I have caller poor audio quality first seconds of a call

Hello there,
In asterisk, i have caller poor audio quality first seconds of a call. This only happens when the caller is at a certain location. I only use ala&ulaw as a codec. How can I find a solution to this problem?
My system and using technologies is like this:
Centos 7, asterisk 16.20. PJSIP / WEBRTC

Thank You,
Onur

Have you tried the opus codec with Asterisk WebRTC ? It can adapt better to changing network conditions.

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