Debian SIP without reading the manual

Most people don't have time to read manuals. The Debian SIP service has a reasonably thorough user guide with screenshots but some people may just want to have this level of detail if they really have to troubleshoot a problem.

WebRTC is the fastest way to start

Just set up your db.debian.org RTC password and use it to log in to the https://rtc.debian.org site.

Lumicall: giving you choice when you dial

One of the distinguishing features of Lumicall, the free and open SIP/VoIP app for Android, is the unique dialing experience.

In fact, even regular softphones for the desktop/laptop don't currently offer the same power when you want to make a call.

In my previous blog, I looked at the ENUM technology that underpins part of this compelling new user experience. Here we can step back and look at the bigger picture.

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.

Launching FreePhoneBox.net

The https://freephonebox.net 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).

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:

DruCall 1.1.0 released

DruCall 1.1.0 for Drupal has just been released.

The main improvement over 1.0.0 is that it now permits the TURN servers to be configured instead of using hard-coded STUN server addresses. TURN servers (such as reTurn from reSIProcate or TurnServer.org) provide the ability to relay media streams on a public IP to ensure guaranteed NAT traversal. Support for ICE/STUN/TURN is a mandatory part of the WebRTC specification.

Understanding the WebRTC architecture from a free software perspective

WebRTC is here, now.

Google launched Chrome 25 with built-in WebRTC support on Monday, 25 February. Within 24 hours, DruCall, an easy-to-use WebRTC module for Drupal was available Free as a shiny new GPL project.

While this technology offers a level of convenience for end-users that is unprecedented, just what is involved for the keen server administrator who wants to deploy this today?