Let's carry out the most basic interaction with a web browser audio/video through WebRTC. We'll start using SIP.js, which uses a protocol very familiar to all those who are old hands at VoIP.
A web page will display a click-to-call button, and anyone can click for inquiries. That call will be answered by our company's PBX and routed to our employee extension (1010). Our employee will wait on a browser with the "answer" web page open, and will automatically be connected to the incoming call (if our employee does not answer, the call will go to their voicemail).
To use this example, download version 0.7.0 of the SIP.js JavaScript library from www.sipjs.com.
We need an "anonymous" user that we can allow into our system without risks, that is, a user that can do only what we have preplanned. Create an anonymous user for click-to-call in a file named /usr/local/freeswitch/conf/directory/default/anonymous.xml :
<include> ...