[Asterisk-Users] Dynamically Choose Codec for Bandwidth Management

[rsenykoff at harrislogic.com]

<snip>
rsenykoff at harrislogic.com wrote:
> Is there any way to set Asterisk to choose what codec to allow for a new 

> call based on current usage?
I think there is a way. Since I'm not in the stage yet to configure my 
extensions.conf on that deep level I found some clues.

http://www.voip-info.org/wiki-Asterisk+variables
${SIP_CODEC}: Used to set the SIP codec for a call

Probably if you make the call go thru an extension which checks current 
bandwidth consumption via an external program. (Something AGI) You could 
make the call jump to an low/normal/high bandwidth setting by set the 
SIP_CODEC for the to be used codec. With a bit of magic you probably can 
  check the amount of free G729 licences too.


Greetings,

Stefan de Konink

ps. The idea is neat... I'm definately going to try to work out some code.
</snip>

I think this would be a great way to provide a high level of service 
without needing to always up the connection. It would just be a nice way 
to handle those "peak" times (big conference call for example).

Let me know if you make any progress. I'll take a look at that variable. 
I'm thinking that rather than check against bandwidth, just provide a 
variable that can be configured to define the maximum number of ulaw 
calls. If that number is currently maxed, then any new calls get g729. 
But, once a ulaw drops off, there is then one available slot, so the next 
call initiated would get ulaw.

-Ron
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