[asterisk-users] g729 open source codec and sample size

[Manoj_Rajkarnikar]

Greetings.

I'm new to the asterisk & voip world and I'm currently trying out trixbox 
2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 
codec from site http://asterisk.hosting.lv/ and is working fine. question 
here is that this codec sends out a packet every 20ms. Though the speech 
quality is very good, I also like to try out 30ms sampling size to bring 
down the overhead payload and reduce bandwidth usage. I've searched for it 
for a couple days with no indication of how to do it. is it possible to 
change it. do i have to compile my own codec module.. or some patch to 
asterisk code?? Please suggest.

Thanks a lot.

Manoj

--