Intermittent One Way Audio, AT&T Support Says Due To High Jitter, Not sure how to fix

You need to correlate the incoming and outgoing media, in the same direction, and also look for logging that corresponds to an incident.

If the problem only arises for remote sources, the task processor overloads may be an unrelated issue, although you should try to get rid of them. Are you using any Asterisk API directly, or using a database on a separate machine?

The database is running local on the FreePBX server.

We do have a custom dashboard we created in PHP that uses Asterisk AMI that allows us to see what phones are ringing / on the phone, put phones on DND, etc.

If that generates an AMI event backlog I believe it can cause a task processor overload, although I think that is a different issue from your audio one.

Meanwhile this is what AT&T is telling me with the call samples I sent them that have the issue today. " We have found two calls with good RTP stats with no lost or out-of-order packets, but the audio is missing, thus it seems the issue occurs before the audio is handed to the router. The next step is to get them analyzed by our networking team for further analysis."

So depending what they come back with, it may be an AT&T issue after all. I’ll continue to reply to this thread regardless of the outcome. Thanks for all the reply’s and suggestions everyone has given me here so far.

AT&T ended up making routing changes which appears to have resolved the issues we were having.

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