[Asterisk-Dev] Preventing Asterisk from codec conversion when reinviting

[Andreas Sikkema]

Hi,

I'm new to Asterisk and SIP, but have experience with developing 
VoIP apps in general and H.323 apps in particular. 

I would like to fix a problem we're having with Asterisk 
performing codec conversion in the following situation:

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 (so Asterisk is performing codec conversion)

In this scenario we'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.

We would like Asterisk to use the codec settings of EP2 
while sending a reinvite to EP1. Currently Asterisk uses 
it's own codec settings as defined in sip.conf. Since we 
are using Asterisk  together with SER, our endpoints are 
not defined in sip.conf so we cannot put information 
about them in it. 

Can someone tell me where to start developing? Which 
files, functions?

TIA.

-- 
Andreas Sikkema                Rits tele.com
Scheepmakersstraat 11      3011 VH Rotterdam
t: +31 (0)10 2245544    f: +31 (0)10 2245540