Verto, in demo example configuration, always ask for authentication to phones (eg, browsers) that are registering. So, it behave like the "internal" SIP profile: the "user_context" variable associated to the authenticated user will be the dialplan context that will route the calls originated by that user (browser).

So, an incoming call from a DID number, sent by our ITSP to our FreeSWITCH server via SIP, can then be routed to the authenticated user registered via its browser as a Verto WebRTC client. Also, a call made by an authenticated Verto user connected via WebRTC browser will be routed by "default" dialplan context, and so it will be able to send outbound calls via the SIP gateway to the external world. This is the simplest possible way to implement a fully featured SIP/WebRTC bidirectional gateway.