I have audio problems in conference with 15 users. Sometimes the problem occur also with less users too. I am using meetme for conferencing with counting option. The audio is robotic sometimes or cut off.
The server is connected with Avaya pbx and the signaling is sip.
The audio path is beetwen Asterisk and Avaya Media boards.
I made statistics with ethereal for rtps and I see max jitter sometimes around 10 ms.
The jitterbuffer in meetme is set for 32 and same in zapata.conf. Also this makes delays in the conversation. The cpu usage for asterisk is usually 2% with 20 users. I use ulaw for all users. I am using Asterisk BE 1.2 and Asterisk BE 2.2.1 and happening on both servers.
I would appreciate if somebody can tell me what could be the reason for the robotic voice and what settings should I use. Is there any other settings for jitter?
Thanks.