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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.
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