A question that comes up more and more these days: what's the quickest way to try WebRTC and see it working? How can a web developer start experimenting with WebRTC in their blog or demo site?
Good news: it's no longer necessary to compile anything from source - and many of the components are available on Debian-based systems (including Ubuntu) or RPM-based solutions like Fedora
A quick look at how easy it is, explanation below:
# apt-get update # apt-get install -t experimental repro resiprocate-turn-server # apt-get install -t unstable chromium sipml5-web-phone # cd /var/www && mkdir jssip && cd jssip # wget -r -nH http://tryit.jssip.net # vi /etc/repro/repro.config # vi /etc/reTurnServer.config # vi /var/www/jssip/js/custom.REMOVE_THIS.js
and then try browsing to /jssip or /sipml5-web-phone
As explained in the RTC Quick Start guide for regular RTC, a SIP proxy is a clean and simple component to start with. The same is true for WebRTC: start with a proxy. There are two I'll emphasize here:
TURN servers help media streams traverse NAT. They are very easy to set up, but must have real IP addresses.
Three working samples to start with:
Users need a recent browser.
The latest Chromium packages in Debian are based on Google Chrome M26 code and this should work.
Please come and ask on the users mailing lists or IRC channels for any of the packages mentioned here.