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

Joshua Colp jcolp at digium.com
Wed Jun 29 04:45:02 CDT 2016

Robert McGilvray wrote:


> Before I rip apart the environment and rebuild on physical I’d like to
> try and confirm that hypothesis. In the past this was a simple matter of
> running dahdi_test which would report the accuracy. I’m not sure how to
> interpret the results of “timing test” in the Asterisk CLI. If I
> increase the number of ticks per second the results are erratic while
> under load. I’m using the timerfd module in Asterisk with a 1000HZ tick
> kernel and high res timers enabled. I’ve tried both hpet and tsc as
> system clock sources, both exhibit the same breaks in audio. It sounds
> like someone presses the mute button in the middle of a sentence.

"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.

Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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