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Versions: (draft-ivov-avt-slic) 00 01 02 03 04 05 06 RFC 6465

Network Working Group                                       E. Ivov, Ed.
Internet-Draft                                                     Jitsi
Intended status: Informational                           E. Marocco, Ed.
Expires: November 10, 2011                                Telecom Italia
                                                               J. Lennox
                                                             Vidyo, Inc.
                                                             May 9, 2011


  A Real-Time Transport Protocol (RTP) Header Extension for Mixer-to-
                     Client Audio Level Indication
            draft-ietf-avtext-mixer-to-client-audio-level-02

Abstract

   This document describes a mechanism for RTP-level mixers in audio
   conferences to deliver information about the audio level of
   individual participants.  Such audio level indicators are transported
   in the same RTP packets as the audio data they pertain to.

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 November 10, 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



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   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  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Protocol Operation . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Header Format  . . . . . . . . . . . . . . . . . . . . . . . .  6
   5.  Audio level encoding . . . . . . . . . . . . . . . . . . . . .  6
   6.  Signaling Information  . . . . . . . . . . . . . . . . . . . .  7
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   9.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   10. Changes From Earlier Versions  . . . . . . . . . . . . . . . . 11
     10.1.  Changes From Draft -01  . . . . . . . . . . . . . . . . . 11
     10.2.  Changes From Draft -00  . . . . . . . . . . . . . . . . . 11
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
     11.1.  Normative References  . . . . . . . . . . . . . . . . . . 11
     11.2.  Informative References  . . . . . . . . . . . . . . . . . 11
   Appendix A.  Reference Implementation  . . . . . . . . . . . . . . 12
     A.1.   AudioLevelCalculator.java . . . . . . . . . . . . . . . . 13
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15


























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1.  Introduction

   The Framework for Conferencing with the Session Initiation Protocol
   (SIP) defined in RFC 4353 [RFC4353] presents an overall architecture
   for multi-party conferencing.  Among others, the framework borrows
   from RTP [RFC3550] and extends the concept of a mixer entity
   "responsible for combining the media streams that make up a
   conference, and generating one or more output streams that are
   delivered to recipients".  Every participant would hence receive, in
   a flat single stream, media originating from all the others.

   Using such centralized mixer-based architectures simplifies support
   for conference calls on the client side since they would hardly
   differ from one-to-one conversations.  However, the method also
   introduces a few limitations.  The flat nature of the streams that a
   mixer would output and send to participants makes it difficult for
   users to identify the original source of what they are hearing.

   Mechanisms that allow the mixer to send to participants cues on
   current speakers (e.g. the CSRC fields in RTP [RFC3550]) only work
   for speaking/silent binary indications.  There are, however, a number
   of use cases where one would require more detailed information.
   Possible examples include the presence of background chat/noise/
   music/typing, someone breathing noisily in their microphone, or other
   cases where identifying the source of the disturbance would make it
   easy to remove it (e.g. by sending a private IM to the concerned
   party asking them to mute their microphone).  A more advanced
   scenario could involve an intense discussion between multiple
   participants that the user does not personally know.  Audio level
   information would help better recognize the speakers by associating
   with them complex (but still human readable) characteristics like
   loudness and speed for example.

   One way of presenting such information in a user friendly manner
   would be for a conferencing client to attach audio level indicators
   to the corresponding participant related components in the user
   interface as displayed in Figure 1.














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                         ________________________
                        |                        |
                        |  00:42 |  Weekly Call  |
                        |________________________|
                        |                        |
                        |                        |
                        | Alice |======    | (S) |
                        |                        |
                        | Bob   |=         |     |
                        |                        |
                        | Carol |          | (M) |
                        |                        |
                        | Dave  |===       |     |
                        |                        |
                        |________________________|


     Figure 1: Displaying detailed speaker information to the user by
               including audio level for every participant.

   Implementing a user interface like the above requires analysis of the
   media sent from other participants.  In a conventional audio
   conference this is only possible for the mixer since all other
   conference participants are generally receiving a single, flat audio
   stream and have therefore no immediate way of determining individual
   audio levels.

   This document specifies an RTP extension header that allows such
   mixers to deliver audio level information to conference participants
   by including it directly in the RTP packets transporting the
   corresponding audio data.


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].


3.  Protocol Operation

   According to RFC 3550 [RFC3550] a mixer is expected to include in
   outgoing RTP packets a list of identifiers (CSRC IDs) indicating the
   sources that contributed to the resulting stream.  The presence of
   such CSRC IDs allows RTP clients to determine, in a binary way, the
   active speaker(s) in any given moment.  RTCP also provides a basic
   mechanism to map the CSRC IDs to user identities through the CNAME



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   field.  More advanced mechanisms, may exist depending on the
   signaling protocol used to establish and control a conference.  In
   the case of the Session Initiation Protocol [RFC3261] for example,
   the Event Package for Conference State [RFC4575] defines a <src-id>
   tag which binds CSRC IDs to media streams and SIP URIs.

   This document describes an RTP header extension that allows mixers to
   indicate the audio-level of every conference participant (CSRC) in
   addition to simply indicating their on/off status.  This new header
   extension uses "General Mechanism for RTP Header Extensions"
   described in [RFC5285].

   Each instance of this header contains a list of one-octet audio
   levels expressed in -dBov, with values from 0 to 127 representing 0
   to -127 dBov(see Section 4 and Section 5).  Appendix A provides a
   reference implementation indicating one way of obtaining such values
   from raw audio samples.

   Every audio level value pertains to the CSRC identifier located at
   the corresponding position in the CSRC list.  In other words, the
   first value would indicate the audio level of the conference
   participant represented by the first CSRC identifier in that packet
   and so forth.  The number and order of these values MUST therefore
   match the number and order of the CSRC IDs present in the same
   packet.

   When encoding audio level information, a mixer SHOULD include in a
   packet information that corresponds to the audio data being
   transported in that same packet.  It is important that these values
   follow the actual stream as closely as possible.  Therefore a mixer
   SHOULD also calculate the values after the original contributing
   stream has undergone possible processing such as level normalization,
   and noise reduction for example.

   Note that in some cases a mixer may be sending an RTP audio stream
   that only contains audio level information and no actual audio.
   Updating a (web) interface conference module may be one reason for
   this to happen.

   It may sometimes happen that a conference involves more than a single
   mixer.  In such cases each of the mixers MAY choose to relay the CSRC
   list and audio-level information they receive from peer mixers (as
   long as the total CSRC count remains below 16).  Given that the
   maximum audio level is not precisely defined by this specification,
   it is likely that in such situations average audio levels would be
   perceptibly different for the participants located behind the
   different mixers.




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4.  Header Format

   The audio level indicators are delivered to the receivers in-band
   using the "General Mechanism for RTP Header Extensions" [RFC5285].
   The payload of this extension is an ordered sequence of 8-bit audio
   level indicators encoded as per Section 5.


       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  |0|  level 1    |0|  level 2    |0|  level 3   ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

             Figure 2: Audio level indicators extension format

   The 4-bit len field is the number minus one of data bytes (i.e. audio
   level values) transported in this header extension element following
   the one-byte header.  Therefore, the value zero in this field
   indicates that one byte of data follows.  RFC 3550 [RFC3550] only
   allows RTP packets to carry a maximum of 15 CSRC IDs.  Given that
   audio levels directly refer to CSRC IDs, implementations MUST NOT
   include more than 15 audio level values.  The maximum value allowed
   in the len field is therefore 14.

   Note that use of the two-byte header defined in RFC 5285 [RFC5285]
   follows the same rules the only change being the length of the ID and
   len fields.


5.  Audio level encoding

   The audio level header extension only carries the level of the audio
   in the RTP payload of the packet it is associated with.  This
   information is carried in an RTP header extension element as defined
   by [RFC5285].

   The audio level is defined in the same manner as is audio noise level
   in the RTP Payload Comfort Noise specification [RFC3389].  The
   overall magnitude of the noise level 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.

   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 3.  The least significant bit of the audio level magnitude is



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   packed into the least significant bit of the byte.  The most
   significant bit of the byte is unused and always set to 0 as shown
   below in Figure 3.


                           0 1 2 3 4 5 6 7
                          +-+-+-+-+-+-+-+-+
                          |0|   level     |
                          +-+-+-+-+-+-+-+-+


                      Figure 3: Audio Level Encoding

   The two-byte header defined in RFC 5285 [RFC5285] may also be used.

   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.G.711], 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,
   this specification provides a sample Java implementation (Appendix A)
   of an audio level calculator that helps obtain such values from raw
   linear PCM audio samples.


6.  Signaling Information

   The URI for declaring the audio level header extension in an SDP



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   extmap attribute and mapping it to a local extension header
   identifier is "urn:ietf:params:rtp-hdrext:csrc-audio-level".  There
   is no additional setup information needed for this extension (i.e. no
   extensionattributes).

   An example attribute line in the SDP, for a conference might be:

           a=extmap:7 urn:ietf:params:rtp-hdrext:csrc-audio-level

   The above mapping will most often be provided per media stream (in
   the media-level section(s) of SDP, i.e., after an "m=" line) or
   globally if there is more than one stream containing audio level
   indicators in a session.

   Presence of the above attribute in the SDP description of a media
   stream indicates that some or all RTP packets in that stream would
   contain the audio level information RTP extension header.

   Conferencing clients that support audio level indicators and have no
   mixing capabilities would not be able to content for this audio level
   extension and would hence have to always include the direction
   parameter in the "extmap" attribute with a value of "recvonly".
   Conference focus entities with mixing capabilities can omit the
   direction or set it to "sendrecv" in SDP offers.  Such entities would
   need to set it to "sendonly" in SDP answers to offers with a
   "recvonly" parameter and to "sendrecv" when answering other
   "sendrecv" offers.

   This speicification does not define use of the audio level extensions
   in video streams.  Therefore, the extension defined in this document
   SHOULD NOT be advertised in anything but audio streams.

   The following Figure 4 and Figure 5 show two example offer/answer
   exchanges between a conferencing client and a focus, and between two
   conference focus entities.
















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     v=0
     o=alice 2890844526 2890844526 IN IP6 host.example.com
     c=IN IP6 host.example.com
     t=0 0
     m=audio 49170 RTP/AVP 0 4
     a=rtpmap:0 PCMU/8000
     a=rtpmap:4 G723/8000
     a=extmap:1/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level

     v=0
     i=A Seminar on the session description protocol
     o=conf-focus 2890844730 2890844730 IN IP6 focus.example.net
     c=IN IP6 focus.example.net
     t=0 0
     m=audio 52543 RTP/AVP 0
     a=rtpmap:0 PCMU/8000
     a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:csrc-audio-level

   A client-initiated example SDP offer/answer exchange negotiating an
   audio stream with one-way flow of of audio level information.

                                 Figure 4


     v=0
     i=Un seminaire sur le protocole de description des sessions
     o=fr-focus 2890844730 2890844730 IN IP6 focus.fr.example.net
     c=IN IP6 focus.fr.example.net
     t=0 0
     m=audio 49170 RTP/AVP 0
     a=rtpmap:0 PCMU/8000
     a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level

     v=0
     i=A Seminar on the session description protocol
     o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net
     c=IN IP6 focus.us.example.net
     t=0 0
     m=audio 52543 RTP/AVP 0
     a=rtpmap:0 PCMU/8000
     a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level

   An example SDP offer/answer exchange between two conference focus
   entities with mixing capabilities negotiating an audio stream with
   bidirectional flow of audio level information.

                                 Figure 5




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7.  Security Considerations

   1.  This document defines a means of attributing audio level to a
       particular participant in a conference.  An attacker may try to
       modify the content of RTP packets in a way that would make audio
       activity from one participant appear as coming from another.
   2.  Furthermore, the fact that audio level values would not be
       protected even in an SRTP session might be of concern in some
       cases where the activity of a particular participant in a
       conference is confidential.
   3.  Both of the above are concerns that stem from the design of the
       RTP protocol itself and they would probably also apply when using
       CSRC identifiers the way they were specified in RFC 3550
       [RFC3550].  It is therefore important that according to the needs
       of a particular scenario, implementors and deployers consider use
       of header extension encryption
       [I-D.lennox-avtcore-srtp-encrypted-header-ext] or a lower level
       security and authentication mechanism.


8.  IANA Considerations

   This document defines a new extension URI that, if approved, would
   need to be added to the RTP Compact Header Extensions sub-registry of
   the Real-Time Transport Protocol (RTP) Parameters registry, according
   to the following data:

           Extension URI: urn:ietf:params:rtp-hdrext:csrc-audio-level
           Description:   Mixer-to-client audio level indicators
           Contact:       emcho@jitsi.org
           Reference:     RFC XXXX

   Note to the RFC-Editor: please replace "RFC XXXX" by the number of
   this RFC.


9.  Acknowledgments

   Lyubomir Marinov contributed level measurement and rendering code.

   Roni Even, Keith Drage, Ingemar Johansson, Michael Ramalho and
   several others provided helpful feedback over the dispatch mailing
   list.

   Jitsi's participation in this specification is funded by the NLnet
   Foundation.





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10.  Changes From Earlier Versions

   Note to the RFC-Editor: please remove this section prior to
   publication as an RFC.

10.1.  Changes From Draft -01

   o  Removed code related the AudioLevelRenderer from "APPENDIX A.
      Reference Implementation" as it was considered an implementation
      matter by the working group.
   o  Modified the AudioLevelCalculator in "APPENDIX A. Reference
      Implementation" to take overload as a parameter.
   o  Clarified non-use of audio levels in video streams
   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  The Open Issues section has been removed as all issues that were
      in there are now resolved or clarified.
   o  Editorial changes for consistency with
      [I-D.ietf-avtext-client-to-mixer-audio-level].

10.2.  Changes From Draft -00

   o  Added code for sound pressure calculation and measurement in
      "APPENDIX A. Reference Implementation".
   o  Changed affiliation for Emil Ivov.
   o  Removed "Appendix: Design choices".


11.  References

11.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 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.

11.2.  Informative References

   [I-D.ietf-avtext-client-to-mixer-audio-level]
              Lennox, J., Ivov, E., and E. Marocco, "A Real-Time



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              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication",
              draft-ietf-avtext-client-to-mixer-audio-level-01 (work in
              progress), March 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.

   [ITU.G.711]
              International Telecommunications Union, "Pulse Code
              Modulation (PCM) of Voice Frequencies", ITU-
              T Recommendation G.711, November 1988.

   [ITU.P56.1993]
              International Telecommunications Union, "Objective
              Measurement of Active Speech Level", ITU-T Recommendation
              P.56, March 1988.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              February 2006.

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
              Initiation Protocol (SIP) Event Package for Conference
              State", RFC 4575, August 2006.


Appendix A.  Reference Implementation

   This appendix contains Java code for a reference implementation of
   the level calculation and rendering methods.The code is not normative
   and by no means the only possible implementation.  Its purpose is to
   help implementors add audio level support to mixers and clients.

   The Java code contains an AudioLevelCalculator class that calculates
   the sound pressure level of a signal with specific samples.  It can
   be used in mixers to generate values suitable for the level extension



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   headers.

   The implementation is provided in Java but does not rely on any of
   the language specific and can be easily ported to another.

A.1.  AudioLevelCalculator.java


/**
 * Calculates the audio level of specific samples of a singal based on
 * sound pressure level.
 */
public class AudioLevelCalculator
{

    /**
     * Calculates the sound pressure level of a signal with specific
     * <tt>samples</tt>.
     *
     * @param samples the samples of the signal to calculate the sound
     * pressure level of. The samples are specified as an <tt>int</tt>
     * array starting at <tt>offset</tt>, extending <tt>length</tt>
     * number of elements and each <tt>int</tt> element in the specified
     * range representing a sample of the signal to calculate the sound
     * pressure level of. Though a sample is provided in the form of an
     * <tt>int</tt> value, the sample size in bits is determined by the
     * caller via <tt>overload</tt>.
     *
     * @param offset the offset in <tt>samples</tt> at which the samples
     * start
     *
     * @param length the length of the signal specified in
     * <tt>samples<tt> starting at <tt>offset</tt>
     *
     * @param overload the overload (point) of <tt>signal</tt>.
     * For example, <tt>overload</tt> may be {@link Byte#MAX_VALUE}
     * for 8-bit signed samples or {@link Short#MAX_VALUE} for
     * 16-bit signed samples.
     *
     * @return the sound pressure level of the specified signal
     */
    public static int calculateSoundPressureLevel(
        int[] samples, int offset, int length,
        int overload)
    {
        /*
         * Calcuate the root mean square of the signal i.e. the
         * effective sound pressure.



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         */
        double rms = 0;

        for (; offset < length; offset++)
        {
            double sample = samples[offset];

            sample /= overload;
            rms += sample * sample;
        }
        rms = (length == 0) ? 0 : Math.sqrt(rms / length);

        /*
         * The sound pressure level is a logarithmic measure of the
         * effectivesound pressure of a sound relative to a reference
         * value and is measured in decibels.
         */
        double db;

        /*
         * The minimum sound pressure level which matches the maximum
         * of the sound meter.
         */
        final double MIN_SOUND_PRESSURE_LEVEL = 0;
        /*
         * The maximum sound pressure level which matches the maximum
         * of the sound meter.
         */
        final double MAX_SOUND_PRESSURE_LEVEL
            = 127 /* HUMAN TINNITUS (RINGING IN THE EARS) BEGINS */;

        if (rms > 0)
        {
            /*
             * The commonly used "zero" reference sound pressure in air
             * is 20 uPa RMS, which is usually considered the threshold
             * of human hearing.
             */
            final double REF_SOUND_PRESSURE = 0.00002;

            db = 20 * Math.log10(rms / REF_SOUND_PRESSURE);

            /*
             * Ensure that the calculated level is within the minimum
             * and maximum sound pressure level.
             */
            if (db < MIN_SOUND_PRESSURE_LEVEL)
                db = MIN_SOUND_PRESSURE_LEVEL;



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Internet-Draft   Mixer-to-Client Audio Level Indication         May 2011


            else if (db > MAX_SOUND_PRESSURE_LEVEL)
                db = MAX_SOUND_PRESSURE_LEVEL;
        }
        else
        {
            db = MIN_SOUND_PRESSURE_LEVEL;
        }

        return (int) db;
    }
}


   AudioLevelCalculator.java


Authors' Addresses

   Emil Ivov (editor)
   Jitsi
   Strasbourg  67000
   France

   Email: emcho@jitsi.org


   Enrico Marocco (editor)
   Telecom Italia
   Via G. Reiss Romoli, 274
   Turin  10148
   Italy

   Email: enrico.marocco@telecomitalia.it


   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack,  NJ  07601
   US

   Email: jonathan@vidyo.com








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