My company currently uses a third party paid service for web RTC, as the user base is increasing, the cost is getting higher.
We want to replace it with Asterisk. Please let me know if this is possible or not with it.
1: We operate a chat app so we want to add audio/video compatibility in it between users.
2: We do not have any plan to transfer web calls to Landline/Mobile. We want web to web calling.
3: Can I manage users in Asterisk so when user A initiate a calls to User B, Asterisk can manage that.
If this is all possible, where should I start from .
Yes Asterisk can do audio and video between users; in most cases its acting as a signalling server (SIP over websocket) but if you want it to terminate or passthrough media there are some things you should know - it can only do vp8 passthrough and not transcoding for video; and for audio it can do OPUS passthrough and with the recent release of Asterisk 14 it can now do OPUS transcoding if required.
Are these just random people who use the service or registered people who would have an extension configured on Asterisk etc?
If you’re not aiming to do PSTN or SIP connectivity, then Asterisk isn’t the best thing out there. Take a look at what signalling transports are out there - https://bloggeek.me/siganling-protocol-webrtc/ - &yet have a node.js based signalling server that utilises websockets using socket.io called signalmaster but there are a load of them out there. https://github.com/andyet/signalmaster
Thanks for reply. We are using custom modified ejabberd server for Text chat, we were looking for a better and opensource webRTC solution.
Yes the users are random, anyone can signup and start chatting like whatsapp, telegram etc.
No we are not aiming for PSTM or SIP connectivity. As suggested by @mkozusnik I will give a try to Openfire server. I am right now experimenting with janus-gateway for signalling.