Hi,
I am using SIP.js in combination with Asterisk and it works just fine. However I have an issue. when the user is in a call and let's say refreshes the browser window/tab the connection drops (call is lost or whatever you wanna call it). I would like this connection to stay alive so the user could be able to continue with the call after page has been refreshed.
I can confirm Verto in combination with freeswitch
https://webrtc.freeswitch.org/verto/index.html is able to do that (tested it). During call you can refresh the page and still return to the call and hear audio/video. I know that in this scenario there seems to be some server side modules for freeswitch which in return might be keeping the session alive. Not sure cause i didn't dive into it and I want to use sipjs not verto :)
First I wanted to see if anyone in the group has encountered this real life problem. Users will be users and I am sure it happens a lot they refresh the page and consequently loose the call. If the user can easily return to the call this would be a much better user experience.
Can someone point me in the right direction how to handle this issue? ANY pointers really appreciated.