Please see http://jscommunicator.org for more details about features, examples, mailing lists and the latest source code.
To use JSCommunicator, you require a SIP proxy that supports the SIP over WebSockets transport.
Only recent versions of the SIP proxies support WebSockets.
repro from reSIProcate http://www.resiprocate.org
See the Real-Time Communication Quick Start Guide for details about setting up a SIP and TURN server to support WebRTC calls.
JsSIP The latest code on the JSCommunicator master branch works with JsSIP version 0.6.x
jQuery (v1.4 or greater has been tested)
ArbiterJS (v1.0 has been tested)
Font Awesome (v4.1 or greater has been tested, earlier versions do not work)
All dependencies can be easily fetched using either of the following methods:
on a Debian/Ubuntu system, use the "deb-setup.sh" script
on other systems, see the "code_grabber" script
For integration in static or dynamically generated web sites, frameworks and Content Management Systems, please see INTEGRATION.md
In a CMS, wiki or other publishing platform:
For example, you can:
send a signal to JSCommunicator telling it which destination to dial
receive notifications from JSCommunicator when calls are made or received and use this information to query or update an address book or CMS, display related information about the caller in another part of the screen, etc
For an example, see the files:
JSCommunicator is based on the JsSIP library and inspired by the TryIt.jssip.net and RetroRTC demo applications produced by Versatica.
Copyright (C) 2013-2015 Daniel Pocock http://danielpocock.com
You may distribute non-source (e.g., minimized or compacted) forms of that code without the full copy of the GNU GPL normally required provided you include this license notice and a URL through which recipients can access the Corresponding Source.