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Versions: 00 01 02 03 04 draft-ietf-avtext-mixer-to-client-audio-level

Network Working Group                                            E. Ivov
Internet-Draft                                          SIP Communicator
Intended status: Informational                                E. Marocco
Expires: July 20, 2011                                    Telecom Italia
                                                               J. Lennox
                                                             Vidyo, Inc.
                                                        January 16, 2011


  A Real-Time Transport Protocol (RTP) Extension Header for Mixer-to-
                     client Audio Level Indication
                         draft-ivov-avt-slic-04

Abstract

   This document describes a mechanism for RTP-level mixers in audio
   conferences to deliver information about the audio level of the
   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 July 20, 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  . . . . . . . . . . . . . . . . . . .  9
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  9
   9.  Open Issues  . . . . . . . . . . . . . . . . . . . . . . . . . 10
   10. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   11. Appendix: Design choices . . . . . . . . . . . . . . . . . . . 10
     11.1.  SIP event package for conference state  . . . . . . . . . 10
     11.2.  The RTP Control Protocol (RTCP) . . . . . . . . . . . . . 11
     11.3.  Encoding levels in the payload  . . . . . . . . . . . . . 11
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
     12.1.  Normative References  . . . . . . . . . . . . . . . . . . 12
     12.2.  Informative References  . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13


























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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 an RTP client 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 is based on the "General Mechanism for RTP Header
   Extensions" [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).

   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.  A value of 15 is not
   allowed by this specification and it MUST NOT be used as the RTP
   header can carry a maximum of 15 CSRC IDs.  The maximum value allowed
   is therefore 14 indicating a following sequence of 15 audio level
   values.

   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

   Audio level indicators are encoded in the same manner as audio noise
   level in the RTP Payload Comfort Noise specification [RFC3389] and
   audio level in the RTP Extension Header for Client-to-mixer Audio
   Level Notification [I-D.lennox-avt-rtp-audio-level-exthdr]
   specification.  The magnitude of the audio level is packed into the
   least significant bits of one audio-level byte with the most
   significant bit unused and always set to 0 as shown below in
   Figure 3.











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


                      Figure 3: Audio Level Encoding

   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.)


6.  Signaling Information

   The URI for declaring the audio level header extension in an SDP
   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 SHOULD always include the direction parameter in
   the "extmap" attribute setting it to "recvonly".  Conference focus
   entities with mixing capabilities MAY omit the direction or set it to
   "sendrecv" in SDP offers.  Such entities SHOULD set it to "sendonly"
   in SDP answers to offers with a "recvonly" parameter and to



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   "sendrecv" when answering other "sendrecv" offers.

   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.

     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























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     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 flwo of audio level information.

                                 Figure 5


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 may 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 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



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   to the following data:

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


9.  Open Issues

   At the time of writing of this document the authors have no clear
   view on how and if the following list of issues should be address
   here:
   1.  Audio levels in video streams.  This specification allows use of
       audio level values in "silent" audio streams that don't otherwise
       carry any payload thus allowing their delivery within systems
       where the various focus/mixer components communicate with each
       other as conference participants.  The same train of thought may
       very well justify audio level transport in video streams.
   2.  It has been suggested to reference ITU P.56 [ITU.P56.1993] for
       level measurement.  This needs to be investigated.


10.  Acknowledgments

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

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


11.  Appendix: Design choices

   During discussions on the subject of audio levels the decision to
   transport audio levels in RTP packets, rather than another protocol
   was questioned several times which is why the authors find it worth
   explaining here.  The following subsections describe alternative
   mechanisms for delivering audio levels and the reasons why authors
   decided not to use them.

11.1.  SIP event package for conference state

   RFC 4575 [RFC4575] defines a conference event package for tightly
   coupled conferences using the Session Initiation Protocol (SIP)
   events framework.  It allows for the delivery of various conference
   related details such as conference descriptions, participant count
   and identity.  The document also provides a way of indicating who the



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   speakers are at any given moment by specifying a mechanism for
   mapping conference participants to RTP SSRC/CSRC identifiers.  All
   these details are dispatched in an asynchronous manner using the SIP
   events framework, or, in other words, through NOTIFY SIP requests
   following an initial SUBSCRIBE from a participant.

   Contrary to "plain" active speaker infomation, where significant
   changes only occur once every several seconds, audio level in human
   speech is obviously a very time sensitive characteristic which would
   require frequent updates (i.e. approximately once every 50-100 ms).
   In order for the update of the user interface to appear "natural" to
   the user, audio level information would probably have to be delivered
   for every one or two RTP packets.  Using RFC 4575 [RFC4575] or SIP in
   general for this would generate traffic on the (often low-bandwidth)
   signalling path comparable to, if not exceeding, the media itself.
   It may also prove relatively hard for client developers to
   synchronize the information they receive from SIP messages with the
   one they obtain from the media flows.

   It is probably also worth mentioning that the use of RFC 4575
   [RFC4575] for such a feature would make the mechanism incompatible
   with non-SIP signaling protocols like, for example, XMPP [RFC3920]
   and its Jingle extensions.

11.2.  The RTP Control Protocol (RTCP)

   Similar to using SIP, delivering audio levels through RTCP would
   cause bandidth and synchronization issues.  Furthermore the RTP
   specification [RFC3550] explicitly recommends that the fraction of
   the session bandwidth added for RTCP be fixed at 5% which could not
   be sufficient for the transport of audio level indicators.

11.3.  Encoding levels in the payload

   Given the content specific nature of audio levels, it has been
   suggested that audio level information be encoded and transmitted as
   part of the payload.  While this is indeed a feasible approach,
   implementing it would require a substantial effort.  In order to
   implement support for such a feature, client developers would need to
   explicitly handle it in all individual codec modules of their
   application.  Compared to RTP extensions, the mechanism would
   therefore represent a substantial additional effort without offering
   any meaningful advantages.


12.  References





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12.1.  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.

12.2.  Informative References

   [I-D.lennox-avt-rtp-audio-level-exthdr]
              Lennox, J., Ivov, E., and E. Marocco, "A Real-Time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication",
              draft-lennox-avt-rtp-audio-level-exthdr-02 (work in
              progress), July 2010.

   [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.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3920]  Saint-Andre, P., Ed., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 3920, October 2004.

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



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              February 2006.

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


Authors' Addresses

   Emil Ivov
   SIP Communicator
   Strasbourg  67000
   France

   Email: emcho@sip-communicator.org


   Enrico Marocco
   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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