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.