Good day, i would like to ask and advice about the best way to debug a “no audio issues”.
Here is the typical situation (sorry the sip packets are doubled)
Incoming call from provider proxy.210 to Janus.186, rtp stream is coming correctly to Janus port 45802 and there is no rtp stream from Janus.
My clients says that he got the call, but no audio sometimes
My problem is that the WEBRTC is encrypted and i dont know how to get traces to pinpoint the issue, basically everything behind the web is a black box, and the only way i have is to debug each call with a remote desk software, but as it is random it’s not simple
Is it possible to collect some kind of data on regular basis for all calls to understand what is being send and received by the customer so i can find the problem ? (For example if the stream was sent/received, ports)
Without this information i can only supposse and never know for sure.
My setup is a multistream version Janus <> sip plugin <> provider
Janus is on dedicated server on public IP.
I have a dedicated STUN / TURN server my customers can activate to force the traffic thru it if the firewall block all but port 3478.
Thank you for your advices.
Greg