sip

A quick look at ENUM mapping telephone numbers to DNS

With my talk about Lumicall at FOSDEM coming this Sunday, I'm going to have a quick look at some of the technologies used in Lumicall over this week. The first is ENUM

Introduction to ENUM

ENUM, as defined by the IETF, is a convenient and powerful way to map telephone numbers to the latest communications technologies such as SIP, XMPP (Jabber) and web sites.

ENUM resolution operates much like reverse DNS lookup:

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).

Debian.org 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.

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