Emil Ivov

The talk will describe the problems of scalability for libre video conferencing and how they are being addressed by the new video routing architectures. It is gong to go over optimization techniques such as Last N and Simulcast. The presentation is going to then go over the Jitsi Videobridge FLOSS video router and talk about how developers can use it in their video conferencing applications.

Until recently high quality, multi-party video conferencing has been an ever elusive dream to many of us. While it has existed for tens of years already, it has always been been prohibitively expensive and a luxury that only rich corporations could afford.

Readily available broadband, has opened up new horizons in the field however. The traditional video mixing servers (MCUs) are becoming irrelevant today and are being replaced by infinitely more scalable video routers, also known as Selective Forwarding Units (SFUs).

Google Hangouts and the underlying Vidyo technology they used, have been one of the more popular examples in the past couple of years. Early 2014, jitsi.org also released Jitsi Videobridge, an open source video router (SFU) that is compatible with WebRTC and that allows FLOSS developers around the world to start building scalable video conferencing applications.

This lecture is going to go over some of the more important concepts of scalable video conferencing, such as selective routing, simulcasting and layered video, RTCP termination and others.

We are then going to look into the privacy issues related to video conferencing today, discuss the possibilities for truly confidential multi-party calls and the interesting impact, both positive and negative, that WebRTC has on the possible solutions.

Are you one of the people who has never installed Skype (and never will)? Do you feel a sense of frustration every time someone asks to call you with Skype? Does the excruciating inefficiency of maintaining multiple profiles across Facebook, Google+ and LinkedIn give you more pain than the sound of fingernails scratching a blackboard? Have you tried free and open source alternatives, but found their usefulness limited (or completely failed to get them working)? Would you like to see a way out of this nightmare?

This talk covers the depth and breadth of VoIP and all other real-time communications, encompassing desktop VoIP, instant messaging, social networking, microblogging, server solutions and also emerging areas like WebRTC, Peer-to-peer, privacy with FreedomBox and the mobile VoIP sector.

Despite this broad agenda, the presentations from each speaker have been tightly coordinated with a coherent focus on the challenges of making it all work together comprehensively and conveniently. The speakers have come together for FOSDEM with the goal of mapping out a strategy for free and open communications standards to finally take their rightful place as the dominant paradigm.

The speakers are the leaders in the flagship open source VoIP projects and will participate in a panel discussion and question/answer session in the telephony devroom immediately after the talk.

Other people: Peter Saint-Andre Bdale Garbee Daniel Pocock Simon Tennant Evan Prodromou Daniel-Constantin Mierla

This talk presents the Jitsi project's evolution of handling multi-party video conferencing. There are various approaches to handling multi-party video conferencing, and the Jitsi project approaches the problem in a fashion similar to Skype and Google. This has culminated in a new XMPP server component that focus agents can control via dedicated XMPP IQs: Jitsi Videobridge.

About a year ago the Jitsi project developers started work on support for video conference calls. We had had audio conferencing for a while at that point and we were using it regularly in our dev meetings. Video seemed like a logical next step so we rolled our sleeves and got to work.

The first choice that we needed to make was how to handle video distribution. The approach that we had been using for audio was for one of the participating Jitsi instances to mix all flows. That's easy to do for audio streams and any recent machine can easily mix a call with six or more participants.

Video however was a different story. Mixing video into composite images is an extremely expensive affair and one could never achieve this real-time with today's desktop or laptop computers. We had to choose between an approach where the conference organizer would simply switch to the active speaker or a solution where a central node would relay all streams to all participants, while every participant keeps sending a single stream.

We finally went for the latter which also seems to be the approach taken by Skype and Google for their respective conferencing services. We started by implementing all this in Jitsi but along the way we also decided to make the RTP relaying part a separate server-side component. This is how Jitsi Videobridge was born: an XMPP server component that focus agents can control via dedicated XMPP IQs.

A description and demo of the Conference Bridge extension (cobri), which enables an XMPP client to export the RTP relaying aspects of a video conference into the cloud.

Video Conferencing is generally a costly and resource consuming affair. Lately however, increasing bandwidths have driven a shift away from video mixing and toward RTP translation/relaying. In other words: rather than mixing it, just send it all over the network. Skype video conferencing and Google hangouts are both examples of this trend.

Running a bridge that is capable of handling that kind of RTP relaying can be very inexpensive and one can even do it on most modern desktops/laptops. The problem with doing it is bandwidth: few Internet connections would allow streaming video to five or six different participants in the same time.

The Conference Bridge extension (cobri) allows an XMPP client to export the RTP relaying part into the cloud, run it as an XMPP component and control it through a bunch of IQ commands.

This talk is about presenting the "cobri" extension and the sample implementations in Jitsi and Jitsi Videobridge.

The University of Strasbourg has built a 'Skype replacement' for ~60,000 users using open source standards and software.

A few years ago the French Ministry of Education and Research, issued a memo that strongly advised against use of Skype in universities and research centers. Security and excessive bandwidth consumption were among the main concerns. In practice however, the directive has been rarely applied for one major reason: (perceived) lack of alternatives. Therefore, in the beginning of 2011, the University of Strasbourg has started an effort that aims to propose an actual Skype alternative to both its faculty and students (~60 000 users). The project started with a number of extensions added to the Jitsi project in order to add support for features such as GTalk interoperability for example. It then continued with a deployment of the actual network using exclusively FLOSS projects such as Openfire, Jinglenodes, FreeSWITCH/Asterisk, TurnServer and others. The talk will also cover aspects such as NAT traversal with ICE and JingleNodes, redundancy, security and confidentiality.

A talk about Jingle, its variants, its features, its NAT traversal, how it compares to SIP and everything you can do with it.