTP over UDP SHOULD implement their own congestion control above the
|
|
UDP layer <xref target="RFC5405"/>. Work in the rmcat working group
|
|
<xref target="rmcat"/> describes the
|
|
interactions and conceptual interfaces necessary between the application
|
|
components that relate to congestion control, including the RTP layer,
|
|
the higher-level media codec control layer, and the lower-level
|
|
transport interface, as well as components dedicated to congestion
|
|
control functions.</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="RFC6838"/> 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:"> the RTP timestamp is incremented with a
|
|
48000 Hz clock rate for all modes of Opus and all sampling
|
|
rates. For data encoded with sampling rates other than 48000 Hz,
|
|
the sampling rate has to be adjusted to 48000 Hz.
|
|
</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 maximum duration of media represented
|
|
by a packet (according to Section 6 of
|
|
<xref target="RFC4566"/>) that a decoder wants to receive, in
|
|
milliseconds rounded up to the next full integer value.
|
|
Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
|
|
multiple of an Opus frame size 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, the default is 120.
|
|
<vspace blankLines='1'/></t>
|
|
|
|
<t hangText="ptime:"> the preferred duration of media represented
|
|
by a packet (according to Section 6 of
|
|
<xref target="RFC4566"/>) that a decoder wants to receive, in
|
|
milliseconds rounded up to the next full integer value.
|
|
Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
|
|
multiple of an Opus frame size 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, the default is 20.
|
|
<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 can 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
|
|
6000 to 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
|
|
is 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
|
|
utilization and encoding complexity. If no value is specified,
|
|
the default is 0 (mono).<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 specifies 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, the default is 0
|
|
(mono).<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,
|
|
the default is 0 (vbr). When cbr is 1, the maximum average bitrate can still
|
|
change, 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.
|
|
Providing 0 when FEC cannot be used on the receiving side is
|
|
RECOMMENDED. 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, the
|
|
default is 0.<vspace blankLines='1'/></t>
|
|
</list></t>
|
|
|
|
<t>Encoding considerations:<vspace blankLines='1'/></t>
|
|
<t><list style="hanging">
|
|
<t>The Opus media type is framed and consists of binary data according
|
|
to Section 4.8 in <xref target="RFC6838"/>.</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: RFC [XXXX]</t>
|
|
<t>Note to the RFC Editor: Replace [XXXX] with the number of the published
|
|
RFC.<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 can use this media type. Some examples are,
|
|
but not limited to, audio and video conferencing, Voice over IP,
|
|
media streaming.</t>
|
|
</list></t>
|
|
|
|
<t>Fragment identifier considerations: N/A<vspace blankLines='1'/></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: IETF Payload Working Group delegated from the IESG</t>
|
|
</section>
|
|
</section>
|
|
|
|
<section title='SDP Considerations'>
|
|
<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 Opus,
|
|
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", "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.,
|
|
maxplaybackrate=48000). 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 desired,
|
|
DTX is not desired</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='SDP Offer/Answer Considerations'>
|
|
|
|
<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
|
|
restricted by this payload format description. The decoder MUST be
|
|
capable of decoding 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, some values might cause application
|
|
performance to 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 "maxplaybackrate" parameter is a unidirectional receive-only
|
|
parameter that reflects limitations of the local receiver. When
|
|
sending to a single destination, a sender MUST NOT use an audio
|
|
bandwidth higher than necessary to make full use of audio sampled at
|
|
a sampling rate of "maxplaybackrate". Gateways or senders that
|
|
are sending the same encoded audio to multiple destinations
|
|
SHOULD NOT use an audio bandwidth higher than necessary to
|
|
represent audio sampled at "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 bitrate, 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, some values might cause
|
|
application performance to suffer, and ought to 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. When sending to a single destination, a sender MUST
|
|
NOT use stereo when "stereo" is 0. Gateways or senders that are
|
|
sending the same encoded audio to multiple destinations SHOULD
|
|
NOT use stereo when "stereo" is 0, as this would lead to
|
|
inefficient use of network resources. The "stereo" parameter does
|
|
not affect interoperability.
|
|
</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>
|
|
|
|
<t>
|
|
The Opus parameters in an SDP Offer/Answer exchange are completely
|
|
orthogonal, and there is no relationship between the SDP Offer and
|
|
the Answer.
|
|
</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", "maxplaybackrate", and
|
|
"maxaveragebitrate" ought to 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 of the payload
|
|
format configuration are recommendations by the decoding side to ensure
|
|
the best performance for the decoder.
|
|
</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 can be provided if necessary
|
|
by declaring multiple RTP payload types; however, the number of types
|
|
ought to be kept small.</t>
|
|
</list></t>
|
|
</section>
|
|
</section>
|
|
|
|
<section title='Security Considerations' anchor='security-considerations'>
|
|
|
|
<t>Use of variable bitrate (VBR) is subject to the security considerations in
|
|
<xref target="RFC6562"/>.</t>
|
|
|
|
<t>RTP packets using the payload format defined in this specification
|
|
are subject to the security considerations discussed in the RTP
|
|
specification <xref target="RFC3550"/>, and in any applicable RTP profile such as
|
|
RTP/AVP <xref target="RFC3551"/>, RTP/AVPF <xref target="RFC4585"/>,
|
|
RTP/SAVP <xref target="RFC3711"/> or RTP/SAVPF <xref target="RFC5124"/>.
|
|
However, as "Securing the RTP Protocol Framework:
|
|
Why RTP Does Not Mandate a Single Media Security Solution"
|
|
<xref target="RFC7202"/> discusses, it is not an RTP payload
|
|
format's responsibility to discuss or mandate what solutions are used
|
|
to meet the basic security goals like confidentiality, integrity and
|
|
source authenticity for RTP in general. This responsibility lays on
|
|
anyone using RTP in an application. They can find guidance on
|
|
available security mechanisms and important considerations in Options
|
|
for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options].
|
|
Applications SHOULD use one or more appropriate strong security
|
|
mechanisms.</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>Many people have made useful comments and suggestions contributing to this document.
|
|
In particular, we would like to thank
|
|
Tina le Grand, Cullen Jennings, Jonathan Lennox, Gregory Maxwell, Colin Perkins, Jan Skoglund,
|
|
Timothy B. Terriberry, Martin Thompson, Justin Uberti, Magnus Westerlund, and Mo Zanaty.</t>
|
|
</section>
|
|
</middle>
|
|
|
|
<back>
|
|
<references title="Normative References">
|
|
&rfc2119;
|
|
&rfc3389;
|
|
&rfc3550;
|
|
&rfc3711;
|
|
&rfc3551;
|
|
&rfc6838;
|
|
&rfc4855;
|
|
&rfc4566;
|
|
&rfc3264;
|
|
&rfc2326;
|
|
&rfc5576;
|
|
&rfc6562;
|
|
&rfc6716;
|
|
</references>
|
|
|
|
<references title="Informative References">
|
|
&rfc2974;
|
|
&rfc4585;
|
|
&rfc5124;
|
|
&rfc5405;
|
|
&rfc7202;
|
|
|
|
<reference anchor='rmcat' target='https://datatracker.ietf.org/wg/rmcat/documents/'>
|
|
<front>
|
|
<title>rmcat documents</title>
|
|
<author/>
|
|
<date/>
|
|
<abstract>
|
|
<t></t>
|
|
</abstract></front>
|
|
</reference>
|
|
|
|
|
|
</references>
|
|
|
|
</back>
|
|
</rfc>
|