path: root/doc/rfc/rfc6465.txt

Internet Engineering Task Force (IETF)                      E. Ivov, Ed.
Request for Comments: 6465                                         Jitsi
Category: Standards Track                                E. Marocco, Ed.
ISSN: 2070-1721                                           Telecom Italia
                                                               J. Lennox
                                                                   Vidyo
                                                           December 2011


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

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 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/rfc6465.

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.




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Table of Contents

   1. Introduction ....................................................2
   2. Terminology .....................................................4
   3. Protocol Operation ..............................................4
   4. Audio Levels ....................................................5
   5. Signaling Information ...........................................7
   6. Security Considerations .........................................9
   7. IANA Considerations ............................................10
   8. Acknowledgments ................................................10
   9. References .....................................................10
      9.1. Normative References ......................................10
      9.2. Informative References ....................................11
   Appendix A. Reference Implementation ..............................12
      A.1. AudioLevelCalculator.java .................................12

1.  Introduction

   "A Framework for Conferencing with the Session Initiation Protocol
   (SIP)" [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 contributing source (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.



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   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.  One possible example is displayed in Figure 1, where
   levels can help users determine that Alice is currently the active
   speaker, Carol is mute, and Bob and Dave are sending some background
   noise.

                         ________________________
                        |                        |
                        |  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 therefore have 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.

   The header extension in this document is different than, but
   complementary to, the one defined in [RFC6464], which defines a
   mechanism by which clients can indicate to audio mixers the levels of
   the audio in the packets they send.








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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.  The RTP Control Protocol
   (RTCP) also provides a basic mechanism to map the CSRC IDs to user
   identities through the CNAME field.  More advanced mechanisms can
   exist, depending on the signaling protocol used to establish and
   control a conference.  In the case of the Session Initiation Protocol
   [RFC3261], for example, "A Session Initiation Protocol (SIP) Event
   Package for Conference State" [RFC4575] defines a <src-id> tag that
   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 contributing conference participant
   (CSRC) in addition to simply indicating their on/off status.  This
   new header extension uses the general mechanism for RTP header
   extensions as 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 Figures 2 and 3).  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.



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   It can 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.

4.  Audio Levels

   The audio level header extension carries the level of the audio in
   the RTP 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 2 and 3 show sample audio level encodings with
   each of these header formats.

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

              Figure 2: 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=3     |0|   level 1   |0|   level 2   |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |0|   level 3   |    0 (pad)    |               ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

              Figure 3: Sample Audio Level Encoding Using the
                          Two-Byte Header Format

   In the case of the one-byte header 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.  In the case of the two-byte header format, the 8-bit
   len field contains the exact number of audio levels carried in the



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   extension.  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
   for the one-byte header format and 15 for the two-byte header format.

      Note: Audio levels in this document are 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.

   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 2 and 3.  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 unused and always set to 0.

   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.  That level is measured as
   a root mean square of all the samples in the measured range.

   To simplify implementation of the encoding procedures described here,
   this specification provides a sample Java implementation (see
   Appendix A) of an audio level calculator that helps obtain such
   values from raw linear Pulse Code Modulation (PCM) audio samples.






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5.  Signaling Information

   The URI for declaring the audio level header extension in a Session
   Description Protocol (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
   extension attributes).

   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 RTP packets in that stream, which contain the
   level extension defined in this document, will be carrying such an
   extension with an ID of 7.

   Conferencing clients that support audio level indicators and have no
   mixing capabilities would not be able to provide 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 specification only defines the use of the audio level extensions
   in audio streams.  They MUST NOT be advertised with other media
   types, such as video or text, for example.

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











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     SDP Offer:

       v=0
       o=alice 2890844526 2890844526 IN IP6 host.example.com
       s=-
       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

     SDP Answer:

       v=0
       i=A Seminar on the session description protocol
       o=conf-focus 2890844730 2890844730 IN IP6 focus.example.net
       s=-
       c=IN IP6 focus.example.net
       t=0 0
       m=audio 52544 RTP/AVP 0
       a=rtpmap:0 PCMU/8000
       a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:csrc-audio-level

      Figure 4: A Client-Initiated Example SDP Offer/Answer Exchange
             Negotiating an Audio Stream with One-Way Flow of
                          Audio Level Information
























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     SDP Offer:

       v=0
       i=Un seminaire sur le protocole de description des sessions
       o=fr-focus 2890844730 2890844730 IN IP6 focus.fr.example.net
       s=-
       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

     SDP Answer:

       v=0
       i=A Seminar on the session description protocol
       o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net
       s=-
       c=IN IP6 focus.us.example.net
       t=0 0
       m=audio 52544 RTP/AVP 0
       a=rtpmap:0 PCMU/8000
       a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level

   Figure 5: 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

6.  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 to be coming from another
       participant.

   2.  Furthermore, the fact that audio level values would not be
       protected even in a Secure Real-time Transport Protocol (SRTP)
       session [RFC3711] might be of concern in some cases where the
       activity of a particular participant in a conference is
       confidential.  Also, as discussed in [SRTP-VBR-AUDIO], an
       attacker might be able to infer information about the
       conversation, possibly with phoneme-level resolution.

   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 in the way specified in RFC 3550
       [RFC3550].  It is therefore important that, according to the



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       needs of a particular scenario, implementors and deployers
       consider the use of header extension encryption [SRTP-ENCR-HDR]
       or a lower-level security and authentication mechanism such as
       IPsec [RFC4301], for example.

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:csrc-audio-level
      Description:   Mixer-to-client audio level indicators
      Contact:       emcho@jitsi.org
      Reference:     RFC 6465

8.  Acknowledgments

   Lyubomir Marinov contributed level measurement and rendering code.

   Keith Drage, Roni Even, Miguel A. Garcia, John Elwell, Kevin P.
   Fleming, Ingemar Johansson, Michael Ramalho, Magnus Westerlund, and
   several others provided helpful feedback over the avt and avtext
   mailing lists.

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

9.  References

9.1.  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.










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9.2.  Informative References

   [ITU.G711] International Telecommunication Union, "Pulse Code
              Modulation (PCM) of Voice Frequencies",
              ITU-T Recommendation G.711, November 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.

   [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.

   [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, Ed., "A
              Session Initiation Protocol (SIP) Event Package for
              Conference State", RFC 4575, August 2006.

   [RFC6464]  Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer 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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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 is 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
   headers.

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

A.1.  AudioLevelCalculator.java

   <CODE BEGINS>

   /*
      Copyright (c) 2011 IETF Trust and the persons identified
      as authors of the code.  All rights reserved.

      Redistribution and use in source and binary forms, with
      or without modification, is permitted pursuant to, and subject
      to the license terms contained in, the Simplified BSD License
      set forth in Section 4.c of the IETF Trust's Legal Provisions
      Relating to IETF Documents (http://trustee.ietf.org/license-info).
   */

   /**
    * Calculates the audio level of specific samples of a signal
    * relative to overload.
    */
   public class AudioLevelCalculator
   {

       /**
        * Calculates the audio level of a signal with specific
        * <tt>samples</tt>.
        *
        * @param samples  the samples whose audio level we need to
        * calculate.  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 whose audio level we




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        * need to calculate.  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> can be {@link Byte#MAX_VALUE}
        * for 8-bit signed samples or {@link Short#MAX_VALUE} for
        * 16-bit signed samples.
        *
        * @return  the audio level of the specified signal.
        */
       public static int calculateAudioLevel(
           int[] samples, int offset, int length,
           int overload)
       {
           /*
            * Calculate the root mean square (RMS) of the signal.
            */
           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 audio level is a logarithmic measure of the
            * rms level of an audio sample relative to a reference
            * value and is measured in decibels.
            */
           double db;

           /*
            * The minimum audio level permitted.
            */
           final double MIN_AUDIO_LEVEL = -127;





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           /*
            * The maximum audio level permitted.
            */
           final double MAX_AUDIO_LEVEL = 0;

           if (rms > 0)
           {
               /*
                * The "zero" reference level is the overload level,
                * which corresponds to 1.0 in this calculation, because
                * the samples are normalized in calculating the RMS.
                */
               db = 20 * Math.log10(rms);

               /*
                * Ensure that the calculated level is within the minimum
                * and maximum range permitted.
                */
               if (db < MIN_AUDIO_LEVEL)
                   db = MIN_AUDIO_LEVEL;
               else if (db > MAX_AUDIO_LEVEL)
                   db = MAX_AUDIO_LEVEL;
           }
           else
           {
               db = MIN_AUDIO_LEVEL;
           }

           return (int)Math.round(db);
       }
   }

   <CODE ENDS>


















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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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