Asterisk12 Webrtc Minimal installation

Good Day,

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.

I’ve been researching webrtc2sip and asterisk integration and am in the process now of building it on a seperate vm from my asterisk vm and i’ll try to integrate the two products. I’m still not sure if this is necessary with asterisk 12 but most of the documentation i’ve found so far says that it was up until asterisk 11.x even with webrtc and ice support added.

can multiparty videoconferencing still be done through the asterisk if webrtc2sip is used?

I’ve installed asterisk in legacy polycom/cisco videoconferencing networks to tie the clients ip phones to their vtc systems by establishing a sip trunk from the pbx to the cisco gateway. would that type of setup still be possible with a third party application like webrtc2sip?

I hope these architecture questions are in the right place :mrgreen: