SIP, WebRTC and Asterisk ARI
The call platform runs on Asterisk, driven via the ARI (Asterisk REST Interface) instead of traditional dialplan logic. That makes routing, IVR steps and call events programmable and testable, without having to maintain dialplan scripts.
Agents connect via WebRTC directly from the browser; external trunks run over standard SIP. No proprietary, closed signalling protocol — whatever comes in via SIP can also go back out via SIP to another trunk or provider.