i am using asterisk 15 with webrtc and sipml5
i am facing an issue in asterisk that my sip users are getting legged.
and their calling getting hamper. it took long time to connect the call (in sipml5 phone showing call in progress) and user getting unreachable after some time.
please help me in that
Thanks in Advance
You haven’t provided enough information. What is the internet connection? What does packet captures show? Does the traffic even reach Asterisk when sipml5 is showing that it is in progress or is it on the browser side?
As I’ve said before with WebRTC, you have to learn the technology and understand what is going on. It can and will go wrong and requires digging into it.