I am working on a webRTC application and would like to be able to support multiple calls and be able to call from the browser to legacy VoIP or Videoconferencing systems as well as browser to browser.
now that Asterisk has added websocket in their latest builds would you need SIP and a SIP proxy in order to communicate with VoIP systems or will Asterisk allow this?
now that H.264 has been open sourced by Cisco would you still need a transcoder in order to call a legacy VTC system?
Is Node.js the preferred technology for implementing webrtc client/server deployments? I've looked into Mobicents SIP Servlets a bit but that seems to be the only alternative technology available beside a node.js solution.
If needed I am planning on creating a SIP trunk between an Asterisk server and our Polycom VBP so the webrtc clients should be able to get presence information through that connection so if no media transcoding is required with the recent changes then media should be able to pass directly from polycom endpoint to browser with the asterisk handling the signalling.
Thank you anyone who is able to answer any of these questions, it is still early in the r&d portion of this project for me and i'd like to get as much information as possible.
also: i did see SIP over websockets to true SIP. I understand that "something" needs to stand in between the webRTC client and the VoIP phone or Legacy SIP endpoint. what I would like to know is if that can be just asterisk with the recent update. if asterisk is all that is required, is there a way to include a media transcoder like red5? I haven't seen anything in the webrtc API that would allow you to include a transcoder, asterisk has transcoding mods but none that will do vp8 to h.26x or Opus to anything as far as i know.
