We are KareXpert Technologies a HealthCloud Platform where a doctor can make video call with their patients. We have our sip client(android), webrtc(webapp) and kamailio as sip server. We need an expertise for debugging and to give solution to make sip call work.
Problem is webrtcToSip call freezing at web side.
I have worked with VoiP on my previous job as support engineer (2 years long), I could help you with fix your issue. I have bunch of resolved as fixed tasks concerned WebRTC2SIP, android dialers, etc.