AVT J. Lennox, Ed.
Internet-Draft Vidyo
Intended status: Standards Track E. Ivov
Expires: March 23, 2012 Jitsi
E. Marocco
Telecom Italia
September 20, 2011
A Real-Time Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication
draft-ietf-avtext-client-to-mixer-audio-level-05
Abstract
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.
Status of this Memo
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This Internet-Draft will expire on March 23, 2012.
Copyright Notice
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document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Audio Levels . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Signaling (Setup) Information . . . . . . . . . . . . . . . . 5
5. Considerations on Use . . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . 7
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8
8.1. Normative References . . . . . . . . . . . . . . . . . . . 8
8.2. Informative References . . . . . . . . . . . . . . . . . . 8
Appendix A. Changes From Earlier Versions . . . . . . . . . . . . 9
A.1. Changes From Draft -03 . . . . . . . . . . . . . . . . . . 9
A.2. Changes From Draft -02 . . . . . . . . . . . . . . . . . . 9
A.3. Changes From Draft -01 . . . . . . . . . . . . . . . . . . 9
A.4. Changes From Individual Submission Draft -01 . . . . . . . 10
A.5. Changes From Individual Submission Draft -00 . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 10
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1. Introduction
In a centralized Real-Time Transport Protocol (RTP) [RFC3550] 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 are intended
to be forwarded to the other conference participants.
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.
As an alternative, this document defines an RTP header extension
[RFC5285] through which senders of audio packets can indicate the
audio level of the packets' payload, reducing the processing load for
a server.
The header extension in this draft is different than, but
complementary with, the one defined in
[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.
2. Terminology
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 RFC 2119 [RFC2119] and
indicate requirement levels for compliant implementations.
3. Audio Levels
The audio level header extension carries the level of the audio in
the RTP [RFC3550] payload of the packet it is associated with. This
information is carried in an RTP header extension element as defined
by the "General Mechanism for RTP Header Extensions" [RFC5285].
The payload of the audio level header extension element can be
encoded using the one-byte or the two-byte header defined in
[RFC5285]. Figure 1 and Figure 2 show sample audio level encodings
with each of them.
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0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len=0 |V| level |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Sample audio level encoding using the one-byte header format
Figure 1
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) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Sample audio level encoding using the two-byte header format
Figure 2
Note that, as indicated in [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.
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 Figure 1 and Figure 2. 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.
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 u-law (audio/pcmu) audio [ITU.G711.1988], 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.)
The audio level for digital silence, for example for a muted audio
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source, MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
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.
To simplify implementation of the encoding procedures described here,
the reference implementation section in
[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.
In addition, a flag bit (labeled V) optionally indicates whether the
encoder believes the audio packet contains voice activity. If the V
bit is in use, the value 1 indicates that the encoder believes the
audio packet contains voice activity, and the value 0 indicates that
the encoder believes it does not. (The voice activity detection
algorithm is unspecified and left implementation-specific.) If the V
bit is not in use, its value is unspecified and MUST be ignored by
receivers. The use of the V bit is signaled using the extension
attribute "vad", discussed in Section 4.
When this header extension is used with RTP data sent using the RTP
Payload for Redundant Audio Data [RFC2198], the header's data
describes the contents of the primary encoding.
Note: This audio level is defined in the same manner as is audio
noise level in the RTP Payload Comfort Noise specification [RFC3389].
In the comfort noise 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.
4. Signaling (Setup) Information
The URI for declaring this header extension in an extmap attribute is
"urn:ietf:params:rtp-hdrext:ssrc-audio-level".
It has a single extension attribute, named "vad". It takes the form
"vad=on" or "vad=off". If the header extension element is signaled
with "vad=on", the "V" bit described in Section 3 is in use, and MUST
be set by senders. If the header extension element is signaled with
"vad=off", the "V" bit is not in use, and its value MUST be ignored
by receivers. If the "vad" extension attribute is not specified, the
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default is "vad=on".
An example attribute line in the SDP, for a conference might hence
be:
a=extmap:6 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on
The "vad" extension attribute only controls the semantics of this
header extension attribute, and does not make any statement about
whether the sender is using any other voice activity detection
features such as discontinuous transmission, comfort noise, or
silence suppression.
Using the mechanisms of [RFC5285], an endpoint MAY signal multiple
instances of the header extension element, with different values of
the vad attribute, so long as these instances use different values
for the extension identifier. However, again following the rules of
[RFC5285], the semantics chosen for a header extension element
(including its vad setting) for a particular extension identifier
value MUST NOT be changed within an RTP session.
5. Considerations on Use
Mixers and forwarders generally ought not base audio forwarding
decisions directly on packet-by-packet audio level information, but
rather ought to 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.)
First of all, audio levels generally ought to 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
ought to 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.
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
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average audio levels during speech, or by measuring background audio
levels (average audio level levels during non-speech).
6. Security Considerations
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, if a device relys on audio level data from
untrusted endpoints, it SHOULD periodically audit the level
information transmitted, taking appropriate corrective action against
endpoints that appear to be sending incorrect data. (However, as it
is valid for an endpoint to choose to measure audio levels prior to
encoding, some degree of discrepancy could be present. This would
not indicate that an endpoint is malicous.)
In the Secure Real-Time Transport Protocol (SRTP) [RFC3711], 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 [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. This mechanism could be
header extension encryption
[I-D.ietf-avtcore-srtp-encrypted-header-ext], or a lower-level
security and authentication mechanism.
7. IANA Considerations
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:
Extension URI: urn:ietf:params:rtp-hdrext:ssrc-audio-level
Description: Audio Level
Contact: jonathan@vidyo.com
Reference: RFC XXXX
Note to RFC Editor: please replace "RFC XXXX" with the number of this
RFC.
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8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
8.2. Informative References
[]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-ietf-avtcore-srtp-encrypted-header-ext-00 (work in
progress), June 2011.
[I-D.ietf-avtext-mixer-to-client-audio-level]
Ivov, E., Marocco, E., and J. Lennox, "A Real-Time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication",
draft-ietf-avtext-mixer-to-client-audio-level-05 (work in
progress), September 2011.
[I-D.perkins-avt-srtp-vbr-audio]
Perkins, C. and J. Valin, "Guidelines for the use of
Variable Bit Rate Audio with Secure RTP",
draft-perkins-avt-srtp-vbr-audio-05 (work in progress),
December 2010.
[ITU.G711.1988]
International Telecommunications Union, "Pulse Code
Modulation (PCM) of Voice Frequencies", ITU-
T Recommendation G.711, November 1988.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
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[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
Appendix A. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
A.1. Changes From Draft -03
o Added vad extension attribute to negotiate use of the V bit.
o Addressed editorial comments made on the mailing list.
A.2. Changes From Draft -02
o Changed encoding related text so that it would cover both the one-
byte and the two-byte header formats.
o Clarified use of root mean square for dBov calculation
o Added references to the sample level calculator in
[I-D.ietf-avtext-mixer-to-client-audio-level].
o Changed affiliation for Emil Ivov.
o Other minor editorial changes.
A.3. Changes From Draft -01
o 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
[I-D.ietf-avtext-mixer-to-client-audio-level].
o Removed the "Limitations" section; it was discussing a potential
extension that consensus indicated was out of scope of this
document.
o 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.
o 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.
o Editorial changes for consistency with
[I-D.ietf-avtext-mixer-to-client-audio-level].
o Moved several descriptions of normative items that previously had
only been described in informative sections of the text.
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o Other editorial clarifications.
A.4. Changes From Individual Submission Draft -01
o This version is primarily a document refresh.
o Emil Ivov and Enrico Marocco have been added as co-authors.
o Additional open issues listed.
A.5. Changes From Individual Submission Draft -00
o The draft name has been changed to clarify that this document
defines Client-To-Mixer Audio Levels, to more clearly distinguish
it from [I-D.ietf-avtext-mixer-to-client-audio-level].
o 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.
o The sections Considerations on Use (Section 5) and Limitations
have been added.
o It has been noted that senders MAY indicate -127 dBov for digital
silence, and that level measurement MAY be done prior to encoding
audio.
o A reference to [I-D.ietf-avtcore-srtp-encrypted-header-ext] has
been added to the security considerations.
o The term "header extension" is now used consistentenly throughout
the document (as opposed to "extension header").
Authors' Addresses
Jonathan Lennox (editor)
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
Emil Ivov
Jitsi
Strasbourg 67000
France
Email: emcho@jitsi.org
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Enrico Marocco
Telecom Itialia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
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