Hi All,
I am new at asterisk , and i am already implemented android application that support P2P voice call using Webrtc library, and i want to add support of conference call (Voice/Video).
I can see from documentation for Asterisk 15 , it is support Webrtc, and i think with this support, i can use my library to connect to asterisk, but after deep searching in internet and forums, i found most people use sipml5 or JSSIP which are SIP protocol over websocket.
may be there is something missing for me, if asterisk 15 is already support webrtc , so why i cannot use normal webrtc library ?!, why i should use SIP library?!
i hope if anyone can make things more clear to me.
WebRTC does not define a signaling method. It was done in SIP because it was the easiest to do and client libraries exist. If you have your own signaling mechanism then you’d have to add your own support.
Thanks for your reply.
yes, i have my own signal server, so can i use webrtc library to connect to asterisk server ? if yes, how i can do that ? if you have any code example.
also, i think (may be wrong) asterisk itself is also doing signaling , is this correct statement ?
Asterisk currently uses SIP as the signaling to transport the SDP and do other things. There is nothing built in to support your situation. You have to write a completely new channel driver implementation, handle SDP, etc. I don’t have a code example because that’s a massive thing to do.
Also it is not an end user activity, so the best place for support is likely to be the developer mailing list or IRC channel, but even then it is unlikely you will get anything useful from such a wide requirement.
Thanks again for your appreciate support
OK, i got your point, but there is a point confusing me, as webrtc user what i gain from statement “Asterisk 15 is supported Webrtc” ?