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Commit 5771a672 authored by Jean-Marc Valin's avatar Jean-Marc Valin
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IESG RTP draft update

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doc/draft-ietf-payload-rtp-opus.xml
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......@@ -8,17 +8,20 @@
<!ENTITY rfc6838 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6838.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 rfc4585 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4585.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 rfc5124 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5124.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 rfc7202 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.7202.xml'>
<!ENTITY nbsp "&#160;">
]>
<rfc category="std" ipr="trust200902" docName="draft-ietf-payload-rtp-opus-08">
<rfc category="std" ipr="trust200902" docName="draft-ietf-payload-rtp-opus-09">
<?xml-stylesheet type='text/xsl' href='rfc2629.xslt' ?>
<?rfc strict="yes" ?>
......@@ -71,14 +74,15 @@
</address>
</author>
<date day='6' month='February' year='2015' />
<date day='10' month='April' year='2015' />
<abstract>
<t>
This document defines the Real-time Transport Protocol (RTP) payload
format for packetization of Opus encoded
speech and audio data necessary to integrate the codec in the
most compatible way. Further, it describes media type registrations
most compatible way. It also provides an applicability statement
for the use of Opus over RTP. Further, it describes media type registrations
for the RTP payload format.
</t>
</abstract>
......@@ -100,7 +104,9 @@
<xref target="RFC3550"/> payload format for packetization
of Opus encoded speech and audio data necessary to
integrate Opus in the
most compatible way. Further, it describes media type registrations for
most compatible way. It also provides an applicability statement
for the use of Opus over RTP.
Further, it describes media type registrations for
the RTP payload format.
</t>
</section>
......@@ -179,7 +185,7 @@
<t>
Opus is highly scalable in terms of audio
bandwidth, bitrate, and complexity. Further, Opus allows
transmitting stereo signals.
transmitting stereo signals with in-band signaling in the bit-stream.
</t>
<section title='Network Bandwidth'>
......@@ -268,7 +274,7 @@
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 restraints on available network bandwidth
transmission is RECOMMENDED unless constraints on available network bandwidth
are severe.
</t>
......@@ -327,7 +333,7 @@
<t>
Opus allows for transmission of stereo audio signals. This operation
is signaled in-band in the Opus payload and no special arrangement
is signaled in-band in the Opus bit-stream and no special arrangement
is needed in the payload format. An
Opus decoder is capable of handling a stereo encoding, but an
application might only be capable of consuming a single audio
......@@ -368,7 +374,7 @@
Opus decoder for decoding, and MUST discard the others.</t>
<t>Opus supports 5 different audio bandwidths, which can be adjusted during
a call.
a stream.
The RTP timestamp is incremented with a 48000 Hz clock rate
for all modes of Opus and all sampling rates.
The unit
......@@ -463,8 +469,14 @@
time increases latency and loss sensitivity, so it ought to be used with
care.</t>
<t>It is RECOMMENDED that senders of Opus encoded data apply congestion
control.</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="draft-ietf-rmcat-app-interaction-01"/> 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'>
......@@ -876,6 +888,26 @@
specification <xref target="RFC3550"/> and any profile from,
e.g., <xref target="RFC3711"/> or <xref target="RFC3551"/>.</t>
<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
formats 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 transports Opus encoded speech or audio data.
Hence, security issues include confidentiality, integrity protection, and
authentication of the speech or audio itself. Opus does not provide
......@@ -915,6 +947,30 @@
<references title="Informative References">
&rfc2974;
&rfc4585;
&rfc5124;
&rfc7202;
<reference anchor='draft-ietf-rmcat-app-interaction-01' target='http://tools.ietf.org/html/draft-ietf-rmcat-app-interaction-01'>
<front>
<title>RTP Application Interaction with Congestion Control</title>
<author initials='M.' surname='Zanaty' fullname='M. Zanaty'>
<organization /></author>
<author initials='V.' surname='Singh' fullname='V. Singh'>
<organization /></author>
<author initials='S.' surname='Nandakumar' fullname='S. Nandakumar'>
<organization /></author>
<author initials='Z.' surname='Sarker' fullname='Z. Sarker'>
<organization /></author>
<date year='2014' month='October' />
<abstract>
<t></t>
</abstract></front>
<seriesInfo name='Internet-Draft' value='draft-ietf-rmcat-app-interaction-01' />
<format type='TXT' target='http://tools.ietf.org/html/draft-ietf-rmcat-app-interaction-01' />
</reference>
</references>
</back>
......
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