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JSCommunicator to power DruCall 2.x

JSCommunicator 1.0.0 appeared over the festive season, making it even easier for people to build WebRTC real-time voice and video capabilities into their web sites.

xWiki: 10 years and a WebRTC success story

Six months ago, I wrote to the leaders of several open source web frameworks and asked them about their vision for WebRTC and if they would come to the WebRTC Conference in Paris this week (now finished). The most promising response was from Ludovic Dubost, founder of the xWiki project.

Get WebRTC going faster

On Saturday, Lumicall began offering free calls from browser to mobile using the free and open WebRTC technology. It should be no surprise that the service has been popular.

Is it really free and open?

The only way to prove this technology is free is to help people implement this for themself.

Using reSIProcate to connect Asterisk with WebRTC

In my previous blog entry about how to get WebRTC going fast I looked at the basics of setting up a SIP proxy (also known as a SIP router) to accept connections from WebRTC clients. As in a traditional, non-WebRTC world, the SIP proxy simply facilitates calling between all the clients it knows. In practice, deployments usually want to add additional functionality in the form of a PBX with queues, voicemail, menus and conferencing.

Get WebRTC going fast

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:

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