[Asterisk-Users] SIP reinvite code negotiation

[Andreas Sikkema]

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