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Versions: (draft-lennox-avt-rtp-audio-level-exthdr) 00 01 02 03 04 05 06 RFC 6464

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

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
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   This Internet-Draft will expire on December 4, 2011.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of



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   publication of this document.  Please review these documents
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   it for publication as an RFC or to translate it into languages other
   than English.

































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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.

   [I-D.lennox-avtcore-srtp-encrypted-header-ext]
              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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