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 -- _______________________________________________ -- 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
