[asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge

Robert McGilvray rmcgilvr at sscinc.com
Wed Jun 29 10:24:29 CDT 2016

"timing test" does similar, it just doesn't do the automatic calculation. Confbridge normally operates at a mixing interval of 20ms, which is 50 ticks per second. That would be what you would want to test.
If you don't get 50 per second then that means ConfBridge will not provide a steady source of media to each participant and it will be up to each remote jitterbuffer to handle the delayed traffic. Enough of it and stuff goes wonky. You could also see this on a packet capture. That would determine if it's timing related or not.


Thanks Joshua. We're talking about pretty long gaps in the audio, probably around 10-15 seconds which is quite a bit of missed ticks at 20ms sampling. I was poking around the timing code trying to get a better understanding of things and found that Asterisk uses timerfd_create with CLOCK_MONOTONIC as the clock. The man page states CLOCK_MONOTONIC is affected by incremental adjustments to the time made by things like NTP.

I may be completely off track here but would something like vmtools that tries to correct the clock skew (caused by VMware) be causing some issues here? Meaning that if asterisk calls timerfd_create but then the time is adjusted could that throw off the timing of the descriptor?


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