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 > > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
