Please note that there can still be updates to the timelines of the schedule. Details about accepted speakers are presented in a dedicated page.


Day 1: April 16, 09:00 – 18:00


09:00 – 09:05 – Welcome!

09:05 – 09:15 – Kamailio – State of the Project

Speaker: Daniel-Constantin Mierla (Asipto, co-founder of Kamailio)

Description: the summary of what happened recently in the development and the environment around Kamailio project, its newest features and use cases. An overview of the development and management process, how you can get involved and help the project evolve the way you would like.

09:15 – 09:25 – Kamailio – Business Report

Speaker: Elena-Ramona Modroiu (Asipto, co-founder of Kamailio)

Description: how and where is Kamailio used, what are the successful business models and the trends to watch for future communication systems, statistics on performances and unusual use cases

09:30 – 10:00 – The future of NG RTC platforms and services

Speaker: Thomas Magedanz, Prof. Dr. (Professor at Technische Universität Berlin and Head of NGNI Competence Center at FhG Fokus)

Description: FhG Fokus is one of the leading world wide research institutes that influenced the evolutions of real time communications. The place where Kamailio was started as SIP Express Router project back in 2001, Fokus continued since then to deliver a stream of innovations in the fields of IP telephony and machine to machine communications. The talk gives an overview of current research directions that can impact the way we communicate tomorrow.

10:05 – 10:35 – Multimodal Real Time Communications – the impact of becoming omnipresent and continuous

Speaker: Alan Duric (Co-founder Telio Holding ASA)

Description: outlook and analysis of the trends in (Real Time) Communications industry – impact on the existing business models – role of OSS (Open Source Software) and Kamailio in shaping the RTC landscape

10:35 – 11:00 – Coffee Break


11:00 – 11:30 – WebRTC and VoIP: bridging the gap

Speaker: Victor Pascual Avila (Technology Strategist)

Description: A look at whether it is possible and how to blend classic VoIP with WebRTC, what are the common elements and incompatibilities between the two technologies.

11:35 – 12:05 – Start Your Own Telephony Service with SIP:Provider

Speaker: Andreas Granig (CTO of Sipwise)

Description: how to become a telephony service provider in less than one hour using SIP:Provider platform. Bundled as out of the box system for running a VoIP operator service, SIP:Provider offers everything needed to take care of security, scalability, media services and billing. Built using many open source applications, wrapped by web administration and customer portals, SIP:Provider gives the valuable flexibility, pretty unique in the telephony market, to add over the top new features without vendor lock.

12:10 – 12:40 – Asterisk Update

Speaker: David Duffet (Director, Asterisk Worldwide Community)

Description: Although Asterisk has now been around for more than 12 years, Digium and the Community continue to refine and optimise it, and add support for new features and technologies. This is what keeps Asterisk as the most popular and well established open source communications platform in the world. Attend this session to see some of the latest additions to Asterisk, and to hear about what’s coming down the line in Asterisk 12.

12:40 – 13:40 – Lunch Break


13:40 – 14:10 – Large IP Telephony Deployment Scenarios in Real World

Speaker: James Body (Head of research and development at Truphone)

Description: how to get most of Kamailio’s scalability, flexibility and reliability and its role in building innovative real time communication services. A trip through an over decade of deployments to disturb the telecom services.

14:15 – 14:45 – SIP and MSRP over WebSocket in Kamailio

Speaker: Peter Dunkley (CTO of Crocodile RCS)

Description: bridging the present and the future of communications using SIP – call your contacts from browser via websockets, text chatting, advertising presence states and transferring files, at the same time, using same device.

14:50 – 15:20 – Deploying IMS Platforms

Speaker: Carsten Bock (CEO of NG Voice)

Description: experiences while deploying IMS and the benefits brought by relying on Kamailio and open source for building flexible IMS platforms. Not many took the challenges of developing IMS extensions, even fewer could keep it going, this talk comes to present that IMS implementation is mature, with key deployments world wide, at a moment when this technology has to become the core of 4G+ networks.

15:20 – 15:50 Coffee Break


15:50 – 16:20 – Embedding Kamailio into BESIP and OpenWrt

Speaker: Lukas Macura (Lecturer at Silesian University in Opava, VoIP and Network Specialist at CESNET)

Description: The aim of the BESIP project is the development and implementation of open source based embedded SIP communication server with an easy integration into the computer network. BESIP should be able to act as a small PBX, edge proxy or SBC. We used Kamailio as core SIP router and our goal is to configure it using NETCONF protocol. We ported several packages into OpenWrt (e.g., Kamailio3, Asterisk11) and we prepared experimental feeds (e.g., kamailio4, mediaproxy), which are waiting to be accepted by community. Presentation will be focused to explain basic goals of BESIP project and interaction with Kamailio.

16:25 – 16:50 – Roll Your Own VoIP Cloud in 30 Minutes or less

Speaker: James Aimonetti (Distributed Systems Architect at 2600hz)

Description: Discussion of the 2600hz implementation of Kamailio within the Kazoo stack, the features of the platform, how to get it up and running quickly and mange it in production.

16:55 – 18:00 – VUC RTC and RCS Panel – Open Discussions

Moderator: Randy Resnick (Producer of VoIP Users Conference)

Participants: James Body, Olle E. Johansson, Simon Woodhead, Thomas Peterseil, Paul Fermedge, Alex Kinch — actually everyone in the room is invited to join the discussion

Description: debate about present and future of real time communication technologies and services

18:00 – 18:15 – Closing Session


Evening – Cocktail and Social Networking Party

Time and Location: 18:30 – 22:00 in the Atrium Area inside the conference building.

Day 2: April 17, 09:00 – 18:00

09:00 – 09:05 – Welcome!

09:05 – 09:25 – SEMS – Adaptive Session Border Controller

Speaker: Vladimir Broz (Director of Engineering at Frafos)

Description: Frafos, the company behind SIP Express Media Server (SEMS), created the ABC SBC product on top of the open source media server. The SBC primarily aims to enable standard-based SIP communications with advanced business logic, allowing programmers and integrators to build telephony applications rapidly, avoiding to load their apps with abundant protocol details.

09:30 – 10:00 – Packet Capture for Security Risk Prevention in Telephony

Speaker: Alexandr Dubovikov (Founder of Homer Project)

Description: how to capture and aggregate signaling traffic from several SIP server nodes and generate reports for technical (troubleshooting) and business (marketing) needs. As the business of small and big telephony operators grows, so does the risk of fraud and sophisticated security attacks against their infrastructure – CDRs are no longer a reliable source for timely fraud detection and revenue protection. Leveraging centralized packet capture and parsing (based on Kamailio and award-winning HOMER and its powerful API) simple and complex scripts can be programmed to detect and react to fraud events and other kinds of attacks in real-time. Actions can be as simple as an email, or as complex as re-configuring routing, banning and spoofing back crafted responses to the attackers by analyzing and profiling the telephony traffic.

10:05 – 10:35 – Jitsi – the softphone for rich communications

Speaker: Emil Ivov (Founder of Jitsi Project)

Description: During the past ten years Jitsi has traveled a long way from a proof of concept call application to arguably the most complete and feature rich FOSS communicator on the planet. Emil will walk us through the various flagship features of Jitsi such as high quality audio/video calling, conferencing, call encryption, file transfer, desktop sharing, instant messaging and many others. The talk will approach also Jitsi’s sister projects: libjitsi, a component that allows building communications client or server-side applications for SIP, XMPP and WebRTC; and the newborn Jitsi Videobridge which makes it possible for anyone to deploy a hangout-like video conferencing solution.

10:35 – 11:00 – Coffee Break


11:00 – 11:30 – VoIP Security Tools

Speaker: Anton Roman Portabales (CTO Quobis)

Description: What are the most common attacks and how to test if your network is protected with existing tools. Fraud is still biggest business threat in VoIP, learn how to stay ahead of attackers and prevent being a victim that can ruin yourself.

11:35 – 12:05 – Managing Large Telephony Systems

Speaker: Henning Westerholt (Head of IT Operations Internet Access & Communications at 1&1 Internet AG)

Description: remarks from years of operating a telephony platform serving millions of subscribers and routing billions of voice minutes per month. How open source can leverage large companies to be competitive and keep the pace on the market with innovative services

12:10 – 12:40 – An Analysis of WebRTC Impact on Future Communications

Speaker: Dean Bubley (Founder of Disruptive Analysis)

Description: Status & forecasts for adoption of WebRTC in fixed & mobile – Key early use-cases and market drivers for WebRTC – How WebRTC fits with (or against) SIP – Implications for VoIP developers, telcos & enterprises

12:40 – 13:40 – Lunch Break


13:40 – 14:10 – Case study – Using Kamailio as IN SIP application server at Bezeq

Speaker: Uri Shacked (Senior Architect and Bezeq)

Description: Bezeq is the largest Israeli telephony operator. The IN system at Bezeq carries all the number translation services, including features like business IVR, call screening, busy rerouting, no answer rerouting, music on hold and other features.  Using Kamailio, Bezeq managed to deploy a telco grade NTS and IVR system serving 2 million calls per day. The system is resilient enough to also support 911 and other lifesaving 1800 services.  In this presentation we will demonstrate how Bezeq leveraged Kamailio’s advantages such as the ability to manipulate headers and call flows, store large amount of data in memory, carry load with high performance, stability and fast implementation in order to meet the project’s tight schedule and goals.

14:15 – 14:45 – VoIP with Free Software in Enterprises

Speaker: Markus Lindenberg (consultant at Gonicus)

Description: The fast changing requirements and the integration of telecommunication and information technology infrastructures open up new opportunities for Open-Source-Software to set foot in the telephony infrastructures of large enterprises. This talk will address the experiences in a project to replace a classic PBX by a new VoIP infrastructure within an insurance company. Aside from technologies used (like Kamailio or Asterisk) the presentation will also explain motivation for such a project as well as talk about implementation and operation of an Open-Source-VoIP-solution for more than 3.500 users.

14:50 – 15:20 – WebRTC Business Opportunities

Speaker: Björn Schwarze (CEO of  Junghanns Communications)

Description: WebRTC allows browsers to act as softphones, by activating a piece of Javascript we can contact any other WebRTC user in the world. The talk will look at WebRTC technology from business perspective, identifying what are the new opportunities for rich communication services.

15:20 – 15:50 Coffee Break


15:50 – 16:20 – Practical deployments in the North American PSTN Sphere

Speaker: Alex Balashov (principal of Evariste Systems)

Description: Key considerations of PSTN-oriented deployments in the North American telephony environment, viewed from the standpoint of regulation, standard information services (number portability queries, calling name/directory queries, law enforcement intercept, etc.). Enriched by Evariste’s experience as a nearly 100% Kamailio-oriented shop, the general trends in the demands (of Kamailio) of North American ITSP and CLEC customers will be discussed.

16:25 – 16:55 – CGRateS – Carrier grade real-time charging

Speaker: Dan Christian Bogos (founder of ITsysCOM)

Description: Internet Telephony has evolved over the years from simple office PBX to clustered voice services. CGRateS project was born out of today’s market demand for performance and scalability when it comes to Real-Time rating. In this talk Dan will briefly introduce CGRateS’s particularities together with the current state of the project and features on the road-map.

17:00 – 17:30 Load Balancing Load Balancers

Speaker: Daniel-Constantin Mierla (Asipto, co-founder of Kamailio)

Description: is your subscriber base growing fast? You keep adding new servers behind load balancer. What happens when you reach hardware limits of load balancers? This is a presentation of how to scale the load balancers farm.

17:30 – 18:00 – Panel – Future of Kamailio Project – Open Discussions

Participants: Kamailio Developers

18:00 -18:10 – Closing Session


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