Alessandro Toppi

Say hello to Janode, a Meetecho official node.js SDK for Janus.
By wrapping the Janus API, Janode will help users build ad-hoc applications servers with Janus at its core.
This talk will give insights about the library internals and share some examples of projects running in production.

Janus is one of the most popular open-source WebRTC servers.
It provides a specific Janus API to let clients create WebRTC sessions and interact with its plugins, including the well-known "VideoRoom" SFU plugin.
There are many reasons not to directly expose the Janus API to clients though. Common examples are security concerns, API tailoring or the need of an Application Server in a complex architecture. 
This talk will introduce Janode, a Meetecho official server-side SDK for Node.js that fully wraps the Janus API, the Janus Admin API and the APIs of the most used plugins (like AudioBridge, Streaming and VideoRoom). The talk will also share some details about its internals and provide successful examples of how the library has been used in production in the last few years.

Janode: the Janus node.js SDK

Alex Vlahopoulos

Presenting Subspace, one of our platinum sponsors.

Platinum Sponsor: Subspace

Amandine Le Pape

A chat about the Online Safety Bill with Matt Warman (MP), Amandine Le Pape and Neil Brown.

A moderated chat about the Online Safety Bill

Andreas Granig

SIP and WebRTC testing at scale is hard, especially with large amounts of media. In this talk, we will connect open source building blocks to orchestrate distributed test agents and aggregate their result metrics in a cloud-centric fashion.

This presentation will give an insight into the architecture of Sipfront, a cloud-based telecom test automation service. We will present the open source components such as sipp, baresip and chromium to generate large scale test traffic, kamailio and rtpengine to extract real-time media statistics, as well as clickhouse, cloki and grafana to store, explore and present the huge amount of metrics being produced during test runs, and how these components interact with each other within a global AWS ECS deployment.

Building a modern RTC test stack in the cloud

Antony Jukes

Two years ago, Callable.io made a decision to move from a commercial CPaaS to an in-house system, based on jambonz.  We will discuss our experiences and lessons learned along this journey, as well as showing you some of the more interesting and novel aspects of the solution we were able to create.

Callable.io is UK-based company providing disruptive wholesale CPaaS to reseller and service providers worldwide.  Two years ago, we made a strategic decision to move our service from a commercial third-party CPaaS provider to an in-house system, based on [Jambonz](https://jambonz.org) which we would manage and grow internally. We knew we were taking on lots of additional responsibility in doing so, but we also felt we needed to drive additional innovation into our platform that we couldn't do with a 3rd party provider.  We will discuss our experiences and lessons learned along this journey, as well as showing you some of the more interesting and novel aspects of the solution we were able to create. 

Callable the journey to CPaaS Autonomy

Artur Shellunts

Do you want to participate in open source community? Or improve your coding skills, learn rtc, learn new programming language?

Then let me share my story and experience from first first patch to merging myself changes to the library. It is fun! (especially with golang)

My story of becoming Pion contributor and learning Go alongside.

How to contribute to open source (rtc edition)

Ashley Blewer

Daily focuses on WebRTC audio and video, but what about audio/video to text? This talk shares how we implemented transcription, including the magic tricks to pull RTP audio streams out of mediasoup, create an LPCM stream, send audio via web sockets, and get the transcription output back to clients.

Live transcription is a practical and highly requested feature for video chat platforms. Transcription helps those who are hard of hearing, supports national and institutional accessibility regulations, allows people to quickly refer back to earlier parts of a call, and more. Historically, transcription services have been expensive, unreliable, or unavailable for general use. 

At [Daily](https://www.daily.co), our primary focus is providing the best developer time to value for anyone working with WebRTC audio and video. But what about audio/video to text? This talk details the planning and design work around optimizing for live transcription and ease of use for our customers. 

This talk will address the following questions:
* Why did we decide to partner with a service provider (Deepgram) and how did we work together to deliver high-quality and AI-powered transcripts of Daily calls?
* What approaches did we take to support additional provider-agnostic audio transcription services, for users that want to Bring Their Own Transcription?
* What are all the technical magic tricks that went into pulling RTP audio streams out of mediasoup, packaging different streams into the easily-digestible LPCM audio stream, sending audio via web sockets, accepting the transcription output, and sending it back to clients via our signaling web socket? 


Pulling text out of a (mediasoup) hat and other magic tricks for a captivating live transcription performance

Christian Stuff

When adding text chat history to Daily Prebuilt, an embeddable WebRTC video chat interface, we needed to avoid storing and maintaining potentially sensitive user data. In this talk, we’ll share patterns for solving shared non-persistent P2P state across thousands of clients via network packets.

We build  [Daily Prebuilt](https://www.daily.co/prebuilt), a ready-to-use embeddable video chat interface, on top of  [daily-js](https://github.com/daily-co/daily-js). Every Prebuilt feature request is a chance to put our WebRTC APIs to the test. 

When adding text chat history, we wanted to avoid storing (and maintaining) potentially sensitive user information in a database. We needed to find a way to share non-persistent state across thousands of clients through network packets, while avoiding potential pitfalls and bottlenecks.  

In this talk, we’ll introduce patterns for solving shared P2P state that we learned from adding a record-less HIPAA compliant chat history to Daily Prebuilt.

Your state is my state and my state is your state: a tale of WebRTC chat history and shared P2P state

CommCon

Lightning talks are your opportunity to talk about something you feel passionately about. Maybe you didn't know about our CFP  or maybe you didn't think you could talk about that "thing" for 30 minutes. This is your chance to  get up  and talk for 5 minutes on that "thing". Whether you're a seasoned speaker or a first timer; you're welcome!  Submit your session to [team@commcon.xyz](mailto:team@commcon.xyz) or talk to us in #lightning-talks on Matrix.

* **Kshitij Gupta** - Horizontal scaling of WebRTC backend systems
* **Marco Vidonis** - Building billing into Broadcaster.VC

Lightning talks are your opportunity to talk about something you feel passionately about. Maybe you didn't know about our CFP  or maybe you didn't think you could talk about that "thing" for 30 minutes. This is your chance to  get up  and talk for 5 minutes on that "thing". Whether you're a seasoned speaker or a first timer; you're welcome!  Submit your session to [team@commcon.xyz](mailto:team@commcon.xyz) or talk to us in #lightning-talks on Matrix.

Lightning Talks

Let's get together for a drink or two... virtually! We're not really sure the format this will take yet, but we'll get together somewhere for a drink - whether that's a cup of tea, coffee, energy drink or a whiskey or two. Whatever your timezone, come join us and have a drink... or three!

Drinks!

Dan Jenkins

Welcome and introduction to CommCon 2021 Virtual.

Hello, CommCon!

[Broadcaster.VC](http://broadcaster.vc/) was born from last year's CommCon Virtual 2020 with our real life experience of using remote tooling into AV mixing software professionals use day in day out... and ultimately we were unimpressed. Come and find out about [Broadcaster.VC](http://broadcaster.vc/) - one of the tools that made CommCon Virtual 2021 happen.

Broadcaster.VC

A reflection on the CommCon week, acknowledgements and closing remarks.

Closing remarks

Darren Cubitt

Dave Horton

Do you find it strange that that we have been rehashing the same issues, over and over, endlessly -- what license should I choose?  What business model is right for me?  We will look at the fields of political economy and evolutionary biology for inspiration on creating vibrant OSS communities.

In 2009, something quite strange happened in the field of economics.  A woman named Elinor Ostrom, who many professional economists had never heard of, won the Nobel Prize in the field.  Her seminal work was in the field of "Design principles for Common Pool Resource Utilization".  Many of the principles that she outlined are relevant for OSS communities that seek to become highly functioning.  We will look at her work, along with the work of David Sloane Wilson, an evolutionary biologist who focused on group selection theory, to shed some light on the conditions necessary for altruistic groups like OSS communities to thrive and prosper.

Looking for love in all the wrong places -- why your OSS License won't save you

David Duffett

Get instantly useable advice and tips that any Geek can use to transform their presentations and connect with their audience.

The gap between “know what you are talking about” and being able to communicate that knowledge and experience to ‘normal’ people with power and clarity is massive!

And yet so many Geeks fall short in this crucial ability, which is so important for many areas of work, including job interviews, sales and presentations and attracting funding for projects.

David Duffett has distilled the task of delivering powerful and captivating presentations into 7 Protocols that any Geek can follow to get great results. Those Protocols are:

Purpose, Plan, Prepare, Practise, Participation, Presentation and Post-Event (there is also a MAGIC FACTOR that runs through all 7 Protocols)

Attend this presentation for a deep dive into one of the above Protocols and receive ‘take home’ advice you can work with immediately.
Which Protocol will we examine? You’ll have to attend to find out!

Deep dive into a Geek Speak Protocol

Fred Posner

What if you could proactively protect your systems from unwanted SIP traffic with an API call? And, what if you could block traffic in iptables quickly via API. Well my friends, APIBAN is here to help.

APIBAN helps prevent unwanted SIP traffic by identifying addresses of known bad actors before they attack your system. Bad actors are collected through globally deployed honeypots and curated by LOD/APIBAN. And, we now offer a very simple self-hosted API for managing a local iptables chain.

This talk will focus on how APIBAN works, how to quickly integrate, and the future goals of the project.

APIBAN

Gavin Henry

SentryPeer  a fraud detection tool. It lets bad actors try to make phone calls and saves the IP address they came from and number they tried to call. This project is all about Peer to Peer sharing of that data.

SentryPeer is a fraud detection tool. It lets bad actors try to make phone calls and saves the IP address they came from and number they tried to call. Those details are then used to block them at the service providers network and the next time a user/customer tries to call a collected number, it's blocked.

Traditionally this data is shipped to a central place, so you don't own the data you've collected. This project is all about Peer to Peer sharing of that data. The user owning the data and various Service Provider / Network Provider related feeds of the data is the key bit for me. I'm sick of all the services out there that keep it and sell it. If you've collected it, you should have the choice to keep it and/or opt in to share it with other SentryPeer community members via p2p methods.

The sharing part...you only get other users' data if you share yours. That's the key. It could be used (the sharing of data logic/feature) in many projects too if I get it right :-)

SentryPeer - A distributed peer to peer list of bad IP addresses and phone numbers collected via a SIP Honeypot.

Joachim Vanheuverzwijn

It all began with a second hand video card and a sleepless night. I started hoarding hardware and talking to the machines.
It escalated and now they talk back…
We’ve been quietly making our own speech recognition models ever since and will explain you what's feasible and how to do it.

2 years ago, we decided to make "reasonable quality" transcripts,  as in enough to search for a couple of keywords for a new service we are making. We (I) got obsessed during the lockdowns and went totally overboard with the efforts and budgets. 
We didn't achieve to do things on a shoestring budget, but ended up with incredibly good models.


 


Speech recognition on a shoestring budget

Jöran Vinzens

For some parts of our setup we stuck with asterisk 11. It is out so Security Fixes for far too long, it uses Libs far too old. How can we get this into modern Hardware, modern Kernel, modern Security?
We found a way....

Using Asterisk for a very long time, we are using an interface not present in current version in the way we need it. We ignored it for too long and now we got stuck with it until we finished the new setup. There are several problems using that old software.
On modern Linux the old libs needed to run Asterisk 11 are not supported anymore and new hardware do not support old Linux.
But there is a way out of it.
We figured out a way using docker. But with all that legacy this turns out quite ugly.
This is a good "bad" example how not to ignore your systems. 

Now we know better but it has been a hard way. 

Deal with legacy, the ugly way

Kwindla Kramer

Presenting Daily, one of our platinum sponsors.

Platinum Sponsor: Daily

Lorenzo Emilitri

Elixir is a language based on Erlang/OTP, a platform designed to support real-time distribuited telephony applications with zero downtime. It sounds like a perfect match for a real-time event processing application based on telephony data received from MS Teams. Or is it? Let's find out.

The Erlang/OTP platform was originally created as a control plane for Ericcson's own telephony switches, and was therefore designed from the start to address the kind of problems found with telephony and real-time communications - real-time responses, distribution, fault tolerance, and very high availability. It is now experiencing a second youth with the advent of Elixir, a language based on on the Erlang VM that offers a better, Ruby-like programming experience, while retaining the strengths of the underlying platform. 

My company is developing a API bridge that can translate Microsoft Teams telephony information into a feed compatible with our  QueueMetrics Live platform. When tasked with developing it, we soon realized that we needed a complex and strongly asynchronous API event processing system, and we took the risk to develop the current prototype in Elixir as an alternative to our currente generation of  Clojure and Kafka-based large-scale stream processors. 

This project has taught us some lessons on places where Elixir fares better than expected, and also some lessons on where it was not as good.


Using Elixir for complex, real-time telephony processing: some lessons learnt

Lorenzo Mangani

The WebRTC ecosystem deserves a proper troubleshooting swiss-army tool designed to slice and dice data at the pace of modern real-time communications without requiring a data science background or an army of consultants. Let's make that happen with the power of cLoki and Clickhouse!

It's like Loki, but for Clickhouse! cLoki is an open-source LogQL API natively compatible with Grafana and designed for ingesting, processing and flexible querying unstructured data such as logs and JSON events from servers and browsers. In this presentation we'll showcase the cutting edge WebRTC Monitoring and Troubleshooting opportunities this technology offers both from a User/Engineer and Integrator perspective, leveraging the raw power of Clickhouse, the fastest open-source columnar database for real-time analytics. The future is here and its more Open-Source than ever.

WebRTC Troubleshooting for Humans with cLoki & Clickhouse

Lorenzo Miniero

WHIP is an exciting new way to use WebRTC for broadcasting ingest. It won't be of much help until it's integrated in widespread broadcasting tools, though. This talk will share some info on how to use OBS to produce media, and use NDI to feed a WHIP client to broadcast it online via Janus/WebRTC.

The broadcasting industry has some staples it typically relies on: almost everyone uses the same protocols to upload (RTMP) and distribute (HLS) the streams they produce, and they often rely on well known tools (like OBS, XSplit or others) to help for the job. While effective, these technologies have some limitations when it comes to lower the distribution latency. For different reasons, though, WebRTC is not that widespread in that community, especially when you want to use WebRTC rather than HLS to distribute your produced media in real-time instead. WHIP, a new standard signalling protocol for broadcasting ingest via WebRTC, is trying to address exactly this missing step, in order to make it easier to use WebRTC for the purpose.

That said, standardizing WHIP is only a first step. Considering how accustomed streamers are to their tools, WebRTC won't really be an option until a way to use WHIP from those applications is made available as well. Taking into account that a native integration of WHIP and WebRTC in those tools may take some time, until then it may be worth to experiment with looser integrations instead. Within that context, when it comes to media production NDI is a de-facto standard in the broadcasting industry, as it makes it easy to exchange uncompressed streams within a LAN to use as media sources. This talk will indeed try to combine those technologies together, and share a few practical information and details about some actual experiments to produce media in OBS, output it via NDI, and then broadcast the result via WebRTC thanks to WHIP and the open source Janus WebRTC Server.

WHIP, NDI and Janus: genesis of a broadcasting demo

Mathis Engelbart

Interactive real-time media transport requires specific congestion control algorithms that pose different requirements and challenges from the ones known from TCP. I present what I learned from implementing and testing bandwidth estimation for Pion in Go.

The internet offers an unreliable, best-effort packet transport, where delays,
bandwidths, and loss rates can vary at any time. How is it possible to send
reliable and high-quality real-time media over a network with zero guarantees?

Real-time applications usually use UDP as the underlying transport to avoid
head-of-line blocking and late retransmissions. UDP does not provide any
congestion control, and the congestion control algorithms known from TCP do not
satisfy the requirements of interactive real-time media flows. Most TCP
congestion control algorithms work well for reliable transfers, e.g., files or
web pages, but impose too much packet loss and delays that are too long for
real-time media transmission.

In real-time communication, peers must send data at a valuable bit rate while
keeping transit delays and packet loss low. At the same time, it has to be fair
to other flows on the same link, be it real-time traffic or bulk transfers.

Learn about my journey in building a bandwidth estimator in Go. I will share
fundamental concepts that make it possible to implement congestion control for
real-time media and the software needed to test it.

Bandwidth Estimation from Scratch: Achieving optimal bandwidth utilization and what I learned along the way.

Matt Warman (MP)

Matthew Hodgson

Element's E2EE voice & video calling makes secure VoIP available on an open, federated platform. We're now building group calling into the Matrix protocol to make encrypted multiparty calling available for everyone to develop and integrate against with a protocol that's free to everyone.

Element's end-to-end encrypted voice and video calling makes secure VoIP availble on an open, federated platform.
Hear how we're extending this to build group calling with the same security into the Matrix protocol to make encrypted multiparty calling available not just to end users, but also for others to develop and integrate against with a protocol that's free to everyone.

Extending Matrix’s E2EE calls to multiparty

Mira Georgieva

Neil Brown

In 20 minutes (to give plenty of time for questions), Neil will cram in as many "legal update" themes as he can, focussing on things relevant to developers and providers of WebRTC services. The focus will be English law, but some of the themes may be relevant more broadly.

# What?

A legal update. From a lawyer. About topics relevant to providers and developers of WebRTC stuff. Without a large bill.

# Why?

Because the more informed you are about the legal environment, the more pitfalls you can avoid, and the better you can run your business.

# Who?

Neil is the managing director of English law firm, [decode.legal](https://decoded.legal). He specialises in Internet and telecoms law, and works with a wide range of real-time communications providers.


WebRTC and the (mostly English) law: an update for developers and providers 

Oleg Agafonov

This presentation is going to be a `SIP3 Features Overview and Usage Guide`. 
Despite on the SIP3 focus the presentation will be built around common practices and modern approaches you should follow to build the most efficient RTC monitoring and troubleshooting platform.

This presentation is going to be a SIP3 Features Overview and Usage Guide. 
Despite on the SIP3 focus the presentation will be built around common practices and modern approaches you should follow to build the most efficient RTC monitoring and troubleshooting platform.

SIP3: New generation of VoIP monitoring

Phil Cluff

Presenting Mux, one of our platinum sponsors.

Platinum Sponsor: Mux

Ravindhran Sankar (Ravi)

Everyone has experienced video call issues over the past 20months - robotic voice, video glitches, low quality video.. What is actually going on? 
In this talk, we'll intentionally deteriorate a call and observe how these issues appear to the user and explore how they manifest in WebRTC metrics.

Debugging WebRTC call issues seems like a dark art. Everyone experiences video call issues, but few people can pinpoint what causes them. With dozens to thousands of call participants distributed across different geographies, environments, machines, and platforms, the range of possible reported issues is infinitely complex and varied. Whether you are a developer integrating real-time video experiences in your product or a user who simply wants to join video calls without interruptions - what are the things to think about & what can you do about them?

At Daily, we have actively debugged 1000s of call issues. We have learned that a small number of constraints are responsible for the majority of user frustrations.

In this talk, we'll play with a video call. We'll deteriorate a call on purpose. We will recreate some of the common issues users experience every day, understand why they happen and how they manifest themselves in WebRTC metrics. 

Hopefully, this talk will give the casual video call user some things to try and the video developer tools to debug these common issues.

The dark arts of debugging WebRTC calls

Finally, machine learning models have made it to the browser. Virtual backgrounds and blurs are everywhere!

In this talk, let's explore how a simple background blur works, how you can code your own blur for a WebRTC call, and most interestingly - what more can we build from here?

Machine learning models and inferences are being run on the browser and have become very performant. Libraries and frameworks like mediapipe, bodypix have made it easy to integrate ML-based experiences into a WebRTC call with relative ease. We're still in the early days of discovering their possibilities in web applications.

In this talk, we will walk through how Daily integrated mediapipe in its WebRTC library. We will explore:

1. How segmentation models, like the ones used blur backgrounds, work.
2. How one can integrate an ML segmentation library (like mediapipe) in a WebRTC video call.
3. What other sorts of custom video applications and experiences one can build using a tech stack like these.

Hopefully, this leaves the audience with ideas for writing custom video processors that might enable the next generation of video experiences on WebRTC.

Video backgrounds are becoming blurry, but the future is clear: more machine learning models in WebRTC applications 

Rick Lewis

Presenting Element, one of our platinum sponsors.

Platinum Sponsor: Element

Rob Pickering

Half hackathon, half holiday, making plans for a pilot 4-week FOSS oriented summer hackathon in 2022.

For at least the next couple of years, I have access to a 10-acre field in the heart of the English Lake District with power, water, and decent WiFi. The plan is to use the space to run a small extended hack camp where groups of folks can come and for around a week or so over a 4-week period. The objective is to have fun with tech and explore solutions to real world problems, whilst also spending time enjoying the wider environment, walking on hills, cycling round forests, paddling on lakes. It won't be commercial, all the (minimal) onsite facilities will be free in the pilot year and the only conditions will be sign up to a CoC, and that projects developed or contributed to are FOSS. For 2022, I'd like to test the idea with a very small scale invite only pilot of folks from within the community, maybe HWM of 10 or so people concurrently spread over a 4-week period. Objective is to work out if this idea is any good: do holidays and hackathons even mix?  If it works, figure out the best and worst bits to run something that generates real value for multiple communities in 2023. Objective of this session is to introduce the idea and test the water.

Hack The Lakes

Ruchir Brahmbhatt

As a software outsourcing company with nearly 15 years of operating experience and 200 employees, Ecosmob works with a wide range of clients and technology stacks.  In this session one our technical subject matter experts will share their thoughts on what 5G means for mobile app development.

5G networks are poised to change many realities for mobile users and networks. How will they impact expectations for voice and video applications and in turn how they are created? More and more, real time communications, and maybe even metaverse experiences, will be seen as features in apps, rather than services in and of themselves.   

Creating these experiences can be accomplished using many tools and frameworks.  In this session, Ecosmob’s experienced developers will report on some of their experiences and share tips from the field on hardware and network realities as well as security best practices. 


Building voice and video mobile apps for a 5G world

Răzvan Crainea

Find out how a SIP proxy can provide RTP high availability for calls that use media relay servers (such as RTPProxy or RTPEngine) using standard SIP capabilities, using a simple technique that re-anchors the ongoing call’s media to a different node/engine.

The new RTP Relay module in OpenSIPS 3.2 provides an easy interface to engage arbitrary RTP media proxies in a call, as well as an easy method of failing over in the middle of the call to a different node if the initial ones fails. This presentation shows how this can be achieve using OpenSIPS 3.2 and plain SIP, through a novel technique called media re-anchoring. The talk presents the new module as well as how and why we had to develop it.

Media high availability/re-anchoring using OpenSIPS 3.2

Saúl Ibarra Corretgé

Jitsi Meet has been powering meetings for millions of users already, but we had limited the meeting size to a maximum of 100 users.

We heard our users loud and clear, they wanted larger meetings! So we rolled up our sleeves and got to work.

In this presentation we'll see how we did it.

Supporting large meetings is no easy feat. It requires adjusting almost every layer of the stack, from the frontend, to provide a good user interface and user experience, to the backend, in order to be able to cope with so many participants in a single meeting.

In this presentation we'll go through all of it. How we got started, the tools that we had to build in order to test it (yeah we even had to do that!) and how we finally accomplished that magical number. Oh, and we'll look at mobile too of course!

Reaching new heights: how we added support for 500 participant meetings to Jitsi Meet

Thibault Martin

Matrix is an open protocol, often used for instant messaging. Its federated nature means there needs to be proper safeguards to make sure it remains a friendly and welcoming place.

Come learn the basics of Matrix for Instant Messaging, how some open communities are using it, and how you too can!

This talk walks people through the basic concepts of Matrix from the perspective of a user and a community manager. Then it gives some example of open communities who successfully started to rely on Matrix for their instant messaging, and concludes with advice on how people can open their community to Matrix too.

Matrix 101

Tim Panton

As part of an IETF hackathon I decided to see if I could build a new open source WebRTC implementation.

WebRTC is just a pile of pre-existing protocols, all of which have multiple existing open source implementations.

I'd just glue them together, right?

How hard could it be?

This talk will introduce whipie - an open source WHIP client for the raspberry pi.

I'll describe the WHIP protocol - which is intended to replace RTMP as a way of doing low latency video 'upload' for sites such as Twitch, Youtube etc. using a constrained version of webRTC.

The audience will learn about how the component protocols of webRTC (ICE, STUN, DTLS, SRTP, RTCP)  fit together and perhaps a bit of software engineering.

I'll also talk about my motivations for doing the project and why I didn't use any of the existing webRTC stacks. 


 

Writing a new open source webRTC stack in a week. 

Vanessa Pyne

Varun Singh

Will Hunt

Matrix is interoperable by nature. The servers are federated, like e-mails, but bridges allow conversation with third parties as well! 

We'll walk through the main bridges Matrix has, how they work, why you should write one, and what the future holds!

The talk is structured as: 

* What is Matrix?

* Bridges, what are they, why should YOU write a bridge?

* Demo Time: let's pop a twilio bridge in 5 minutes

* Success stories

* What's new and what's next.

Speakers

Alessandro Toppi

Alex Vlahopoulos

Amandine Le Pape

Andreas Granig

Antony Jukes

Artur Shellunts

Ashley Blewer

Christian Stuff

CommCon

Dan Jenkins

Darren Cubitt

Dave Horton

David Duffett

Fred Posner

Gavin Henry

Joachim Vanheuverzwijn

Jöran Vinzens

Kwindla Kramer

Lorenzo Emilitri

Lorenzo Mangani

Lorenzo Miniero

Mathis Engelbart

Matt Warman (MP)

Matthew Hodgson

Mira Georgieva

Neil Brown

Oleg Agafonov

Phil Cluff

Ravindhran Sankar (Ravi)

Rick Lewis

Rob Pickering

Ruchir Brahmbhatt

Răzvan Crainea

Saúl Ibarra Corretgé

Thibault Martin

Tim Panton

Vanessa Pyne

Varun Singh

Will Hunt

William King

MC

Dan Jenkins