WebRTC calling from Firefox on CyanogenMod to Chromium on Debian wheezy

Just how easy is it to make a call with WebRTC?

Here I have two endpoints:

DruCall updated, Lumicall talk at FOSDEM next weekend

The latest version of DruCall is now pushed to the Drupal Git Repository. As described previously, this version uses JSCommunicator (which is based on the popular JsSIP) as the underlying phone framework. I would be very grateful if somebody from the Drupal community could kindly help my projects escape from the sandbox.

Test numbers for SIP and WebRTC

Want to test or troubleshoot WebRTC calling or SIP calling without bothering anybody?

Now there are some test numbers you can call by just clicking in your desktop or mobile web browser.


The service has now gone live. This is a great way for people without a SIP account to call those who do have one (like all the Debian Developers who got SIP on Saturday) using nothing more than a web browser (no plugin required). enabled for SIP federation, who will be next?

The Debian community has announced the availability of SIP for all of approximately 1,000 Debian Developers who comprise the membership of the organisation.

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: