Network Working Group                                         A. Sollaud
Internet-Draft                                            France Telecom
Updates: 4749 (if approved)                             February 8, 2008
Intended status: Standards Track
Expires: August 11, 2008


             G.729.1 RTP Payload Format update: DTX support
                  draft-ietf-avt-rfc4749-dtx-update-00

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Copyright Notice

   Copyright (C) The IETF Trust (2008).

Abstract

   This document updates the Real-time Transport Protocol (RTP) payload
   format to be used for the International Telecommunication Union
   (ITU-T) Recommendation G.729.1 audio codec.  It adds Discontinuous
   Transmission (DTX) support to the RFC 4749 specification, in a
   backward-compatible way.  An updated 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  . . . . . . . . . . . . . . . . . . . . . . . 3
   4.  Payload Format  . . . . . . . . . . . . . . . . . . . . . . . . 4
   5.  Payload Format Parameters . . . . . . . . . . . . . . . . . . . 5
     5.1.  Media Type Registration Update  . . . . . . . . . . . . . . 5
     5.2.  Mapping to SDP Parameters . . . . . . . . . . . . . . . . . 5
       5.2.1.  DTX Offer-Answer Model Considerations . . . . . . . . . 6
       5.2.2.  DTX Declarative SDP Considerations  . . . . . . . . . . 6
   6.  Congestion Control  . . . . . . . . . . . . . . . . . . . . . . 6
   7.  Security Considerations . . . . . . . . . . . . . . . . . . . . 6
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . . . 6
     9.1.  Normative References  . . . . . . . . . . . . . . . . . . . 6
     9.2.  Informative References  . . . . . . . . . . . . . . . . . . 7
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . . . 7
   Intellectual Property and Copyright Statements  . . . . . . . . . . 8
































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1.  Introduction

   The International Telecommunication Union (ITU-T) Recommendation
   G.729.1 [1] is a scalable and wideband extension of the
   Recommendation G.729 [6] audio codec.  RFC 4749 [3] specifies the
   payload format for packetization of G.729.1 encoded audio signals
   into the Real-time Transport Protocol (RTP) [4].

   The ITU-T is about to release an Appendix, to add Discontinuous
   Transmission (DTX) support to G.729.1.  This document updates the RTP
   payload format to allow usage of this Appendix.

   Only changes or additions to RFC 4749 [3] will be described in the
   following sections.

   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 [2].


2.  Background

   G.729.1 supports Discontinuous Transmission (DTX), a.k.a. silence
   suppression.  It means that the coder includes a Voice Activity
   Detection (VAD) algorithm, to determine if an audio frame contains
   silence or actual audio.  During silence periods, the coder may
   significantly decrease the transmitted bit rate by sending a small
   frame called Silence Insertion Descriptor (SID), and then stop
   transmission.  The receiver's decoder will generate comfort noise
   according to the parameters contained in the SID.

   G.729.1 SID has an embedded structure.  The core SID is the same as
   the legacy G.729 SID [7].  A first enhancement layer adds some
   parameters for narrowband comfort noise, while a second enhancement
   layer adds wideband information.  In the current state of the
   specification, SID can be 2, 3, or 6 octets long.


3.  RTP Header Usage

   The fields of the RTP header must be used as described in RFC 4749,
   except for the Marker (M) bit.

   If DTX is used, the first packet of a talkspurt, that is, the first
   packet after a silence period during which packets have not been
   transmitted contiguously, SHOULD be distinguished by setting the M
   bit in the RTP data header to one.  The M bit in all other packets
   MUST be set to zero.  The beginning of a talkspurt MAY be used to



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   adjust the playout delay to reflect changing network delays.

   If DTX is not used, the M bit MUST be set to zero in all packets.


4.  Payload Format

   The payload format is the same as in RFC 4749, with the option to add
   a SID at the end.

   So the complete payload consists of a payload header of 1 octet (MBS
   and FT fields), followed by zero or more consecutive audio frames at
   the same bit rate, followed by zero or one SID.

      Note that it is consistent with the payload format of G.729
      described in section 4.5.6 of RFC 3551 [8].

   To be able to transport a SID alone, that is, without actual audio
   frames, we assign the FT value 14 to SID.  The actual SID size (2, 3,
   or 6 octets) is inferred from the payload size.

   When a SID is appended to actual audio frames, the FT value remains
   the one describing the encoding rate of the audio frames.  Since the
   SID is much smaller than any other frame, it can be easily detected
   at the receiver side, and it will not hinder the calculation of the
   number of frames.

   The full FT table is given for convenience:

               +-------+---------------+-------------------+
               |   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     |
               | 12-13 |   (reserved)  |         -         |
               |   14  |      SID      | 2, 3, or 6 octets |
               |   15  |    NO_DATA    |         0         |
               +-------+---------------+-------------------+



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   DTX has no impact on the MBS definition and use.


5.  Payload Format Parameters

   Parameters defined in RFC 4749 are not modified.  We add a new
   optional parameter to configure DTX.

5.1.  Media Type Registration Update

   We add a new optional parameter to the audio/G7291 media subtype:

   dtx:  indicates that discontinuous transmission (DTX) is used or
      preferred.  Permissible values are 0 and 1. 0 means no DTX. 1
      means DTX support, as described in Appendix ZZZ of ITU-T
      Recommendation G.729.1. 0 is implied if this parameter is omitted.

   When DTX is turned off, the RTP payload MUST NOT contain SID, and the
   FT value 14 MUST NOT be used.

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)
   [5], which is commonly used to describe RTP sessions.

   The mapping described in RFC 4749 remains unchanged.

   The "dtx" parameter goes in the SDP "a=fmtp" attribute.

   Some example partial SDP session descriptions utilizing G.729.1
   encodings follow.

   Example 1: default parameters (DTX off)

      m=audio 57586 RTP/AVP 96
      a=rtpmap:96 G7291/16000

   Example 2: recommended packet duration of 40 ms (=2 frames), maximum
   bit rate is 20 kbps, DTX supported and preferred.

      m=audio 49987 RTP/AVP 97
      a=rtpmap:97 G7291/16000
      a=fmtp:97 maxbitrate=20000; dtx=1
      a=ptime:40






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5.2.1.  DTX Offer-Answer Model Considerations

   The offer-answer model considerations of RFC 4749 fully apply.  In
   this section we only define the management of the "dtx" parameter.

   The "dtx" parameter concerns both sending and receiving, so both
   sides of a bi-directional session MUST have the same DTX setting.  If
   one party indicates it does not support DTX, DTX must be deactivated
   both ways.  In other words, DTX is actually activated if, and only
   if, "dtx=1" in the offer and in the answer.

   A special rule apply for multicast: the "dtx" parameter becomes
   declarative and MUST NOT be negotiated.  This parameter is fixed, and
   a participant MUST use the configuration that is provided for the
   session.

5.2.2.  DTX Declarative SDP Considerations

   The "dtx" parameter is declarative and provides the parameter that
   SHALL be used when receiving and/or sending the configured stream.


6.  Congestion Control

   The congestion control considerations of RFC 4749 apply.

   The use of DTX can help congestion control by reducing the number of
   transmitted RTP packets and the average bandwidth of audio streams.


7.  Security Considerations

   DTX introduces no new security issue.


8.  IANA Considerations

   It is requested that one new parameter for media subtype (audio/
   G7291) is registered by IANA, see Section 5.1.


9.  References

9.1.  Normative References

   [1]  International Telecommunications Union, "G.729 based Embedded
        Variable bit-rate coder: An 8-32 kbit/s scalable wideband coder
        bitstream interoperable with G.729", ITU-T Recommendation



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        G.729.1, May 2006.

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

   [3]  Sollaud, A., "RTP Payload Format for the G.729.1 Audio Codec",
        RFC 4749, October 2006.

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

   [5]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
        Description Protocol", RFC 4566, July 2006.

9.2.  Informative References

   [6]  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.

   [7]  International Telecommunications Union, "A silence compression
        scheme for G.729 optimized for terminals conforming to
        Recommendation V.70", ITU-T Recommendation G.729 Annex B,
        October 1996.

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


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@orange-ftgroup.com











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