[asterisk-users] Codec Selection in asterisk

[Tim Panton]

>
>
> 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 ?
> >

>
> 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
>
>
>

On 7 Dec 2006, at 21:29, Vicky wrote:
> I am still on asterisk 1.2 branch svn ( afraid of word beta on  
> server :( ) . I will try out that patch.

Alternatively try setting
${SIP_CODEC}
before you place the call to your provider.

I'd love to hear if it works.



Tim Panton

www.mexuar.net
www.westhawk.co.uk/