Design Clinics are informal whiteboard discussions that give you the opportunity to explain your situation in detail to a matter Expert,
It is your chance to talk regarding your project or application and receive their opinions and perspectives on the best strategy to your
concerns and suggestions for solutions you may not have considered - Get full details and availability here or book a session below.

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Event schedule 2016

The Amsterdam Summit is confirmed for May 2016!
The detailed schedule is available below.
Join the OpenSIPS user list to receive updates.

Check back soon for schedule updates
Clinics Day details are available here

Extra Activities

Event schedule and extra activities will be coordinated Live using the conference channels for voting and more! Let our Bots help you get around and make the best of the experience.

Slack #OpenSIPS

Event Archive

All keynote talks and presentations details with PDFs and Videos from all past editions are available online for the community to use and enjoy.

#Youtube Channel

Radisson Blu Hotel, "Hollandia" Lobby - Rusland 17 (Center) 1012CK Amsterdam, NL
Guest Registartion & Badge Pick-Up
Complemetary Coffee & Tea
OpenSIPS 2.2, a better SIP experience. The 2.2 release follows the 2.x line in terms of expanding its async capabilities (RADIUS, LDAP support), but with a strong focus on native clustering capabilities, for automatic sharing and replicating data between OpenSIPS instances.
In OpenSIPS 2.2, the whole SIP capturing process was dramatically refactored, to make it more flexible, higher performance and more powerful. The re-work covers the whole SIP capturing flow (filtering, HEP, DB). Below the floating line several additions like WSS support, enhanced Events handling and SQLite support make OpenSIPS a better platform for reliable services, with additional help from high-end features like SQL caching, end-peer monitoring or Accounting functions.

Presented by Bogdan-Andrei Iancu OpenSIPS Project
The SIPCAPTURE stack has greatly evolved over the last years migrating from a SIP-only application into a dynamic, modular and fully programmable RTC "framework" for capturing, monitoring and troubleshooting real-time protocols. Learn how to integrate and correlate SIP signaling, webRTC sessions, application logs, media reports and custom metrics using the latest modules, components and capture agents from the HEP/EEP ecosystem, and how get your own capture stack up and running in minutes using Docker and native HEP/EEP integrations

Presented by Alexandr Dubovikov & Lorenzo Mangani QXIP BV SIPCAPTURE Project
Different options, techniques and strategies to implement load balancing and high availability for FreeSWITCH. How to implement each one using best practices.

Presented by Giovanni Maruzzelli OpenTelecom.IT
Lunch & Food Costs are NOT included in the Ticket price.
Various locations available in the city center.
An exploration into what the new WebSockets module within OpenSIPS means for end users and a brief example of how to get up and running making browser to browser calls and browser to PSTN calls.

Presented by Pete Kelly Sourcevox Ltd
This workshop describes how you can use the new OpenSIPS 2.2 to share dialog information, ratelimit counters and user location information across multiple OpenSIPS nodes using the new clusterer module. It contains hands-on use cases, configuration and provisioning tips for replicating data to multiple nodes in the most efficient and scalable method using the latest OpenSIPS release.
Presented by Razvan Crainea OpenSIPS Project
Complementary Coffee and Tea available in the Conference Lobby
In this presentation you will learn how to integrate OpenSIPS with ElasticSearch to index logs, cdrs and fraud detection information

Presented by Flávio de Andrade Gonçalves SipPulse
This presentation will examine the topic in two parts - how the OpenSIPS server fits within the classic network design model and how it can be hardened against various types of attack. To put the topic in context, John will briefly examine the mechanisms of the most prevalent type of attack seen on the Internet - those characterised by the so-called "sipvicious" SIP request probes and generally referred to as 'International Revenue Share Fraud'. John comes to this topic as an OpenSIPS integrator with a wide range of experience gained over many years and working with numerous Internet Telephony Service Providers. He will focus on practical advice, using relatively simple techniques, that other OpenSIPS solution developers can take with them and easily build into their code. This starts by looking at best practice for integration of OpenSIPS within your network architecture then moves on to discuss identification and handling of malicious SIP requests, logging, fail2ban integration and traffic analysis using concurrent call counting and the Fraud Detection module.

Prensented by John Quick SmartVox Limited

BBQ Farm Chicken & Fries + Beers

Rokin 75
1012 KL Amsterdam

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 Charging. In this talk Dan will introduce the LCR component of CGRateS, adding dynamic behaviour to LCR rules through internal mechanisms like real-time QoS metrics or Account Thresholds with automatic Triggers manipulating routing table. CGRateS is a battle-tested, real-time Charging Engine which supports 4 different charging modes: Prepaid, Pseudo-prepaid, Postpaid and Rated.

Presented by Dan Christian Bogos ITSysCom GmbH
Presentation and workshop offering insights into the generic event reporting interface built into OpenSIPS and all possible backends it may interact with, starting with OpenSIPS 2.2.

Presented by Liviu Chircu OpenSIPS Project
Complementary Coffee and Tea available in the Conference Lobby
Maintaining accurate and realtime cdrs within your class 5 infrastructure can be tricky. Although opensips can provide you a class 4 cdr with the start and end of a call, tracking what happens to a call within the class 5 environment becomes cumbersome when trying to unify with the opensips acc. Using rabbitmq, opensips, and freeswitch we'll look at methods to track calls and unify the class 4 and class 5 cdrs. If time permits, we'll discuss other solutions to using rabbitmq to make realtime data available to your web gui.

Presented by Alex Goulis RateTel
This workshop will present the complete set of steps and information to capture SIP traffic with OpenSIPS and HOMER. Going through the OpenSIPS configuration file we will show the modules, the parameters and necessary scripting to implement the whole flow of traffic capturing:
- filtering and packing traffic into hep protocol on the SIP server side;
- routing HEP traffic and saving it to database on the capturing node;
- visualizing the SIP traffic in HOMER web interface;

Presented by Ionut Ionita OpenSIPS Project
Lunch & Food Costs are NOT included in the Ticket price.
Various locations available in the city center.
AG Projects has been operating a free service for years at This service uses our SIPThor technology for horizontal scalability. We'll take a tour on different scalability models and on how SIPThor operates to achieve horizontal scalability which meets our customer's needs.

Presented by Saúl Ibarra Corretgé AG Projects
Janus is an open source, general purpose, WebRTC gateway. Its modular and extensible nature allows it to be used for different use cases involving real-time multimedia streams, ranging from more traditional applications like web conferencing, streaming, SIP gatewaying and contact centers, to more innovative scenarios like social TV, surveillance systems, home automation and more. Thanks to its lightweight implementation, it can be deployed even on systems with limited resources. It has an active community and is currently being used in several open source and commercial projects.

Presented by Lorenzo Minero Meetecho
Complementary Coffee and Tea available in the Conference Lobby
In my session i will show you how to set up a graphing environment in which you can easily spot the health of all your opensips boxes. The graphing will be handled by grafana and the storage backend is influxdb. I will show how to store opensips statistics and make them look good. When this is running, I will put some load on the boxes and show how this influences the different graphs.

Presented by Rik Broers Motto Communications
Be IP develops and distributes full-featured IP telephony and unified communication solutions to companies and public institutions. OpenSIPS is a core component of the solution. We will present possible architectures, but also what features are implemented using the OpenSIPS software, as well as future developments were are working on.

Presented by Damien Sandras BeIP
Food & Beverage costs are NOT included in the Ticket price.
Various locations available in the city center.
Clinics and Workshops
Book your Clinics / Design Session
Lunch & Food Costs are NOT included in the Ticket price.
Various locations available in the city center.
Clinics and Workshops
Book your Clinics / Design Session
Live VUC Session, hosted by Randy Resnick
featuring Bogdan-Andrei Iancu and the OpenSIPS Core Team
VoIP Users Conference (VUC) Community
Bogdan is the OpenSIPS project founder with an experience of 15+ year in the SIP world. Practicing the symbioses between managing the Open Source project and building commercial products around OpenSIPS, gives the best results in producing a viable SIP Server software for the read-life needs.
OpenSIPS developer for 4 years, have both developed a series of modules (Sangoma transcoding, REST client, math operations) and done extensive troubleshooting / improvements in both the project's critical areas of code (i.e. SIP transactions, SIP dialogs, TCP and UDP processing and scaling) and its scripting language. Also experienced with troubleshooting SIP / VoIP and glue scripting in bash 4.0+ and Python 2.
Bachelor's degree in Computer Science at Politehnica University of Bucharest, almost two years experience in SIP and OpenSIPS, ongoing master in Complex Network Security
OpenSIPS maintainer and developer, involved in both design and development of new modules, as well as core functions
Alex is an owner at Ratetel, Inc, a wholesale and hosted pbx provider in Houston, TX. He was the first OpenSIPS certified professional and has since been very active in the project. He also is a sales engineer for OpenSIPS Solutions.
Peter has over a decade of experience working with OpenSIPS and open source VoIP implementations in the UK, Europe and the United States. As a VoIP consultant for Sourcevox he spends his time advising carriers and operators on how they can make best use of OpenSIPS within their platforms.
Flavio E. Goncalves is the author of two books, Building Telephony Systems with OpenSIPS and Configuration Guide for Asterisk PBX. He is also one of the first Digium Certified Asterisk Professionals, achieved in May 2006. He is currently CEO of SipPulse Routing and Billing Solutions for SIP, a company that specializes in SIP softswitches.
Giovanni Maruzzelli (OpenTelecom.IT) is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. He's a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC, Kamailio and OpenSIPS. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies - Then supervisor of Internet operations and architect of the first engine for paid access to , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank - IFC. Since 2005 he's based in Italy, and serves ICT and Telco companies worldwide.
He is employed as Senior Voice Engineer for QSC AG, one of the major German voice and data providers. Alexandr holds a diploma in physics of Odessa State University. He has 20 years of experience in telecommunication techniques and has contributed to many OpenSource projects like FeeeSwitch, SER, Kamailio, SEMS, Asterisk, SIPP, Wireshark. Alexandr is the main developer of Homer SIP Capture project. He is also founder of IRC RusNet Network, one of the biggest national IRC networks in the world.
He is founder of Amsterdam based QXIP BV, Co-Founder and Developer of HOMER / SIPCAPTURE Project and voice specialist of the NTOP Team. Formerly a Sound Engineer, Lorenzo has been deeply involved with telecommunications and VoIP for well over a decade and has contributed ideas, design concepts and code to many voice-related Open-Source and commercial projects specializing in active and passive monitoring solutions with his team. Currently he is Sr. Voice Engineer and Designer for the largest international cable operator worldwide.
It was mid 2005 when a good friend told me "Hey come and see this! It's called Asterisk and you can make calls with it! It uses a protocol called SIP and..." Since then I've been working on VoIP and liking it more every day. I also love IM, presence and gadgets... everything that I can make 'talk'
He is the founder of ITsysCOM, experienced communications architect and VoIP specialist. Dan is a double graduate of Politechnica University, Timisoara, with post-graduate specialization in Communication Protocols and Software Development. For the past couple of years, he has focused mainly on Cloud Computing Technologies interoperability, subject of his PhD thesis research. A frequent and well-known contributor to the Open Source community, most noticeably being the co-founder of CGRateS Project, Dan is a firm believer in merging the very best production-ready software to create high-quality, scalable and cost-effective communications solutions.
He is the chairman and co-founder of Meetecho, a company providing both consultancy services and communication platforms. He got is degree and Ph.D at the Computer Science Department at the University of Napoli Federico II, where he started working on multimedia conferencing and met the colleagues with whom he co-founded Meetecho as an academic spin-off. He is an active contributor to the Internet Engineering Task Force (IETF) standardization activities, especially in the framework of real-time multimedia applications. He is most known as the author of the Janus WebRTC Gateway, an open source WebRTC server-side implementation.
John works as a freelance VoIP consultant specialising in OpenSIPS based solutions. He has worked in the voice and telecoms industry for more than 25 years. He established his own company, Smartvox Limited, in 2004 with a focus on open source VoIP solutions and, since then, has worked with around 40 different Internet Telephony Service Providers on a wide range of projects.
He is a 25 year young voice engineer for Motto Communications, One of the biggest voip providers in the netherlands. For college he did Network Infrastructure Design, where also the first love for linux operating systems was founded.
Damien Sandras is the creator and developer of the Ekiga VoIP and videoconferencing software. Apart from this, he participated to the creation of FOSDEM (Free and Open Source Developers’ European Meeting) and co-organized the event during the first 7 editions.


Registration to the conference is just $199 (per person)

or just $149 if you have a discount code or are
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Event Location

OpenSIPS Summit 2016

Radisson Blu Hotel
Rusland 17 (Center)
1012CK Amsterdam, NL
Available Hotels