AVT J. Lennox, Ed.
Internet-Draft Vidyo
Intended status: Standards Track E. Ivov
Expires: December 4, 2011 Jitsi
E. Marocco
Telecom Italia
June 2, 2011
A Real-Time Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication
draft-ietf-avtext-client-to-mixer-audio-level-02
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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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Audio Levels . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Signaling (Setup) Information . . . . . . . . . . . . . . . . 6
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 -01 . . . . . . . . . . . . . . . . . . 9
A.2. Changes From Draft -00 . . . . . . . . . . . . . . . . . . 9
A.3. Changes From Individual Submission Draft -01 . . . . . . . 10
A.4. 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 should 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 element carries the level of the
audio in the RTP payload of the packet it is associated with, and
also an indication as to whether voice activity has been detected in
the packet. This information is carried in an RTP header extension
element as defined by [RFC5285].
The payload of the audio level header extension element is as
follows:
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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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1
The length field takes the value 0 to indicate that 1 byte follows.
The two-byte header defined in RFC 5285 [RFC5285] may also be used.
The magnitude of the audio level is packed into the seven least
significant bits of the single byte of the header extension, shown in
Figure 1. 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
source, MAY be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
Implementations MAY choose to measure audio levels prior to encoding
them in the payload carried in the RTP payload, e.g. on raw linear
PCM input.
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.
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
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values from raw linear PCM audio samples.
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.
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 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.
4. Signaling (Setup) Information
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).
5. Considerations on Use
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.)
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
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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
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, 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.)
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. One solution is header
extension encryption [I-D.lennox-avtcore-srtp-encrypted-header-ext].
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
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Description: Audio Level
Contact: jonathan@vidyo.com
Reference: RFC XXXX
Note to RFC Editor: please replace "RFC XXXX" with the number of this
RFC.
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
[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-02 (work in
progress), May 2011.
[]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-lennox-avtcore-srtp-encrypted-header-ext-00 (work in
progress), March 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.
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[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.
[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 -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.
o Other editorial clarifications.
A.2. Changes From Draft -00
o Added references to the sample level calculator in
[I-D.ietf-avtext-mixer-to-client-audio-level].
o Changed affiliation for Emil Ivov.
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A.3. 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.4. 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.lennox-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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