[asterisk-users] Codec Selection in asterisk

[Pavel Jezek]

you can try this patch,
0004825: [patch][post 1.4] New codec negotiation algorithm
http://bugs.digium.com/view.php?id=4825

I'm think, this is one of the most wanted feature,
but unfortunately will not be in asterisk 1.4 and we must wait for 1.6 
to be officially supported feature :'(
PJ




Vicky wrote:
> I have around 20-30 softphones behind NAT  .. My sip.conf has nat=yes and
> they all are able to register and make calls with no problem . My voip
> carrier supports gsm as well as ilbc .. Server takes calls from sip 
> phones ,
> does call recording in between and forwards to voip carrier . My 
> problem is
> that half of my softphones use ilbc and  rest use gsm and my provider
> supports both gsm as well as ilbc .  Now when i put allow=gsm&ilbc in my
> voip carrier's extension then it uses gsm ( first preference ) to send 
> calls
> but half of my softphones use ilbc so asterisk does codec transcoding in
> between using lot of cpu ..  how ever my carrier does support ilbc 
> tooo but
> when i put allow=ilbc&gsm then it uses ilbc again and does codec 
> transcoding
> from gsm to ilbc for rest of softphones . How can i make asterisk to be
> smart in choosing codec .. and use ilbc to voip carrier if softphone is
> using ilbc or use gsm when softphone is using gsm ( but still should 
> do call
> recording in between ) .. I am using freepbx for most of configuration
> btw... Any suggestions ?
>
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