
The package jscommunicator-web-phone provides a very basic demonstration
of how to integrate the JavaScript from a HTML web site.

This demonstrates how it could be integrated into a contact form
on a web site or an intranet-based corporate phone book.

It doesn't just work with any SIP server/proxy: it requires a SIP proxy
that supports the SIP over WebSockets transport.  This is currently
under development in most leading SIP proxies.  The latest version
of the Debian packages "repro" and "kamailio" provide WebSocket support.

Furthermore, users accessing the site require a WebRTC capable browser:

  Firefox nightly build
      http://nightly.mozilla.org/

  Chrome 25 or later
      https://www.google.com/intl/en/chrome/browser/beta.html

WebRTC requires a TURN server.  There are three TURN servers available
in Debian:

  reTurn from reSIProcate:
      http://packages.debian.org/resiprocate-turn-server

  Open TurnServer.org:
      http://packages.debian.org/turnserver

  rfc5766 TURN server project:
      http://packages.debian.org/rfc5766-turn-server

Finally, the WebRTC browser/phone may insist on some of the following:

  SRTP: any device you call must also support SRTP

  AVPF (SAVPF): many standard SIP devices just support regular AVP.

  Codecs: Opus and G.711 are the core codecs for WebRTC.  Your browser
  may support others.  Most deskphones support G.711, but not Opus.


