The SIP proxy accepts the WebSocket connections from the web browsers and helps users locate each other and set up their calls.
The easiest way to set up a SIP proxy is to install the packages on a Linux system using the instructions from the RTC Quick Start guide.
Once your server is running, just install the packages for one of the following SIP proxies:
For quick testing, just create two users in the SIP proxy, use these two accounts for calling each other.
The easiest solution is to test JSCommunicator as a static HTML page in an Apache server
Install the Apache package to your server (it does not have to be the same server where you run the SIP proxy)
Download the JSCommunicator files and JSSIP into a web server directory, for example:
cd /var/www git clone https://github.com/opentelecoms-org/jscommunicator.git JSCommunicator cd JSCommunicator wget -O JsSIP.js http://jssip.net/download/jssip-devel.js
Now edit the file config.js to include your SIP proxy and TURN server settings.
Finally, browse to http://my-server-name/JSCommunicator/phone.shtml and the phone should start