Saúl Ibarra Corretgé

About a year ago we added support for 500 participants in Jitsi Meet. Today I'll show you how we are taking that to the next level: 10000 yes, that is 10 thousand participants!

Reaching 500 participants was a huge ordeal because it stressed every component in our architecture, from the User Interface all the way to our XMPP server and everything in between. In order to reach 10000 participants thinking outside of the box was necessary. Thanks to some clever tricks like using the visitor role and multiple XMPP MUCs we were able to get there.

This presentation will get into the gory details of what it took to get there!

QuickJS is a (small) JavaScript engine by Fabrice Bellard.

Over the past 6 years I've built a couple of small JS runtimes, first using Duktape, then using QuickJS. The premise was to always use existing libraries to add functionality, including WASM support.

This presentation will go through the differences of both libraries and the 2 projects I built leveraging them.

There are many libraries out there which one can use to build a JS runtime these days, with the most well known one being V8.

Node and Deno are both using V8, but how would a different take at building a JS runtime look like?

I asked myself this question some years ago and started building a small project with Duktape. That project was ultimately a failure, but I learned very valuable lessons in runtime design and always wanted to revisit that project.

Fast forward a few years, QuickJS was released by Fabrice Bellard, and I was excited! It was so easy to integrate that I began to implement a new runtime using it. This runtime is called txiki.js and while it's a small project it already integrates libuv for fast networking and the event loop, curl for making HTTP requests and wasm3 for WASM (and WASI!) support.

The Jitsi project was founded almost 20 years ago, under a different name and just for academic purposes.

Almost 20 years later, the project could not be in a better shape! It has gone through a name change and several acquisitions by large corporations, while remaining current in the industry and innovating all along.

This presentation will take a trip down memory lane and see where Jitsi came from, how it has managed to stick around for so long, and what the future holds.

When the pandemic (first) hit, many turned towards online solutions for having their business meetings, or gatherings of any kind really. Jitsi Meet being an Open Source and privacy respecting service saw a huge surge in usage (through the free meet.jit.si service) which we hadn't anticipated.

In this presentation we'll share our lessons learned keeping up with the huge load, what improvements were made since and where the project is heading in 2021.

With the advent of Zoom's questionable use of the term "End to End Encryption" many turned their eyes to FLOSS solutions. Thanks to the insertable streams feature which shipped in Chrome at just the right time, we were able to ship a working E2EE implementation in a reasonably short amount of time.

In this presentation we'll walk through all that was needed, what our plans for the future are and how others can leverage what we learned to add E2EE to their WebRTC applications.

Jitsi is a set of Open Source projects which provide state-of-the-art videocconferencing capabilities. In this presentation we will explore the Jitsi ecosystem from a privacy minded point of view.

Communicating privately via a public network (specially if we are using video) can be challenging. Jitsi provides the necessary tools to do so and we'll explore the seccurity model employed by our tools and how to setup a Jitsi instance with ease, while respecting your privacy.

NOTE: This talk was moved from the 9:00 original time, replacing a talk on "Migrating reSIProcate to C99 stdint types" by Izabela Bakollari.

Jitsi Meet started out as a web application, but when mobile became a target we tackled it in a relatively unusual way: the web and mobile share all the business logic wth mobile using React Native for platform support and presentation.

Jitsi Meet started out as a web application, but when mobile became a target we tackled it in a relatively unusual way: the web and mobile share all the business logic wth mobile using React Native for platform support and presentation.

The talk will go through the challendes and solutions we found along the way.

libuv is the platform abstraction layer used in Node, Julia, NeoVim and many other projects. It features an event loop, timers, signals, as well as many cross platform utilities including threading and name resolution.

This lightning talk will introduce libuv and its current development state.

Jitsi offers seamless and secure multiparty video conferencing while being 100% Open Source. Come and see what's new in Jitsi land! This talk will dive into the latest developments in the Jitsi Videobridge such as simulcast support and the new React Native based Jitsi Meet, the video groupchat client which just went mobile.

Jitsi started out as "SIP Communicator" back in 2003 and has been in the forefront of Real Time Communications ever since. One of Jitsi's biggest innovations was the introduction of the videobridge, which made video groupchat possible. Lately the Jitsi team has been focused on adding new features to the videobridge such as simulcast along with the development of a React Native based client for mobile platforms. All of this while being secure and 100% Open Source!

Many companies and individuals have already deployed large SIP networks with tons of endpoints. With WebRTC becomeming so ubiquitous, it makes sense to adapt for these new types of devices.

In 2012 SylkServer (http://sylkserver.com) allowed for seamless SIP to XMPP interoperability, now we are adding WebRTC as a first class citizen. Using SIP as the core routing and rendezvous protocol and either SIP, XMPP or WebRTC at the edges we can reuse existing SIP insfrastructure with the right gateway in place.

Introduction to OP^2 Project, which provides a framework for implementing SIP hardware phones.

Project OP^2 started as a self-learning experience to see if the features present in SIP desktop clients could be brought to a hardware device. Traditional VoIP hardware phones are very closed systems, and Project OP^2 was designed with openness and flexibility in mind. The first prototypes were built using a RaspberryPi as the hardware platform.

Introduction to WebRTC and application development using Python on the server side. Let's build a "Call Roulette"!

WebRTC is a recent technology present in modern web browsers allowing developers to create rich multimedia applications using realtime audio, video and data. This talk will give an introduction to WebRTC and walk through the implementation of a simple Call Roulette application, designing the necessary signaling protocol and implementing it on the server side using Python3 and asyncio.

Last year I talked about how PEP-3156 and Tulip (the canonical implementation) would change the async i/o landscape in Python. A year later it became real, merged into stdlib, and will be part of Python 3.4. We'll dive deep into the internals of this new module and learn how those coroutines, tasks, and future work together.

Greenlet is a pretty well-known way for implementing cooperative micro-threads in Python, but how does it actually work? How is it similar and different from Stackless? We'll take a peek at how PyPy implemented it using a small library called 'stacklet' and how the python-fibers project takes advantage of it to build a similar project.

Since the C10K problem, event loops have been used together with IO multiplexors to solve IO scalability issues. There are different event loops which can be used in Python, some high level ones like Twisted or Tornado, and some low level ones like pyev or pyuv, or you can even roll your own!

As the author of pyuv I'll go through the inners of the libuv low level event loop and show how it can be used to build a higher level one, that is, how to use it with Twisted, Tornado and even Gevent.

SIP and XMPP are the two main protocols used nowadays for realtime communications with presence and instant messaging. They both offer similar features even though their development started quite differently. They both coexist and the interoperability between the two is key in order to offer a communications solutions which covers a broader number of users.

Throughout the course of time several gateways have been built to convert SIP to XMPP (and the other way around) but never in a fully transparent way. SylkServer implements a transparent gateway for SIP and XMPP. It's Open Source and during the talk I'll comment on its design, what advantages it has and what problems had to be solved.

SIP is the de facto protocol for VoIP these days. It's now 10 years old and there are many extensions to it, which have turned it into a complex beast which it not always easy to manage. Should we dump it?! Join me and let's go shopping to find the proper protocol for solving some of the well known problems in SIP.

SIP is the de facto protocol for VoIP these days. It's now 10 years old and there are many extensions to it, which have turned it into a complex beast which it not always easy to manage. Should we dump it?! Join me and let's go shopping to find the proper protocol for solving some of the SIP well known problems such as instant messaging and presence. We may find a better protocol, or a better way of doing things in SIP to give our users the proper communication experience with different media types, chat and availability (presence) while using Open Standards and Open Source Software.

Being the most relevant protocols for realtime communications it stands to reason that interoperating between SIP and XMPP has gains to both. I spent most of 2012 implementing such a gateway and while the xmpp-sip drafts helped a lot there were gaps that needed to be filled. I'll walk through the implementation decisions and challenges we found along the road.

SIP can be used for much more than 'plain phone calls', and this talk will demonstrate chat, file transfer, contact lists and more!

PSTN emulation and audio calls are by far the most used applications for VoIP and SIP these days. As our communication needs grow we are in need for new means of communication, however. Why use the phone to do VoIP, email to send files and jabber to have a chat channel in the context is a single conversation? We can merge those 3 communication channels within the same call with SIP today. In order to find the perfect moment for the conversation presence may be used and policies for presence together with our own contact list may be stored in a remote server and synced with all of our devices. All this can be done today, with SIP, in a standard way and with Open Source tools.