Hi,
We're routing SIP calls through Asterisk and we want to
be able to reinvite calls without Asterisk performing
codec conversion.
We've performed the following test:
Asterisk has license for G.729 installed
sip.conf
[general]
context=default
autocreatepeer=yes
disallow=all
allow=alaw
allow=g729
canreinvite=yes
nat=no
We have configured two endpoints:
EP1, preferred codec order aLaw, G.729
EP2, preferred codec order G.729
EP1 places call to EP2, we see two call legs:
EP1 to * is aLaw
* to EP2 is G.729
Is there a sip.conf parameter to disable codec conversion
when using reinvite?
If not, is it difficult / a lot of work to develop?
In this scenario you'd like to use G.729 for both call
legs with the media stream bypassing Asterisk. Thus
reducing the CPU load on the * machine and the need
for additional G.729 licenses.
Creating seperate configuration per user is not an
option, as we do not know which codecs users have.
Some may have both G.729 and aLaw, while other might
only have G.729 or aLaw.
--
Andreas Sikkema
