Issues with quality voice

I am a user of the latest version of freepbx, 15, with asterisk 16.
For some time now, I have noticed that the calls stop hearing as when the mobile loses coverage per second. I have two teams and in both cases the same. I thought it could be a router problem and it would change without satisfactory result. I thought it could be a problem for the internet operator and what it is with a different one without satisfactory results. Does anyone know what could be happening and how to fix it? It has always worked perfectly. a hug.
Issues of any asterisk update?

We’d need to see some logs. You may also need to turn on SIP debug and watch the traffic.

It sounds to me like you are losing packets, which would be an indicator of network congestion. It’s not going to last long, but if someone is sucking up all of the data headroom in your inbound/outbound pipe, it can happen. It only takes a couple of lost packets to ruin the audio.

Make sure your OS, FreePBX modules and Asterisk are up to date.
Audio being based on RTP protocol, you need to set a QoS on your network. But I think that’s ok here.

Check if the QoS with still set on your network.
Other thing, You may have a network loop somewhere and overload the bandwidth.
If that worked and oneday no, maybe the issue is on your network.
I don’t say that your issue come from your network, It’s just to add a clue on your issue.

Also, sometimes you could have a device (Workstation…etc) which mess your audio.
Otherwise, try to downgrade Asterisk 16 and see what happens.

Just an idea like that

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.