For those following the use of Openfire for audio and video communication, The Jitsi project released very recently, the VideoBridge plugin for Openfire. If you use Jitsi with Openfire, you find it very useful for creating both audio and video conferences with multiple people.  I am hoping someone will try and make a Spark plugin for it

In the meantime, I did a quick experiment to see if I could convert the Redfire plugin for Spark to use WebRTC instead of RTMP/RTMFP

Guess what!!. It worked

It works the same way as the Redfire plugin for Spark. Click on the webrtc button to share audio and video with the person you are chatting with. On the other side, that person will receive a prompt to accept or reject the offer.

So here it is attached, if you want to play with it. It does not depend on any Openfire plugin, but will use the JingleNodes plugin if available to relay WebRTC audio/video media when a direct connection cannot be made between two Spark clients..

It needs Google Chrome version 26+ running on the client desktop as the default browser. It supports Google Chrome frame for Internet Explorer. The source code is in there as well. When Firexfox WebRTC support becomes stable, I will probably add support for that too.

*** UPDATE ***

I finally got multi-user video conferencing working in group chat like Redfire. Please note that there is no prompt. Each participant MUST click on the webrtc button to join the video conference and the plugin ensures a webrtc peer-connection is made between each and every participant. This should be done after everyone joins the groupchat.

If a new person enters the room and wishes to join the video-conference, each participant should close their browser windows and click on the webrtc button again.

Bla, bla bla...

WebRTC is on the march

With a stable implementation in Chrome since ver 23, applications are appearing as fast as flowers blooming in Spring. The nightly and developer builds (ver 25+) have more features including full desktop screen capture and ability to send it as a video stream over a peer connection. By adding DTLS encryption to Chrome nightly build, Google have achieved interoperability with Firefox nightly build.

Back in 2007 at ignite realtime, we built the first open source Flash to SIP gateway for SparkWeb and followed that up a few months later with Red5phone, the first open source web soft phone. Red5phone has now evolved into Apache OpenMeetings Red5Sip.

Fast-forward to today and we now have the iNum initiative from Voxbone which enables us to have a new geographical independent phone number for free phone calls on the Internet. iNums are global, supported by most telephone carriers including my former employer BT in the UK. Combine WebRTC with iNum and you can send and recieve free phone calls anywhere in the world provided you have a web browser and an iNum.

That was all the incentive I needed to get started on webrtc-phone.

Ok, so what does it do?

The prime purpose is to make and recieve free phone calls with iNums. It can also make SIP calls with domains that accept them.

For more details go to the project page.

How did I make it. What did I use?

• Customized version of Phono SDK for webrtc only that connects to the Voxeo SIP cloud and handles the Jingle signalling and WebRTC media. Flash support has completely been removed. It only works with Chrome ver 23+ for now. Firefox support will come later
• Openfire Server with websockets and redfire plugins to lookup iNums and map to Phono SIP addresses using XMPP Presence for anonymous users.
• Candybar and DialPad? GUI components from AT&T Foundry and &Yet
• AddThis social sign on service for Facebook, Google and Twitter

Free calls! Whats the catch?

Calls from iNum to/from the public PSTN still costs money. If you divert your iNum to a mobile or landline, your voice service provider will charge you for diverting the call. You will be paying for all calls from your iNum to your mobile or landline phone.

Calls from the public PSTN to your iNum cannot be intercepted and are therfore are not supported by webrtc-phone. If your iNum is dialed from a regular phone, the call will end up on whatever device you have forwarded it to. It is only when the number is dialed from a webrtc-phone elsewhere that it will be re-mapped to your webrtc-phone session.

Can I play?....

Live demo is hosted at https://webrtc.free-solutions.org:8443/webrtc-phone  by the very kind folks at free-solutions.ch who are also members of the ignite realtime community.

Project source code is hosted on a Git repository at http://code.google.com/p/webrtc-phone/source/checkout, so go ahead and clone