/usr/share/doc/jscommunicator-web-phone/README.Debian is in jscommunicator-web-phone 2.1.3-1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 | 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.
The package does not explicitly depend on all the requirements because
you don't have to run them on the same host as the web server process.
Therefore, please review the list of requirements and the setup instructions
below.
For a thorough overview of the SIP and WebRTC architecture, please
see the Real-Time Communications Quick Start Guide:
http://rtcquickstart.org
Requirements
------------
It doesn't work with just any SIP server/proxy: it requires a SIP proxy
that supports the SIP over WebSockets transport. 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
CoTurn:
http://packages.debian.org/coturn
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.
Setup
-----
You basically need to do one of four things:
a) Include /etc/jscommunicator-web-phone/apache.conf into one of
your Apache config files
b) symlink /etc/jscommunicator-web-phone/apache.conf into
/etc/apache2/conf-available and enable it
c) copy and paste things from /etc/jscommunicator-web-phone/apache.conf
into your own conf file
d) symlink the files from /usr/share/doc/jscommunicator-web-phone into
some other directory that is already accessible through Apache
You can choose any of those four strategies.
You can use another web server such as Nginx too. The Apache configuration
is very trivial and easy to adapt.
Once that is done, you need to modify the config.js file to include
the connection details for your SIP proxy, TURN server and domain name.
You can copy the /etc/jscommunicator-web-phone/config.js file to
each site or you can create a symlink to it.
Using JSCommunicator in other projects
--------------------------------------
If you are trying to use JSCommunicator as part of another project,
you don't usually need the jscommunicator-web-phone package, just the
libjs-jscommunicator package. Look at the file
/usr/share/doc/libjs-jscommunicator/README.Debian for more details
about embedding JSCommunicator and using it as a dependency.
|