Asterisk PJSIP WebRTC

Hi, I will try that, thank you !

Hello, i have a other question about webrtc communication (again and again), I’m calling a webrtc client with a sip phone, and when I answer the call, I have a delay of 4/5 s to start the conversation.

I found you article about dlls_autogenerate_cert but in my case it seems to not fix the problem. Do you have an idea why ? Is it Ice server ?

Whats happening on the CLI during this time? keep your verbosity, and debug high (say 7), and check whats happening. Also you can check the RTP debug too.