[payload] registration of iSac media subtype

"Roni Even" <ron.even.tlv@gmail.com> Thu, 13 December 2012 06:48 UTC

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From: Roni Even <ron.even.tlv@gmail.com>
To: media-types@iana.org
Date: Thu, 13 Dec 2012 08:45:57 +0200
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Cc: draft-ietf-avt-rtp-isac@tools.ietf.org, payload@ietf.org
Subject: [payload] registration of iSac media subtype
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Hi,

The iSac draft is currently in WGLC in the Payload WG
http://tools.ietf.org/html/draft-ietf-avt-rtp-isac-02 

the media type registration is in section 4 and also included below.

Please review the registration.

 

Thanks

Roni Even 

Payload WG co-chair

 

   

Media type name:   audio

 

   Media subtype:   isac

 

   Required parameters:  None

 

   Optional parameters:

 

      *  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).

 

      *  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:  RFC XXXX

 

   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 Google.  The

      codec operates on 30 or 60 ms speech frames at a sampling rate

      clock of 16 kHz or 32 kHz.

 

   Person to contact for further information:

      Tina le Grand [tlegrand@google.com]

 

   Restrictions on usage:

      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.

 

   Note to the RFC Editor / IANA: Please replace "RFC XXXX" above with

   the number of this RFC when published, and remove this note.