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PROPOSED STANDARD

Internet Engineering Task Force (IETF)                    J. Lennox, Ed.
Request for Comments: 6464                                         Vidyo
Category: Standards Track                                        E. Ivov
ISSN: 2070-1721                                                    Jitsi
                                                              E. Marocco
                                                          Telecom Italia
                                                           December 2011


       A Real-time Transport Protocol (RTP) Header Extension for
                 Client-to-Mixer Audio Level Indication

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 that 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 is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc6464.
















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RFC 6464         Client-to-Mixer Audio Level Indication    December 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
   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 ....................................................2
   2. Terminology .....................................................3
   3. Audio Levels ....................................................3
   4. Signaling (Setup) Information ...................................5
   5. Considerations on Use ...........................................6
   6. Security Considerations .........................................6
   7. IANA Considerations .............................................7
   8. References ......................................................7
      8.1. Normative References .......................................7
      8.2. Informative References .....................................8

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.



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RFC 6464         Client-to-Mixer Audio Level Indication    December 2011


   The header extension in this document is different than, but
   complementary with, the one defined in [RFC6465], 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 with which it is associated.
   This information is carried in an RTP header extension element as
   defined by "A General Mechanism for RTP Header Extensions" [RFC5285].

   The payload of the audio level header extension element can be
   encoded using either the one-byte or two-byte header defined in
   [RFC5285].  Figures 1 and 2 show sample audio level encodings with
   each of these header formats.

                    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: Sample Audio Level Encoding Using the
                          One-Byte Header Format


      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)    |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

              Figure 2: Sample Audio Level Encoding Using the
                          Two-Byte Header Format

   Note that, as indicated in [RFC5285], the 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, the length field
   takes the value of 1.




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   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 Figures 1 and 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], 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 [ITU.G711].)

   The audio level for digital silence -- for a muted audio source, for
   example -- 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 with which it is associated, 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,
   Appendix A of [RFC6465] provides a sample Java implementation of an
   audio level calculator that helps obtain such values from raw linear
   Pulse Code Modulation (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.






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   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 [RFC3389], 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
   default is "vad=on".

   An example attribute line in the Session Description Protocol (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.








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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 achieving this, e.g., by measuring peak audio levels,
   by average audio levels during speech, or by measuring background
   audio levels (average audio 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, or falsely exaggerating its reported levels,
   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.  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 malicious.)





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   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 [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 [SRTP-ENCR-HDR], or a lower-level security and
   authentication mechanism such as IPsec [RFC4301].

7.  IANA Considerations

   This document defines a new extension URI in 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 6464

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.










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RFC 6464         Client-to-Mixer Audio Level Indication    December 2011


8.2.  Informative References

   [ITU.G711] International Telecommunication 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.

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

   [RFC6465]  Ivov, E., Ed., Marocco, E., Ed., and J. Lennox,
              "A Real-time Transport Protocol (RTP) Header Extension for
              Mixer-to-Client Audio Level Indication", RFC 6465,
              December 2011.

   [SRTP-ENCR-HDR]
              Lennox, J., "Encryption of Header Extensions in the Secure
              Real-Time Transport Protocol (SRTP)", Work in Progress,
              October 2011.

   [SRTP-VBR-AUDIO]
              Perkins, C. and JM. Valin, "Guidelines for the use of
              Variable Bit Rate Audio with Secure RTP", Work
              in Progress, July 2011.





















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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 Italia
   Via G. Reiss Romoli, 274
   Turin  10148
   Italy

   EMail: enrico.marocco@telecomitalia.it
























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