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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: May 17, 2012                                              Jitsi
                                                              E. Marocco
                                                          Telecom Italia
                                                       November 14, 2011


 A Real-Time Transport Protocol (RTP) Header Extension for  Client-to-
                      Mixer Audio Level Indication
            draft-ietf-avtext-client-to-mixer-audio-level-06

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
   Task Force (IETF).  Note that other groups may also distribute
   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
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on May 17, 2012.

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
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


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 -05 . . . . . . . . . . . . . . . . . .  9
     A.2.  Changes From Draft -04 . . . . . . . . . . . . . . . . . .  9
     A.3.  Changes From Draft -03 . . . . . . . . . . . . . . . . . .  9
     A.4.  Changes From Draft -02 . . . . . . . . . . . . . . . . . . 10
     A.5.  Changes From Draft -01 . . . . . . . . . . . . . . . . . . 10
     A.6.  Changes From Individual Submission Draft -01 . . . . . . . 10
     A.7.  Changes From Individual Submission Draft -00 . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11






















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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 highest-intensity
   signal encodable by the payload format.  (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, MUST be represented as 127 (-127 dBov), regardless of the



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   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.  For payload formats that
   contain extra error-correction bits or loss-concealment information,
   the level corresponds only to the data that would result from the
   payload's normal decoding process, not what it would produce under
   error or packet loss concealment.  The level is measured as a root
   mean square of all the samples in the audio encoded by the packet.

   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



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   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
   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



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   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, or could dominate a conference (by seizing its
   speaker-selection algorithm) without actually speaking.  Thus, if a
   device relies 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.ietf-avtcore-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 MUST 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 such as IPsec [RFC4301].


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:







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   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.


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.

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, December 2005.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

8.2.  Informative References

   [I-D.ietf-avtcore-srtp-encrypted-header-ext]
              Lennox, J., "Encryption of Header Extensions in the Secure
              Real-Time Transport Protocol (SRTP)",
              draft-ietf-avtcore-srtp-encrypted-header-ext-01 (work in
              progress), October 2011.

   [I-D.ietf-avtcore-srtp-vbr-audio]
              Perkins, C. and J. Valin, "Guidelines for the use of
              Variable Bit Rate Audio with Secure RTP",
              draft-ietf-avtcore-srtp-vbr-audio-03 (work in progress),
              July 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-



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              Client Audio Level Indication",
              draft-ietf-avtext-mixer-to-client-audio-level-05 (work in
              progress), September 2011.

   [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 -05

   o  Added an informative reference to RFC 4301 (IPsec).  (Brought up
      by Stephen Farrell)
   o  Clarified the meaning of "overload point of the system".  (Brought
      up by Robert Sparks).
   o  Clarified that levels correspond only to the audio carried in the
      normal decoding process, not error or packet loss concealment.
      (Brought up by Robert Sparks).
   o  Added security consideration that false audio levels could be used
      to seize a speaker-selection algorithm (Brought up by Robert
      Sparks and Stewart Bryant).
   o  Updated reference to [I-D.ietf-avtcore-srtp-vbr-audio].

A.2.  Changes From Draft -04

   o  Adjusted IPR header.

A.3.  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.







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A.4.  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.5.  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.6.  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.7.  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.




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   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


   Enrico Marocco
   Telecom Itialia
   Via G. Reiss Romoli, 274
   Turin  10148
   Italy

   Email: enrico.marocco@telecomitalia.it













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