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

[Andres]

Manoj_Rajkarnikar wrote:

>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 
>  
>
you need to use the following parameter in your sip definitions (not 
sure if Trixbox will take it though)
disallow=all
allow=g729:30    ;30 is the frame size in ms

Andres
http://www.neuroredes.com

>asterisk code?? Please suggest.
>
>Thanks a lot.
>
>Manoj
>
>  
>