so far i’ve built a few small webrtc applications that mainly focused on browser to browser communication so nothing special was required. I am currently working on a project where the client has two important must-haves. The first is that they be able to communicate with their Polycom VTC systems, the second is that they be able to do multiparty calls.
With the current iteration of Asterisk and the added support of ICE, websockets and webrtc is anything necessary for an installation of this kind besides just Asterisk? will I need to use a SIP library (i.e. jssip) and Proxy (Kamailio or OpenSIPs) in order to communicate with the legacy devices. And lastly are there any examples of asterisk using an external transcoder? I’m assuming that transcoding is still necessary from the browser even though H.264 has been open sourced.
I don’t need a step by step or anything, i’m just trying to find out what items are still required to make a browser–>legacy call work at this time.