One document matched: draft-ietf-avtext-client-to-mixer-audio-level-03.xml
<?xml version='1.0'?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd" [
<!ENTITY hdrext SYSTEM
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<!ENTITY slic SYSTEM
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<!ENTITY srtp SYSTEM
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<!ENTITY hdrcrypt SYSTEM
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<!ENTITY vbr-srtp SYSTEM
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<?rfc toc="yes" ?>
<?rfc strict="yes" ?>
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<rfc category='std' ipr='pre5378Trust200902'
docName='draft-ietf-avtext-client-to-mixer-audio-level-03'>
<front>
<title abbrev='Client-to-Mixer Audio Level Indication'>
A Real-Time Transport Protocol (RTP) Header Extension for
Client-to-Mixer Audio Level Indication
</title>
<author initials='J.' surname='Lennox'
fullname='Jonathan Lennox'
role="editor">
<organization abbrev='Vidyo'>
Vidyo, Inc.
</organization>
<address>
<postal>
<street>433 Hackensack Avenue</street>
<street>Seventh Floor</street>
<city>Hackensack</city> <region>NJ</region>
<code>07601</code>
<country>US</country>
</postal>
<email>jonathan@vidyo.com</email>
</address>
</author>
<author initials='E.' surname='Ivov'
fullname='Emil Ivov'>
<organization abbrev='Jitsi'>
Jitsi
</organization>
<address>
<postal>
<street> </street>
<city>Strasbourg</city>
<code>67000</code>
<country>France</country>
</postal>
<email>emcho@jitsi.org</email>
</address>
</author>
<author initials='E.' surname='Marocco'
fullname='Enrico Marocco'>
<organization abbrev='Telecom Italia'>
Telecom Itialia
</organization>
<address>
<postal>
<street>Via G. Reiss Romoli, 274</street>
<city>Turin</city>
<code>10148</code>
<country>Italy</country>
</postal>
<email>enrico.marocco@telecomitalia.it</email>
</address>
</author>
<date />
<area>RAI</area>
<workgroup>AVT</workgroup>
<keyword>I-D</keyword>
<keyword>Internet-Draft</keyword>
<!-- TODO: more keywords -->
<abstract>
<t>This document defines a mechanism by which packets of
Real-Time Transport Protocol (RTP) audio streams can indicate,
in an RTP header extension, the audio level of the audio
sample carried in the RTP packet. In large conferences, this
can reduce the load on an audio mixer or other middlebox which
wants to forward only a few of the loudest audio streams,
without requiring it to decode and measure every stream that
is received.</t>
</abstract>
</front>
<middle>
<section title='Introduction' anchor='introduction'>
<t>In a centralized <xref target='RFC3550'>Real-Time Transport Protocol
(RTP)</xref> audio conference, an audio mixer or
forwarder receives audio streams from many or all of the conference
participants. It then selectively forwards some of them to
other participants in the conference. In
large conferences, it is possible that such a server might be
receiving a large number of streams, of which only a few should be
forwarded to the other conference participants.</t>
<t>In such a scenario, in order to pick the audio streams to forward,
a centralized server needs to decode, measure audio levels, and
possibly perform voice activity detection on audio data from a large
number of streams. The need for such processing limits the size or
number of conferences such a server can support.</t>
<t>As an alternative, this document defines
an <xref target='RFC5285'>RTP header extension</xref> through which
senders of audio packets can indicate the audio level of the
packets' payload, reducing the processing load for a server.
</t>
<t>The header extension in this draft is different than, but
complementary with, the one defined in
<xref target='I-D.ietf-avtext-mixer-to-client-audio-level' />, which
defines a mechanism by which audio mixers can indicate to clients the
levels of the contributing sources that made up the mixed audio.</t>
</section>
<section title='Terminology'>
<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'>RFC 2119</xref> and indicate requirement levels for
compliant implementations.</t>
</section>
<section title='Audio Levels'>
<t>The audio level header extension carries the level of the audio
in the RTP payload of the packet it is associated with. This information
is carried in an RTP header extension element as defined by the
<xref target='RFC5285'>"General Mechanism for RTP Header Extensions"
</xref>.</t>
<t>The payload of the audio level header extension element can be
encoded using the one or the two-byte header defined in <xref
target='RFC5285'/>. <xref target='exthdr'/> and <xref target='exthdr2'/>
show sample audio level encodings with each of them.</t>
<figure anchor='exthdr'>
<artwork>
<![CDATA[
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len=0 |V| level |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]>
</artwork>
<postamble>Sample audio level encoding using the one-byte header format
</postamble>
</figure>
<figure anchor='exthdr2'>
<artwork>
<![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len=1 |V| level | 0 (pad) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]>
</artwork>
<postamble>Sample audio level encoding using the two-byte header format
</postamble>
</figure>
<t>Note that, as indicated in <xref target='RFC5285'/> length field in
the one-byte header format takes the value 0 to indicate that 1 byte
follows. In the two-byte header format on the other hand it takes the
value of 1.</t>
<t>The magnitude of the audio level itself is packed into the seven
least significant bits of the single byte of the header extension,
shown in <xref target='exthdr'/> and <xref target='exthdr2'/>. The least
significant bit of the audio level magnitude is packed into the least
significant bit of the byte. The most significant bit of the byte is
used as a separate flag bit "V", defined below.</t>
<t>The audio level is expressed in -dBov, with values from 0 to 127
representing 0 to -127 dBov. dBov is the level, in decibels, relative
to the overload point of the system, i.e. the maximum-amplitude signal
that can be handled by the system without clipping. (Note:
Representation relative to the overload point of a system is
particularly useful for digital implementations, since one does not
need to know the relative calibration of the analog circuitry.) For
example, in the case of <xref target='ITU.G711.1988'>u-law (audio/pcmu)
audio</xref>, the 0 dBov reference would be a square wave with values
+/- 8031. (This translates to 6.18 dBm0, relative to u-law's dBm0
definition in Table 6 of G.711.)</t>
<t>The audio level for digital silence, for example for a muted audio
source, MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.</t>
<t> The audio level header extension only carries the level of the
audio in the RTP payload of the packet it is associated with, with no
long-term averaging or smoothing applied. That level is measured as a
root mean square of all the samples in the measured range.</t>
<t>To simplify implementation of the encoding procedures described here,
the reference implementation section in <xref
target='I-D.ietf-avtext-mixer-to-client-audio-level' /> provides a
sample Java implementation of an audio level calculator that helps
obtain such values from raw linear PCM audio samples.</t>
<t>In addition, a flag bit (labeled V) indicates whether the encoder
believes the audio packet contains voice activity (1) or does not (0).
The voice activity detection algorithm is unspecified and left
implementation-specific.</t>
<t>When this header extension is used with RTP data sent using
<xref target='RFC2198'>the RTP Payload for Redundant Audio Data</xref>,
the header's data describes the contents of the primary encoding.</t>
<t> Note: This audio level is defined in the same manner as is audio
noise level in the <xref target="RFC3389">RTP Payload Comfort Noise
specification</xref>. In the comfort noice specification, the overall
magnitude of the noise level in comfort noise is encoded into the first
byte of the payload, with spectral information about the noise in
subsequent bytes. This specification's audio level parameter is defined
so as to be identical to the comfort noise payload's noise-level byte.
</t>
</section>
<section title='Signaling (Setup) Information'>
<t>The URI for declaring this header extension in an extmap attribute is
"urn:ietf:params:rtp-hdrext:ssrc-audio-level". There is no
additional setup information needed for this extension (i.e. no
extensionattributes).</t>
</section>
<section title='Considerations on Use' anchor='use'>
<t>Mixers and forwarders generally should not base audio forwarding
decisions directly on packet-by-packet audio level information, but
rather should apply some analysis of the audio levels and trends.
This general rule applies whether audio levels are provided by
endpoints (as defined in this document), or are calculated at a
server, as would be done in the absence of this information. This
section discusses several issues that mixers and forwarders may
wish to take into account. (Note that this section provides design
guidance only, and is not normative.)</t>
<t>First of all, audio levels should generally be measured over longer
intervals than that of a single audio packet. In order to avoid
false-positives for short bursts of sound (such as a cough or a
dropped microphone), it is often useful to require that a
participant's audio level be maintained for some period of time
before considering it to be "real", i.e. some type of low-pass
filter should be applied to the audio levels. Note, though, that such
filtering must be balanced with the need to avoid clipping of
the beginning of a speaker's speech.</t>
<t>Additionally, different participants may have their audio input set
differently. It may be useful to apply some sort of automatic gain
control to the audio levels. There are a number of possible
approaches to acheiving this, e.g. by measuring peak audio levels,
by average audio levels during speech, or by measuring background
audio levels (average audio level levels during non-speech).</t>
</section>
<section title='Security Considerations' anchor='security'>
<t>A malicious endpoint could choose to set the values in this
header extension falsely, so as to falsely claim that audio or voice
is or is not present. It is not clear what could be gained by
falsely claiming that audio is not present, but an endpoint falsely
claiming that audio is present could perform a denial-of-service
attack on an audio conference, so as to send silence to suppress
other conference members' audio. Thus, a device relying on audio
level data from untrusted endpoints SHOULD periodically audit the
level information transmitted, taking appropriate corrective action
if endpoints appear to be sending incorrect data. (Note that
as it is valid for an endpoint to choose to measure audio levels prior to encoding,
some degree of discrepancy SHOULD be tolerated.)</t>
<t>In the <xref target='RFC3711'>Secure Real-Time Transport Protocol
(SRTP)</xref>, RTP header extensions are authenticated but not
encrypted. When this header extension is used, audio levels are
therefore visible on a packet-by-packet basis to an attacker
passively observing the audio stream. As discussed in
<xref target='I-D.perkins-avt-srtp-vbr-audio' />, such an attacker
might be able to infer information about the conversation,
possibly with phoneme-level resolution. In scenarios where this is a
concern, additional mechanisms SHOULD be used to protect the
confidentiality of the header extension. One solution is
<xref target='I-D.lennox-avtcore-srtp-encrypted-header-ext'>header
extension encryption</xref>.</t>
</section>
<section title='IANA Considerations' anchor='iana'>
<t>This document defines a new extension URI to the RTP Compact Header
Extensions subregistry of the Real-Time Transport Protocol (RTP)
Parameters registry, according to the following data:</t>
<t>
<list style='hanging'>
<t hangText='Extension URI:'>urn:ietf:params:rtp-hdrext:ssrc-audio-level</t>
<t hangText='Description:'>Audio Level</t>
<t hangText='Contact:'>jonathan@vidyo.com</t>
<t hangText='Reference:'>RFC &rfc.number;</t>
</list>
</t>
<t>Note to RFC Editor: please replace "RFC &rfc.number;" with the number of this RFC.</t>
</section>
</middle>
<back>
<references title='Normative References'>
&hdrext;
&red;
&rfc2119;
&rtp;
</references>
<references title='Informative References'>
<?rfc include="reference.RFC.3389"?>
<reference anchor="ITU.G711.1988">
<front>
<title>Pulse Code Modulation (PCM) of Voice Frequencies</title>
<author>
<organization>
International Telecommunications Union
</organization>
</author>
<date month="November" year="1988" />
</front>
<seriesInfo name="ITU-T" value="Recommendation G.711" />
</reference>
&slic;
&srtp;
&vbr-srtp;
&hdrcrypt;
</references>
<section title='Changes From Earlier Versions'>
<t>Note to the RFC-Editor: please remove this section prior to publication
as an RFC.</t>
<section title='Changes From Draft -02'>
<t>
<list style='symbols'>
<t>Changed encoding related text so that it would cover both
the one-byte and the two-byte header formats.</t>
<t>Clarified use of root mean square for dBov calculation
</t>
<t>Other minor editorial changes.</t>
</list>
</t>
</section>
<section title='Changes From Draft -01'>
<t><list style='symbols'>
<t>Changed the URI for declaring this header extension from
"urn:ietf:params:rtp-hdrext:audio-level" to
"urn:ietf:params:rtp-hdrext:ssrc-audio-level" for consistency with
<xref target="I-D.ietf-avtext-mixer-to-client-audio-level"/>.</t>
<t>Removed the "Limitations" section; it was discussing a potential extension that
consensus indicated was out of scope of this document.</t>
<t>Closed the P.56 open issue. It was agreed on IETF 80 that P.56 is
mostly about speech levels and the levels transported by the extension
defined here should also be able to serve as an indication for noise.
</t>
<t>Closed the open issue about transmitting noise floor information.
Noise floor is (loosely) inferrable by observing the per-packet level
information over a period of time, so the additional complexity seemed
unnecessary.</t>
<t>Editorial changes for consistency with <xref
target="I-D.ietf-avtext-mixer-to-client-audio-level"/>.</t>
<t>Moved several descriptions of normative items that previously had only been described
in informative sections of the text.</t>
<t>Other editorial clarifications.</t>
</list></t>
</section>
<section title='Changes From Draft -00'>
<t><list style='symbols'>
<t>Added references to the sample level calculator in
<xref target='I-D.ietf-avtext-mixer-to-client-audio-level' />.</t>
<t>Changed affiliation for Emil Ivov.</t>
</list></t>
</section>
<section title='Changes From Individual Submission Draft -01'>
<t><list style='symbols'>
<t>This version is primarily a document refresh.</t>
<t>Emil Ivov and Enrico Marocco have been added as co-authors.</t>
<t>Additional open issues listed.</t>
</list></t>
</section>
<section title='Changes From Individual Submission Draft -00'>
<t><list style='symbols'>
<t>The draft name has been changed to clarify that this document
defines Client-To-Mixer Audio Levels, to more clearly distinguish
it from <xref target='I-D.ietf-avtext-mixer-to-client-audio-level' />.
</t>
<t>The header extension format has been changed from a two-byte to a
one-byte payload, eliminating the 7 reserved bits and the one
must-be-zero bit.</t>
<t>The sections <xref target='use'>Considerations on Use</xref> and
Limitations have been added.</t>
<t>It has been noted that senders MAY indicate -127 dBov for digital
silence, and that level measurement MAY be done prior to encoding
audio.</t>
<t>A reference
to <xref target='I-D.lennox-avtcore-srtp-encrypted-header-ext' />
has been added to the security considerations.</t>
<t>The term "header extension" is now used consistentenly throughout
the document (as opposed to "extension header").</t>
</list></t>
</section>
</section>
</back>
</rfc>
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