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Versions: 00 01 02 draft-ietf-avt-rtp-isac

AVT                                                         T. le Grand
Internet Draft                                      Global IP Solutions
Intended status: Standards Track                               P. Jones
Expires: April 2010                                               Cisco
                                                               P. Huart
                                                                  Cisco
                                                       October 15, 2009


                   RTP Payload Format for the iSAC Codec
                       draft-legrand-rtp-isac-02.txt


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Abstract

   iSAC is a proprietary wideband speech and audio codec developed by
   Global IP Solutions, suitable for use in Voice over IP applications.
   This document describes the payload format for iSAC generated bit
   streams within a Real-Time Protocol (RTP) packet.  Also included here
   are the necessary details for the use of iSAC with the Session
   Description Protocol (SDP).

Conventions used in this document

   In examples, "C:" and "S:" indicate lines sent by the client and
   server respectively.

   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].

Table of Contents

   1. Introduction...................................................2
   2. iSAC Codec Description.........................................3
   3. RTP Payload Format.............................................4
      3.1. iSAC Payload Format.......................................4
      3.2. Payload Header............................................5
      3.3. Encoded Speech Data.......................................5
      3.4. Multiple iSAC frames in an RTP packet.....................6
   4. IANA Considerations............................................6
      4.1. Media Type registration of iSAC...........................6
   5. Mapping to SDP Parameters......................................8
      5.1. Example Initial Target Bit Rate...........................8
      5.2. Example Max Bit Rate......................................9
   6. Security Considerations........................................9
   7. Acknowledgments................................................9
   8. References.....................................................9
      8.1. Normative References......................................9
      8.2. Informative References...................................10
   Author's Addresses...............................................10

1. Introduction

   This document gives a general description of the iSAC wideband speech
   codec and specifies the iSAC payload format for usage in RTP packets.
   Also included here are the necessary details for the use of iSAC with
   the Session Description Protocol (SDP).




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2. iSAC Codec Description

   The iSAC codec is an adaptive wideband speech and audio codec that
   operates with short delay, making it suitable for high quality real
   time communication.  It is specially designed to deliver wideband
   speech quality in both low and medium bit rate applications.  It also
   handles non-speech audio well, such as music and background noise
   [5].

   The iSAC codec compresses speech frames of 16 kHz, 16-bit sampled
   input speech, each frame containing 30 or 60 ms of speech.

   The codec runs in one of two different modes called channel-adaptive
   mode and channel-independent mode.  In both modes iSAC is aiming at a
   target bit rate, which is neither the average nor the maximum bit
   rate that will be reach by iSAC, but corresponds to the average bit
   rate during peaks in speech activity.  The bit rate will sometimes
   exceed the target bit rate, but most of the time will be below.  The
   average bit rate obtained is on average about a factor of 1.4 times
   lower than the target bit rate.

   In channel-adaptive mode the target bit rate is adapted to give a bit
   rate corresponding to the available bandwidth on the channel.  The
   available bandwidth is constantly estimated at the receiving iSAC and
   signaled in-band in the iSAC bit stream.  Even at dial-up modem data
   rates (including IP, UDP, and RTP overhead) iSAC delivers high
   quality by automatically adjusting transmission rates to give the
   best possible listening experience over the available bandwidth.  The
   default initial target bit rate is 20000 bits per second in channel-
   adaptive mode.

   In channel-independent mode a target bit rate has to be provided to
   iSAC prior to encoding.

   After encoding the speech signal the iSAC coder uses lossless coding
   to further reduce the size of each packet, and hence the total bit
   rate used.

   The adaptation and the lossless coding described above both result in
   a variation of packet size, depending both of the nature of speech
   and the available bandwidth.  Therefore the iSAC codec operates at
   transmission rates from about 10 kbps to about 32 kbps.

   The main characteristics can be summarized as follows:

   o  Wideband, 16 kHz, speech and audio codec



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   o  Variable bit rate, which depends on the input signal

   o  Adaptive rate with two modes: channel-adaptive or channel-
      independent mode

   o  Bit rate range from around 10 kbps to 32 kbps

   o  Operates on 30 or 60 ms of speech

3. RTP Payload Format

   The iSAC codec uses a sampling rate clock of 16 kHz, so the RTP
   timestamp MUST be in units of 1/16000 of a second.

   The RTP payload for iSAC has the format shown in Figure 1.  No
   additional header fields specific to this payload format are
   required.  For RTP based transportation of iSAC encoded audio, the
   standard RTP header [2] is followed by one payload data block.

     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
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |                      RTP Header                               |
     +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
     |                                                               |
     +                    iSAC Payload Block                         +
     |                                                               |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                   Figure 1: RTP packet format for iSAC

3.1. iSAC Payload Format

   The iSAC payload block consists of a payload header and one or two
   encoded 30 ms speech frames.  The iSAC payload is generated in the
   following manner:

   o  Parameters representing one or two 30 ms frames of speech data are
      determined by the encoder.  The parameters are quantized to
      generate encoded data corresponding to the one or two speech
      frames.  The length of the encoded data is variable and depends on
      the signal characteristics and the target bit rate.

   o  The payload header is generated (described in Section 3.2) and
      added before the encoded parameter data for the speech frame(s).





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   o  Lossless coding is applied to the complete iSAC payload block,
      including payload header, to generate a compressed payload.  The
      length depends on the length of the data generated to represent
      the speech and the effectiveness of the lossless coding.

   No part of the payload header or the encoded speech data can be
   retrieved without partly or fully decoding the packet.

   The following figure shows an iSAC payload block containing 60 ms of
   encoded speech data:

     +--------+--------+--------+--------+--------+--------+--------+
     |Payload |       30 ms Encoded      |     30 ms Encoded        |
     |Header  |         Speech Data      |       Speech Data        |
     +--------+--------+--------+--------+--------+--------+--------+

                     Figure 2: Payload format for iSAC

3.2. Payload Header

   The payload header holds information for the receiver about the
   available bandwidth (BEI), and the length of the speech data in the
   current payload (FL).  The header has the format defined in Figure 3.
   Note that the size of the header can vary due to the lossless
   encoding described in section 2 and in section 3.1.  Also note that
   the BEI is always estimated and transmitted, even if iSAC runs in
   channel-independent mode.

                               +-+-+-+-+-+-+
                               | BEI |  FL |
                               +-+-+-+-+-+-+

                         Figure 3: Payload Header

   o  BEI: Bandwidth Estimation Index.  The bandwidth estimate is
      quantized into one out of 24 values.  Valid values are 0 to 23.

   o  FL: The length of the speech data (Frame Length) present in the
      payload, given in number of speech samples.  Valid frame lengths
      are 480 (30 ms) and 960 (60 ms) samples.

3.3. Encoded Speech Data

   The iSAC encoded speech data consist of parameters representing one
   or two frames of 30 ms speech.  The length of the speech data is
   signaled in the header (in number of samples), and the length may



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   change at any time during a session.  In channel-adaptive mode the
   length is changed to best utilize the available bandwidth.

   The iSAC payload is padded to whole octets, and has a variable length
   depending on the input source signal, number of 30 ms speech frames,
   and target bit rate.

   The number of octets used to describe one frame of 30 ms speech
   typically varies from around 50 to around 120 octets.  For the case
   of 60 ms speech (two 30 ms speech frames), the number of octets
   varies from around 100 to around 240 octets.  The absolute maximum
   allowed payload length is 400 octets.  The user can choose to lower
   the maximum allowed payload length.  Minimum value is 100 octets.  It
   is possible for the user to choose a maximum bit rate instead of a
   maximum payload length.  The maximum payload length is then dependent
   on the length of the speech data represented in the payload (30 or 60
   ms).  Possible maximum rates are in the range of 32000 to 53400 bits
   per second.

   The sensitivity to bit errors is equal for all bits in the payload.

3.4. Multiple iSAC frames in an RTP packet

   More than one iSAC payload block MUST NOT be included in an RTP
   packet by a sender.

   Further, iSAC payload blocks MUST NOT be split between RTP packets.

4. IANA Considerations

   This document defines the iSAC media type.

4.1. Media Type registration of iSAC

   Media type name: audio

   Media subtype: isac

   Required parameters: None

   Optional parameters:








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   o  ibitrate: The parameter indicates the upper bound of the initial
      target bit rate the device would like to receive.  For channel-
      adaptive mode, the target bit rate may vary with time; for
      channel-independent mode, the target bit rate will remain at that
      level unless instructed otherwise.  An acceptable value for
      ibitrate is in the range of 20000 to 32000 (bits per second).

   o  maxbitrate: The parameter indicates the maximum bit rate the
      endpoint expects to receive.  The recipient of this parameter
      SHOULD NOT transmit at a higher bit rate.

   Encoding considerations:

      This media format is framed and binary.

   Security considerations:

      See section 6.

   Interoperability considerations: None

   Published specification:

   Applications which use this media type:

      This media type is suitable for use in numerous applications
      needing to transport encoded voice or other audio.  Some examples
      include Voice over IP, Streaming Media, Voice Messaging, and
      Conferencing.

   Additional information: None

   Intended usage: COMMON

   Other Information/General Comment:

      iSAC is a proprietary speech and audio codec owned by Global IP
      Solutions.  The codec operates on 30 or 60 ms speech frames at a
      sampling rate clock of 16 kHz.

   Person to contact for further information:

      Tina le Grand [tina.legrand@gipscorp.com]

   Restrictions on usage:




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      This media type depends on RTP framing, and hence is only defined
      for transfer via RTP [2].  Transport within other framing
      protocols is not defined at this time.

   Change controller:

      IETF Audio/Video Transport working group delegated from the IESG.

5. 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 iSAC codec, the mapping is as
   follows:

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

   o  The media subtype (payload format name) goes in SDP "a=rtpmap" as
      the encoding name.

   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.

   The optional parameter ibitrate MUST NOT be higher than the parameter
   maxbitrate.

   The iSAC parameters in an SDP offer are completely independent from
   those in the SDP answer.  For both ibitrate and maxbitrate it is
   legal for the answer to contain a value that is different than what
   is provided in an offer.  The parameter may be present in the answer,
   even if absent in the offer.

   When conveying information by SDP, the encoding name SHALL be "isac"
   (the same as the media subtype).

5.1. Example Initial Target Bit Rate

   The offer indicates that it wishes to receive a bitstream with an
   initial target rate of 20000 bits per second.  The remote party MAY
   change its initial target rate to the requested value.

      m=audio 10000 RTP/AVP 98
      a=rtpmap: 98 isac/16000
      a=fmtp:98 ibitrate=20000



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5.2. Example Max Bit Rate

   The offer indicates that it wishes to receive a bitstream with an
   initial target rate of 20000 bits per second, and a maximum bit rate
   of 45000 bits per second.  The remote party MAY change its initial
   target rate and SHOULD NOT transmit at a higher rate than 45000.

      m=audio 10000 RTP/AVP 98
      a=rtpmap: 98 isac/16000
      a=fmtp:98 ibitrate=20000;maxrate=45000

6. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in RFC
   3550 [2].

   As this format transports encoded speech, the main security issues
   include confidentiality and authentication of the speech itself.  The
   payload format itself does not have any built-in security mechanisms.
   External mechanisms, such as SRTP [3], MAY be used.

7. Acknowledgments

   This document was prepared using 2-Word-v2.0.template.dot.

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 Jacobson, V.,
         "RTP: A Transport Protocol for Real-Time Applications", STD 64,
         RFC 3550, July 2003.

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

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






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8.2. Informative References

   [5]   iSAC datasheet at Global IP Solutions website,
         http://www.gipscorp.com/files/english/datasheets/iSAC.pdf

Author's Addresses

   Tina le Grand
   Global IP Solutions
   Magnus Ladulasgatan 63B
   SE-118 27 Stockholm
   Sweden
   Email: tina.legrand@gipscorp.com

   Paul E. Jones
   Cisco Systems, Inc,
   7025 Kit Creek Rd.
   Research Triangle Park, NC 27709
   USA
   Tel: +1 919 476 2048
   Email: paulej@packetizer.com

   Pascal Huart
   Cisco Systems
   400, Avenue Roumanille
   Batiment T3
   06410 BIOT - SOPHIA ANTIPOLIS
   FRANCE
   Tel: +33 4 9723 2643
   Email: phuart@cisco.com



















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