[spditner]

Indeed, I ran a large-ish asterisk user group for years in Toronto and know a number of people at Digium and Sangoma, and there was always a bit of animosity between them. Glad to see them finally coming together.

Most of the talent I know took a step to the right (like myself) and are working with WebRTC. It's all the same stuff! Just more signalling layers on top of RTP channel management. I quite often end up doing SIP over WebRTC with things like sip.js (still backing onto Asterisk and FreeSWITCH) and XMPP over WebRTC with jitsi-meet.

[jchassoul]

Indeed the SIP over WebRTC open source stack is a thing of beauty!