[asterisk-users] Asterisk 11.24.1 garbled audio

[Matthew Jordan]

On Thu, Nov 10, 2016 at 4:00 PM, Jerry Geis <jerry.geis at gmail.com> wrote:

> I found dahdi_test...
>
>  dahdi_test
> Opened pseudo dahdi interface, measuring accuracy...
> 99.999% 99.904% 99.974% 99.814% 98.070% 97.850% 99.985% 99.887%
> 99.708% 99.899% 99.805% 99.708% 99.902% 100.000% 99.949% 99.883%
> 99.891% 99.906% 99.784% 99.719% 99.827% 99.903%
> --- Results after 22 passes ---
> Best: 100.000% -- Worst: 97.850% -- Average: 99.698465%
> Cummulative Accuracy (not per pass): 99.991
>
> seems like low numbers and not even running audio at this time.
>
> I'm thinking with the PRI card removed there is no reliable timing source.
>
> How do I get ConfBridge to have a reliable timing source?
>
>
Information on timing sources can be found here:

https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces

As noted on that page, ConfBridge can use any timing interface Asterisk
provides, and is not limited to the DAHDI timing interface. Generally,
timerfd is a good timing interface.

That aside, I would try to rule out external issues with the garbled audio
before changing the timing source. Things like:
 - Analysis of the RTP traffic (along with potential jitter)
 - CPU utilization with an active conference (95% idle doesn't mean that
some core isn't pegged)
 - Any potential transcoding issues or codec issues

Matt

-- 
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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