Roll your own video conferences with WebRTC and the Jitsi VideoBridge

von Philipp Hancke (ESTOS GmbH)

WebRTC brings voice, video and peer-to-peer communications to the open web platform. For users that means hassle-free video conferencing without installing plugins or dedicated software. Just go to a web page and wait for your communication partner. But is it secure? You're allowing a web page access to your microphone and camera, after all!

For developers, it means learning a lot of new tools. WebSockets, data channels, signalling protocols (like XMPP) for communication between browsers, STUN and TURN servers for dealing with NAT, and many more three letter acronyms. All that while the standard is still evolving and currently only implemented in two browsers.

For operators, there are more questions. Can you run all this yourself, on your own infrastructure? Is it interoperable with existing VoIP software? Will anyone pay for it or do they expect to use it for free?

And everyone is interested in features. Will WebRTC enable us to do things that we couldn't do with more traditional VoIP systems, like whiteboarding and shared editing? And how about multi-user conferencing, like Hangouts?

That's a lot of questions! We'll cover many of the answers by introducing you to JitMeet and the Jitsi VideoBridge, two open-source applications that provide code that you can use, build on, and deploy for a simple but fairly complete WebRTC solution.

Über den Autor Philipp Hancke:

Philipp Hancke is a long time contributor to the Extensible Messaging and Presence Protocol and is a member of the thirteenth Council, providing technical leadership to the community. He got involved in XMPP via the psyced server in 2004. This lead him to author a number of specifications related to server-to-server connections.

Philipp’s day job as Techlead WebRTC at ESTOS GmbH involves making xmpp <3 webrtc. One of the results of this is the strophe.jingle library, which allows federated videochat using WebRTC and Jingle.