Bad sound recordings pjsip/websocket/webrtc

Hello Guys,

I have a problem that the calling are normal, but the recordings come with some cuts in the speech.

I used Wireshark to see a possible latency, jitter, delta issues or something on the network and so far I haven’t found anything specific.

Not only that, but I don’t have any hardware issues like CPU,I/O etc. I’ve already changed the hardware.

I’m using mixmonitor to record the calls in an extension WAV with Asterisk version 16.19.0, PJ SIP with WebRTC (codec opus).

Someone could give me a light?


Can any kind soul help me with this situation?

Has anyone ever been in a similar situation?

You could try simplifying things, such as eliminating the use of opus. That would be a codec that realistically not many are using, let alone in combination with recordings.

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I did what you said jcolp and the codec opus wasn’t the real problem and changed for ulaw got worse, I discovered that in the network communication had a jitter problem, I used the HOMER SERVER to capture the jitter error but wasn’t very clear, then I changed some agents for softphone and by wireshark was clear to understand the exact point of the problem on the call audio.

I would like to know a better way to capture messages and audio between pjsip and wireshark in this case, I tried to used sngrep with hep than i can see all messages but when i open in wireshark can’t listen the rtp audio.

Thanks for help

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