I’m using only SIP channels, ztdummy is installed and working correctly. MusicOnHold application sounds fine. I configured 32 buffers in meetme.conf and restarted asterisk.
My test scenario is a dual Xeon 3.0Ghz machine with no other activity, 2 SIP channels in a MeetMe conference. One participant is playing music (generated by a non-asterisk enterprise phone system MoH, no problems there), and the other participant is just listening to the music.
I experience multi-second audio drop-outs and stuttery audio for a second or two, many times per minute in the conference. I’m using Asterisk 1.4.12.1, but I’ve experienced this in all releases of Asterisk 1.2 I’ve used over the last 18 months.
When I just play music on hold application directly generated from the same system as the MeetMe conference, it is perfectly clear.
So the problem is not an IP packet network problem, or a busy-machine / hardware problem, or a timing issue with ztdummy… check these zttest results during a time when there were multi-second audio drop-outs in a meetme:
Opened pseudo zap interface, measuring accuracy…
99.975586% 99.975586% 99.975586% 99.975586% 99.987793% 99.975586% 99.975586%
99.975586% 99.975586% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586% 99.987793%
99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%
99.975586% 99.987793% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%
99.975586% 99.975586% 99.975586% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586%
— Results after 39 passes —
Best: 99.987793 – Worst: 99.975586 – Average: 99.977464
Seems like something specifically with app_meetme
Have other folks experienced this same issue?