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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