[AVT] RE: Dynamic selection of Codec??

["Ingemar Johansson S \(LU/EAB\)" <[e-mail]>]

Hi

As I forgot to change the subject line in my last email I need to repost it, sorry for the double posting

There are a few speech/audio codecs available that support dynamic changes in bitrate during a session (without the need for a SIP reINVITE). From the AMR codec family one can chose from

AMR:     
  Narrowband speech codec (8kHz sample rate).
  Bitrates 4.75-12.2kbps
  (aka the codec used in GSM, UMTS)
  http://en.wikipedia.org/wiki/AMR-WB
AMR-WB:  Widedband speech codec (16kHz sample rate). 
  Bitrates 6.60-23.85kbps
  http://en.wikipedia.org/wiki/AMR-WB
Payload format in RFC3267 and soon updated by draft-ietf-avt-rtp-amr-bis

AMR-WB+: 
  Speech and audio stereo/mono codec. 
  Audio bandwidth varies between 6.4kHz and 20kHz depending on
  bitrate (5.2-48kbps), the sample rate remains the same though due to
  internal "dynamically adjustable" resampling. Worth notice is also
  that it is possible to even switch between stereo and mono
  almost seamlessly during a session
  http://en.wikipedia.org/wiki/AMR-WB%2B
Payload format in RFC4352

C-code for the codecs mendioned above can be downloaded from http://www.3gpp.org/ftp/Specs/html-info/26-series.htm
The floating point c-code are specs 26.104, 26.204 and 26.304
Also: The bitrate switching and the associated signaling is specified in 3GPP TS26.114 available from the same website, this does actually adress the dynamic codec rate question described below.

Regards
Ingemar

> -----Original Message-----
> From: [e-mail] [mailto:[e-mail]] 
> Sent: den 27 mars 2007 16:41
> To: [e-mail]
> Subject: avt Digest, Vol 35, Issue 30
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> Today's Topics:
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>    1. RE: Dynamic selection of Codec?? ([e-mail])
>    2. RE: Dynamic selection of Codec?? ([e-mail])
>    3. RE: Dynamic selection of Codec?? (Even, Roni)
> 
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> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Tue, 27 Mar 2007 20:05:08 +0530
> From: <[e-mail]>
> Subject: RE: [AVT] Dynamic selection of Codec??
> To: "'Taddei, Herve'" <[e-mail]>,	<[e-mail]>
> Message-ID: <[e-mail]>
> Content-Type: text/plain;	charset="iso-8859-1"
> 
> Hi, 
>  
> Is this only limited for wideband / scalable codecs?? If 
> that's the case, then how can we it be implemented if I don't 
> have scalable codecs?
> 
> Thanks & regards,
> Vamsi. 
> 
> -----Original Message-----
> From: Taddei, Herve [mailto:[e-mail]]
> Sent: Tuesday, March 27, 2007 7:03 PM
> To: [e-mail]; [e-mail]
> Subject: AW: [AVT] Dynamic selection of Codec??
> 
> Hi,
> 
> You can use a scalable codec for example G.729.1 standardized 
> in the ITU-T and decide on the bitrate you want to say based 
> on the network conditions.
> You don't need any codec re-negotiation when you change of 
> bitrate. RTP payload format is RFC 4749.
> 
> Cheers
> 
> Hervé
> 
> -----Ursprüngliche Nachricht-----
> Von: [e-mail] [mailto:[e-mail]]
> Gesendet: Dienstag, 27. März 2007 15:14
> An: [e-mail]
> Betreff: [AVT] Dynamic selection of Codec??
> 
> Hello, 
>    Can we select codec dynamically in the mid session when a 
> SIP call is going on?  
> 
> Suppose in 10 min. of a SIP call, Initially my bandwidth is 
> excellent so I choose for quality codec rather than bandwidth 
> consumption codec, and after few minutes my bandwidth is very 
> less, so at that time, how can I manage my quality of call?  
> 

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