[asterisk-users] Codec Selection in asterisk

[Vicky]

I am still on asterisk 1.2 branch svn ( afraid of word beta on server :( ) .
I will try out that patch.
On 08/12/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
>
> 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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