Does sipML provide any info about call quality? Something like dropped packets or packets arriving out of order? I have looked at sipML API documenta ...
Does sipML provide any info about call quality? Something like dropped packets or packets arriving out of order? I have looked at sipML API documenta ...
I am using asterisk 15.5 as voip server and twillio trunk to make outgoing and incoming call but when I hangup on an incoming call to sip client then ...
I have setup Kamailio with websocket module. When I register with sipML5 its going well. But returns 488 Not Acceptable Here when I trying to call. 4 ...
I am trying to connect audio call through sipML5 API in MS edge using webrtc and adapter.js, but it gives error Timeout for addremoteCandidate. Consid ...
I want to setup asterisk 13 that working on ubuntu 16.04 on local machine to enable WebRTC, I'am testing with https://www.doubango.org/sipml5/ on fire ...
I had built a WebRTC system based on Asterisk and sipml5, and I could make audio calls on my smartphone(Android), but when I enables the video, the ca ...
I am trying to make a web client for my SIP call request. I have done invite call successfully from browser. But, I am not getting how to refer to the ...
I tried to call on sipml5 through 2 browsers. Even though the call has been initiated, we can't listen anything from another side. How to tackle this ...
regarding this problem: Call get disconnected while I am refreshing the SIPML5 demo page . can be found here https://groups.google.com/forum/#!msg/do ...
With the recent release of Firefox Version 58, I have encountered a no audio issue using sipML5, I suspect it has to do with the change they did where ...
I have attempted using sipML5 instead of tryit jsip but I have not been able to figure out the configuration. I am having kamailio listen on port 1500 ...
I have an issue getting the remote display name with the sipML5 library. When I register the user, I set in the stack object the display_name. Aft ...