Network Working Group                                         A. Sollaud
Internet-Draft                                            France Telecom
Expires: May 21, 2006                                  November 17, 2005


  RTP payload format for the future scalable and wideband extension of
                           G.729 audio codec
               draft-sollaud-avt-rtp-g729-scal-wb-ext-02

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Copyright Notice

   Copyright (C) The Internet Society (2005).

Abstract

   This document specifies a real-time transport protocol (RTP) payload
   format to be used for the future scalable and wideband extension of
   the International Telecommunication Union (ITU-T) G.729 audio codec.
   A media type registration is included for this payload format.







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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Background . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  RTP header usage . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Payload format . . . . . . . . . . . . . . . . . . . . . . . .  4
     4.1.  Payload structure  . . . . . . . . . . . . . . . . . . . .  5
     4.2.  Payload Header: MBS field  . . . . . . . . . . . . . . . .  5
     4.3.  Payload Header: FT field . . . . . . . . . . . . . . . . .  6
     4.4.  Audio data . . . . . . . . . . . . . . . . . . . . . . . .  7
   5.  Payload format parameters  . . . . . . . . . . . . . . . . . .  7
     5.1.  Media type registration  . . . . . . . . . . . . . . . . .  7
     5.2.  Mapping to SDP parameters  . . . . . . . . . . . . . . . .  9
     5.3.  Offer-answer model considerations  . . . . . . . . . . . .  9
   6.  Security considerations  . . . . . . . . . . . . . . . . . . . 10
   7.  IANA considerations  . . . . . . . . . . . . . . . . . . . . . 11
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
     8.1.  Normative references . . . . . . . . . . . . . . . . . . . 11
     8.2.  Informative references . . . . . . . . . . . . . . . . . . 11
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 12
   Intellectual Property and Copyright Statements . . . . . . . . . . 13






























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1.  Introduction

   The International Telecommunication Union (ITU-T) is working on a
   scalable and wideband extension of its recommendation G.729 [7].
   This future audio codec will be called G.729EV in the following text.
   This document specifies the payload format for packetization of
   G.729EV encoded audio signals into the real-time transport protocol
   (RTP).

   The payload format itself and the handling of variable bit rate are
   described in Section 4.  A media type registration and the details
   for the use of G.729EV with SDP are given in Section 5.

   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 [1].


2.  Background

   G.729EV is mainly designed to be used as a speech codec, but it can
   be used for music at the highest bit rates.  The sampling frequency
   is 16000 Hz and the frame size is 20 ms.

   This G.729-based codec produces an embedded bitstream providing an
   improved narrow band quality [300, 3400 Hz] at 12 kbps, and an
   enhanced and gracefully improving wideband quality [50, 7000 Hz] from
   14 kbps to 32 kbps, by steps of 2 kbps.  At 8 kbps it generates a
   G.729 bitstream.

   It has been mainly designed for packetized wideband voice
   applications (Voice over IP or ATM, Telephony over IP, private
   networks...) and particularly for those requiring scalable bandwidth,
   enhanced quality above G.729, and easy integration into existing
   infrastructures.

   G.729EV is also designed to cope with other services like high
   quality audio/video conferencing, archival, messaging, etc.

   For all those applications, the scalability feature allows to tune
   the bit rate versus quality trade-off, possibly in a dynamic way
   during a session, taking into account service requirements and
   network transport constraints.

   G.729EV produces frames that are said embedded because they are
   composed of embedded layers.  The first layer is called the core
   layer and is bitstream compatible with the ITU-T G.729 with annex B
   coder.  Upper layers are added while bit rate increases, to improve



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   quality and enlarge audio bandwidth from narrowband to wideband.  As
   a result, a received frame can be decoded at its original bit rate or
   at any lower bit rate corresponding to lower layers which are
   embedded.  Only the core layer is mandatory to decode understandable
   speech, upper layers provide quality enhancement and wideband
   enlargement.

   Audio codecs often support voice activity detection (VAD) and comfort
   noise generation (CNG).  During silence periods, the coder may
   significantly decrease the transmitted bit rate by sending only
   comfort noise parameters in special small frames called silence
   insertion descriptors (SID).  The receiver's decoder will generate
   comfort noise according to the SID information.  This operation of
   sending low bit rate comfort noise parameters during silence periods
   is usually called discontinuous transmission (DTX).

   G.729EV will be first released without support for DTX.  Anyway, this
   functionality is planned and will be defined in a separate annex
   later.  Thus this specification provides DTX signalling, even if the
   size of a SID frame is not yet standardized.


3.  RTP header usage

   The format of the RTP header is specified in RFC 3550 [2].  This
   payload format uses the fields of the header in a manner consistent
   with that specification.

   The RTP timestamp clock frequency is the same as the sampling
   frequency, that is 16 kHz.  So the timestamp unit is in samples.

   The duration of one frame is 20 ms, corresponding to 320 samples per
   frame.  Thus the timestamp is increased by 320 for each consecutive
   frame.

   The M bit should be set as specified in the applicable RTP profile,
   for example, RFC 3551 [3].

   The assignment of an RTP payload type for this packet format is
   outside the scope of the document, and will not be specified here.
   It is expected that the RTP profile under which this payload format
   is being used will assign a payload type for this codec or specify
   that the payload type is to be bound dynamically (see Section 5.2).


4.  Payload format





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4.1.  Payload structure

   The complete payload consists of a payload header of 1 octet,
   followed by audio data representing one or more consecutive frames at
   the same bit rate.

      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
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |  MBS  |   FT  |                                               |
     +-+-+-+-+-+-+-+-+                                               +
     :                one ore more frames at the same bit rate       :
     :                                                               :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

4.2.  Payload Header: MBS field

   MBS (4 bits): maximum bit rate supported.  Indicates a maximum bit
   rate to the encoder at the site of the receiver of this payload.  The
   value of the MBS field is set according to the following table:

                         +-------+--------------+
                         |  MBS  | max bit rate |
                         +-------+--------------+
                         |   0   |    8 kbps    |
                         |   1   |    12 kbps   |
                         |   2   |    14 kbps   |
                         |   3   |    16 kbps   |
                         |   4   |    18 kbps   |
                         |   5   |    20 kbps   |
                         |   6   |    22 kbps   |
                         |   7   |    24 kbps   |
                         |   8   |    26 kbps   |
                         |   9   |    28 kbps   |
                         |   10  |    30 kbps   |
                         |   11  |    32 kbps   |
                         | 12-14 |  (reserved)  |
                         |   15  |    NO_MBS    |
                         +-------+--------------+

   The MBS is used to tell the other party the maximum bit rate one can
   receive.  The encoder MUST follow the received MBS.  It MUST NOT send
   frames at a bit rate higher than the received MBS.  Thanks to the
   embedded property of the coding scheme, note that it can send frames
   at the MBS rate or any lower rate.  As long as it does not exceed the
   MBS, it can change its bit rate at any time without previous notice.

   The MBS received is valid until the next MBS is received, i.e. a



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   newly received MBS value overrides the previous one.

   If a payload with an invalid MBS value is received, the MBS MUST be
   ignored.

   Note that the MBS is a codec bit rate, the actual network bit rate is
   higher and depends on the overhead of the underlying protocols.

   The MBS field MUST be set to 15 for packets sent to a multicast
   group.

   The MBS field MUST be set to 15 in all packets when the actual MBS
   value is sent through non-RTP means.  This is out of the scope of
   this specification.

4.3.  Payload Header: FT field

   FT (4 bits): Frame type of the frame(s) in this packet, as per the
   following table:

                  +-------+---------------+------------+
                  |   FT  | encoding rate | frame size |
                  +-------+---------------+------------+
                  |   0   |     8 kbps    |  20 octets |
                  |   1   |    12 kbps    |  30 octets |
                  |   2   |    14 kbps    |  35 octets |
                  |   3   |    16 kbps    |  40 octets |
                  |   4   |    18 kbps    |  45 octets |
                  |   5   |    20 kbps    |  50 octets |
                  |   6   |    22 kbps    |  55 octets |
                  |   7   |    24 kbps    |  60 octets |
                  |   8   |    26 kbps    |  65 octets |
                  |   9   |    28 kbps    |  70 octets |
                  |   10  |    30 kbps    |  75 octets |
                  |   11  |    32 kbps    |  80 octets |
                  | 12-14 |   (reserved)  |            |
                  |   15  |    NO_DATA    |      0     |
                  +-------+---------------+------------+

   The FT value 15 (NO_DATA) indicates that there is no audio data in
   the payload.  This MAY be used to update the MBS value when there is
   no audio frame to transmit.  The payload will then be reduced to the
   payload header.

   If a payload with an invalid FT value is received, the whole payload
   MUST be ignored.





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4.4.  Audio data

   Audio data of a payload contains one or more consecutive audio frames
   at the same bit rate.  The audio frames are packed in order of time,
   that is the older first.

   The actual number of frame is easy to infer from the size of the
   audio data part:

      nb_frames = (size_of_audio_data) / (size_of_one_frame).

   This is compatible with DTX, with the restriction that the SID frame
   MUST be at the end of the payload (it is consistent with the payload
   format of G.729 described in section 4.5.6 of RFC 3551 [3]).  Since
   the SID frame is much smaller than any other frame, it will not
   hinder the calculation of the number of frames at the receiver side
   and can be easily detected.  Actually the presence of a SID frame
   will be inferred by the result of the above division not being an
   integer.

   Note that if FT=15, there will be no audio frame in the payload.


5.  Payload format parameters

   This section defines the parameters that may be used to configure
   optional features in the G.729EV RTP transmission.

   The parameters are defined here as part of the media subtype
   registration for the G.729EV codec.  A mapping of the parameters into
   the Session Description Protocol (SDP) [4] is also provided for those
   applications that use SDP.  In control protocols that do not use MIME
   or SDP, the media type parameters must be mapped to the appropriate
   format used with that control protocol.

5.1.  Media type registration

   This registration is done using the template defined in [8] and
   following RFC 3555 [9]

   Type name: audio

   Subtype name: G729EV

   Required parameters: none

   Optional parameters:




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   dtx: indicates that discontinuous transmission (DTX) is used or
      preferred.  DTX means voice activity detection and non
      transmission of silent frames.  Permissible values are 0 and 1. 0
      means no DTX. 0 is implied if this parameter is omitted.  The
      first version of G.729EV will not support DTX.

   mbs: indicates an initial value of MBS, that is the current maximum
      codec bit rate supported as a receiver.  Permissible values are
      between 0 and 11 (see table in Section 4.2 of RFC XXXX).  The
      maximum MBS, that is 11, is implied if this parameter is omitted.
      Note that this parameter will be dynamically updated by the MBS
      field of the RTP packets sent, it is not an absolute value for the
      session.  The goal is to announce this value, prior to the sending
      of any packet, to avoid the remote sender to exceed the MBS at the
      beginning of the session.

   ptime: the recommended length of time in milliseconds represented by
      the media in a packet.  See RFC 2327 [4].

   maxptime: the maximum length of time in milliseconds which can be
      encapsulated in a packet.

   Encoding considerations: This media type is framed and contains
   binary data.

   Security considerations: See Section 6 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: Audio and video conferencing
   tools.

   Additional information: none

   Person & email address to contact for further information: Aurelien
   Sollaud, aurelien.sollaud@francetelecom.com

   Intended usage: COMMON

   Restrictions on usage: This media type depends on RTP framing, and
   hence is only defined for transfer via RTP [2].

   Author/Change controller: IETF Audio/Video Transport working group
   delegated from the IESG





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5.2.  Mapping to SDP parameters

   The information carried in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [4], which is commonly used to describe RTP sessions.  When SDP is
   used to specify sessions employing the G.729EV codec, the mapping is
   as follows:

   o  The media type ("audio") goes in SDP "m=" as the media name.

   o  The media subtype ("G729EV") goes in SDP "a=rtpmap" as the
      encoding name.  The RTP clock rate in "a=rtpmap" MUST be 16000 for
      G.729EV.

   o  The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
      "a=maxptime" attributes, respectively.

   o  Any remaining parameters go in the SDP "a=fmtp" attribute by
      copying them directly from the media type string as a semicolon
      separated list of parameter=value pairs.

   Some example SDP session descriptions utilizing G.729EV encodings
   follow.

   Example 1: default parameters

      m=audio 53146 RTP/AVP 98
      a=rtpmap:98 G729EV/16000

   Example 2: recommended packet duration of 40 ms (=2 frames), DTX off,
   and initial MBS set to 26 kbps

      m=audio 51258 RTP/AVP 99
      a=rtpmap:99 G729EV/16000
      a=fmtp:99 dtx=0; mbs=8
      a=ptime:40

5.3.  Offer-answer model considerations

   The following considerations apply when using SDP offer-answer
   procedures to negotiate the use of G.729EV payload in RTP:

   o  Since G.729EV is an extension of G.729, the offerer SHOULD
      announce G.729 support in its "m=audio" line, with G.729EV
      preferred.  This will allow interoperability with both G.729EV and
      G.729-only capable parties.

      Below is an example of such an offer:



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         m=audio 55954 RTP/AVP 98 18
         a=rtpmap:98 G729EV/16000
         a=rtpmap:18 G729/8000

      If the answerer supports G.729EV, it will keep the payload type 98
      in its answer and the conversation will be done using G.729EV.
      Else, if the answerer supports only G.729, it will leave only the
      payload type 18 in its answer and the conversation will be done
      using G.729 (the payload format for G.729 is defined in RFC 3551
      [3]).

   o  The "dtx" parameter concerns both sending and receiving, so both
      sides of a bi-directional session MUST use the same "dtx" value.
      If one party indicates it does not support DTX, DTX must be
      deactivated both ways.

   o  The "mbs" parameter is not symmetric.  Values in the offer and the
      answer are independent and take into account local constraints.
      Anyway, one party MUST NOT start sending frames at a bit rate
      higher than the "mbs" of the other party.

   o  The parameters "ptime" and "maxptime" will in most cases not
      affect interoperability.  The SDP offer-answer handling of the
      "ptime" parameter is described in RFC 3264 [5].  The "maxptime"
      parameter MUST be handled in the same way.


6.  Security considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in the
   RTP specification [2] and any appropriate profile (for example, RFC
   3551 [3]).

   As this format transports encoded speech/audio, the main security
   issues include confidentiality and authentication of the speech/audio
   itself.  The payload format itself does not have any built-in
   security mechanisms.  Confidentiality of the media streams is
   achieved by encryption, therefore external mechanisms, such as SRTP
   [6], MAY be used for that purpose.

   This payload format and the G.729EV encoding do not exhibit any
   significant non-uniformity in the receiver-end computational load and
   thus in unlikely to pose a denial-of-service threat due to the
   receipt of pathological datagrams.






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7.  IANA considerations

   It is requested that one new media subtype (audio/G729EV) is
   registered by IANA, see Section 5.1.


8.  References

8.1.  Normative references

   [1]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", BCP 14, RFC 2119, March 1997.

   [2]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
        "RTP: A Transport Protocol for Real-Time Applications", STD 64,
        RFC 3550, July 2003.

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

   [4]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", RFC 2327, April 1998.

   [5]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        Session Description Protocol (SDP)", RFC 3264, June 2002.

   [6]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
        Norrman, "The Secure Real-time Transport Protocol (SRTP)",
        RFC 3711, March 2004.

8.2.  Informative references

   [7]  International Telecommunications Union, "Coding of speech at 8
        kbit/s using conjugate-structure algebraic-code-excited linear-
        prediction (CS-ACELP)", ITU-T Recommendation G.729, March 1996.

   [8]  Freed, N. and J. Klensin, "Media Type Specifications and
        Registration Procedures", draft-freed-media-type-reg-05 (work in
        progress), August 2005.

   [9]  Casner, S. and P. Hoschka, "MIME Type Registration of RTP
        Payload Formats", RFC 3555, July 2003.









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Author's Address

   Aurelien Sollaud
   France Telecom
   2 avenue Pierre Marzin
   Lannion Cedex  22307
   France

   Phone: +33 2 96 05 15 06
   Email: aurelien.sollaud@francetelecom.com









































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