Event Agenda

The OpenSIPS Summit caters to both beginners and advanced users.
You’ll get an intimate look at native clustering support, enhanced SIP capturing capabilities, Quality Based Routing, VoIP security and many more of the tools available to you in the latest OpenSIPS release.

  • Edward

    Day 1
    OPENSIPS SUMMIT

    First day of the OpenSIPS Summit 2017 with presentations, workshops and for the first time this year we are introducing the Round Tables talks.

  • Lucy

    Day 2
    OPENSIPS SUMMIT

    Second day of the OpenSIPS Summit 2017 with more workshops and interesting presentatins from international speakers and great collaborations.

  • Nathan

    Day 3
    Design Clinics

    Design Clinics are one to one conversations with regarding your project or application and receive their opinions and perspectives on best strategies and solutions.

  • Jacob

    Day 4
    Advanced Training

    A full day of Advanced OpenSIPS training with the project developers and leads (registration will open soon and space is limited!)

T.B.D.

Presented by Lorenzo Mangani QXIP/SIPCAPTURE
Exploring the capturing capabilities of OpenSIPS 2.3 in terms of non-SIP traffic with Homer6 - collecting data from transport layer (TLS/WSS), from REST query level, from Management Interface level. A complex set of data, with the ability of correlating all the information around the SIP calls, that proves to be a holly grail when comes to operating and troubleshooting SIP platforms.

Presented by Ionut Ionita OpenSIPS
Unlike other industries, the VoIP and RTC ecosystem lacks an accessible network for threat intelligence, leaving many Operators on their own reinventing the security wheel. It’s time for our Community to secure itself with a real-time trusted, voluntary, federated Pub/Sub Exchange, instantly exposing Attackers, Abusers and new Threats, acting as an open and distributed Lock natively integrated at the core of our favorite stacks.

Presented by Alexandr Dubovikov QXIP/SIPCAPTURE
SIP-I trunks are the cost effective replacements for the traditional SS7 interconnections. And now, OpenSIPS 2.3 helps in integrating the SIP-I trunks with the end-user devices by performing the SIP-I to SIP conversion and ISUP manipulation.

Presented by Razvan Crainea OpenSIPS
We are still using ISUP for our interconnect to several carriers and providers. Unfortunately we depended on log files for debugging. For all of our SIP traffic we use homer for debugging and monitoring so we knew there is a better way. Thanks to the integration of ISUP into homer we now have a single debugging and tracing tool for our developers and customer support. In my talk I'll give an overview of how it looks and how we use it.
Presented by Jöran Vinzens SIPGATE
An interactive guide to configuring the latest version of OpenSIPS to allow you to make WebRTC based calls.
Presented by Peter Kelly SourceVOX
All conference passes include lunch buffet served at the Expo tunnel (downstairs)
Together with it’s 2.3 release, OpenSIPS introduces integration for CGRateS processing directly within OpenSIPS script, thus simplifying a task which for long was considered as “last mile” - billing for SIP proxied requests. In this talk Dan will reveal details regarding integration mechanism as well as discussing advanced use cases available now directly in your OpenSIPS scripts. CGRateS is a battle-tested Online Charging System with support for both prepaid and postpaid billing modes.
Presented by Dan Christian Bogos CGrates
Contact center on Steroids with OpenSIPS The Best Combination For Your Inbound/Outbound Call Center Needs!
    Inbound:
  • Distributed Queue Application – Treat waiting times as marketable opportunities.
  • Real Time Monitoring – Always know how well your call center is doing with a glance.
  • Customer Oriented Call Flow Manager – Send every customer to the right destination!

  • Outbound:
  • Predictive/Productive Dialer – Save time and increase productivity automatically.
  • Dynamic GEO Caller ID With Anti-Spam Mechanism – Make sure calls get through to customers.
  • Quality Based Routing – Get the best connection with minimal signal & quality loss.

  • Flexibility: The Most Flexible Solution For Your Call Center Needs
  • Flexible Topology – create the system that works best for your business.
  • Per User License – Get exactly what you need, when you need it.
  • Be Part Of A Bigger Cloud – Instead of one cloud, be part of our global network, increasing up time and survivability.

  • Voicenter Connect:
    If you need to update your Older IP based PBX but don’t want the trouble of swapping out hardware – Voicenter connect can help bring the features of the future to you.
    All these are part of many other new, exciting and customizable features included in Voicenter services!
    Special offer for the OpenSips Community – 1 month free of Voicenter’s contact center services.

Presented by Shlomi Gutman Voicenter
A brief catch-up of the more recent developments in the Asterisk project, and why Asterisk continues to be the widely used Open Source communications platform in the world.
Presented by David Duffett Digium
Complementary Coffee and Tea available in the Conference Lobby
Migrating a telephone system based on 16 stations Ericsson MD110 to VoIP has not been easy. There are services to deploy, feature to replicate and, above all, be able to make co-operate two networks based on different technologies without creating inefficiencies. I would just tell you how I did it.

Prensented by Michele Pinassi University of Siena
Veteran entrepreneur Suzanne Bowen and veteran developer Bogdan Iancu share POVs, stories and advice on Accessibility, Community, Opportunity, Profitability, and Scalability in relation to communications technology past, present and future. Interactivity among participants will be welcome.

Presented by Suzanne Bowen & Bogdan Andrei Iancu DIDx
T.B.D.

Prensented by Suzanne Bowen
The Round Tables are open discussions around given topics - usually new or interesting subjects . Each tables has its own topic and the discussion is moderated by a person with advance knowledge and experience on the topic. People should join to a Round Table either to share, either to learn the latest news and tricks on the discussed topic. Such discussion are a real melting pot of the real-life experiences.
    Available topics :
  • OpenSIPS as SBC
  • WebRTC
  • SIP Security
  • SIP-I integration
  • Front-ending to Class5 clusters
  • Capturing with Homer
  • CDR handling
Interesting ideas must be backed up by practices - the Interactive Demos are 100% practical lessons, in from of the monitors, on how to installed, configure and run certain setups. Everything is A to Z explained and then shown in realtime, to the audience. Why is it interactive? The participants will have the opportunity to see for real how things are done, they can ask questions and debate aspects related to the demo case.
Radisson Blu Hotel, "Hollandia" Lobby - Rusland 17 (Center) 1012CK Amsterdam, NL
Guest Registartion & Badge Pick-Up
OpenSIPS 2.3 version is built around the integration concept - the OpenSIPS ability to integrate and work together in all possible means with other projects, protocols, systems or concepts. Understanding the integration capabilities of OpenSIPS 2.3 will unveil a powerful tool for building SIP systems.

Presented by Bogdan-Andrei Iancu OpenSIPS Project
How Vonage leverages OpenSIPS to provide superior calling for businesses. Will discuss some of the scaling challenges that have been encountered and the solutions we adopted. Will provide insight into some of the new architectural designs we are considering.

Presented by Norman Brandinger Vonage
This presentation will explore how Telnyx has created a purpose-built network for offering voice services to customers on the internet or those with the ability to form direct connections to the Telnyx backbone. The discussion will include details on building a global backbone, using a cloud agnostic model to build data centers, and adding automation and SDN elements to deliver a quality experience for consumers of the Telnyx product.

Presented by David Casem & Jason Craft Telnyx
Growing your FreeSWITCH platform beyond the single machine: differences between single domain case and multi tenant case.

Presented by Giovanni Maruzzelli OpenTelecom
How to achieve accurate load balancing and optimal resource usage when routing calls over a cluster of FreeSWITCH servers. The key is to get access to realtime feedback from FreeSWTICH to understand the full load context of it, in terms of channels and CPU usage.
Presented by Liviu Circhu OpenSIPS Project
In the realm of real time communications and messaging, what are some of the attributes of a modern software stack? From the perspective of a telecommunications operator and US CLEC, this presentation will cover topics such as containerization, test-driven development, continuous integration plus deployment, logging and introspection, service registry and discovery, and load testing, and how these minimize dev-ops response latencies. Understanding the problems addressed strategically by a modern software stack provides the foundations for building tactical responses to dev-ops challenges.
Presented by William King Flowroute
All conference passes include lunch buffet served at the Expo tunnel (downstairs).
At Jitsi, we believe every video chat should look and sound amazing, between two people or 200. Whether you want to build your own massively multi-user video conference client, or use ours, all our tools are 100% free, open source, and WebRTC compatible. In this presentation we'll dive into how Jitsi Meet works and how we bridged the gap with SIP video devices, in a really innovative way.
Presented by Saúl Ibarra Corretgé Atlassian
Talk about mediasoup, a powerful WebRTC SFU for Node.js. Live demo included.
Presented by Iñaki Baz Castillo
Janus is an open source, general purpose, WebRTC server. It was conceived as a modular architecture, and different plugins can implement a different logic on the WebRTC media it manages. While so far only a single plugin existed to handle SIP gatewaying, based on the Sofia library, two new plugins are in the works to provide the same functionality in different ways and for different needs. This talk will address the different approaches that can be followed to have WebRTC and SIP co-exist, depending on the developers' requirements.
Presented by Lorenzo Miniero Meetecho
Complementary Coffee and Tea available in the Conference Lobby
Talk about progress of our work at Sippy Labs / Sippy Software, announce some new projects, updates about existing ones.

Presented by Maksym Sobolyev Sippy Software
Learn how to create your own call recording solution using opensips and rtpproxy to capture rtp. We'll also use tools like lame and sox to convert recordings for your listening pleasure.

Prensented by Alex Goulis Ratetel
The Round Tables are open discussions around given topics - usually new or interesting subjects . Each tables has its own topic and the discussion is moderated by a person with advance knowledge and experience on the topic. People should join to a Round Table either to share, either to learn the latest news and tricks on the discussed topic. Such discussion are a real melting pot of the real-life experiences.
    Available topics :
  • OpenSIPS as SBC
  • WebRTC
  • SIP Security
  • SIP-I integration
  • Front-ending to Class5 clusters
  • Capturing with Homer
  • CDR handling

BBQ Farm Chicken & Fries + Beers

BierFabriek
Rokin 75
1012 KL Amsterdam

A one day training focused on OpenSIPS frontending for a FreeSWITCH cluster. Starting from a basic configuration, the training will walk you and teach you how to address certain aspects related to frontending a Class5 FS cluster with OpenSIPS:
  • performing mid-registration on OpenSIPS
  • load-balancing calls to FreeSWITCH with realtime load feedback
  • perform fault detection and re-routing for the FreeSWITCH cluster
  • handling special cases as Call Transfer or Conferencing
  • routing outbound calls to carrier (with failover)
  • monitoring and tracing with Homer
Students are required to bring their own laptop - as work material, they will receive a virtual machine holding the setup, ready to be used.
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
Suzanne Bowen is co-founder and once CEO of Super Technologies, Inc. in 1999 that offers the world changemaking DIDX wholesale DID marketplace community, and now co-founder of Suzahdi leather jacket fashions and VP Marketing DIDX. She and the DIDX service are award-winning collaborators according to GetVoIP, Kamailio, TMCnet, Total Telecom, Pulvermedia, VON, Teradata and ComputerWorld.
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.
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 - www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , 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.
He has known the VoIP working in a local telephone company. He started working at the University of Siena, where he made his first Asterisk PBXs. After a few years, he was appointed Head of Telephony in order to migrate the entire structure of the company by Ericsson MD110 telephone to VoIP. Always user of open source software, choose to perform the service with Opensips and Asterisk, also developing the web interface both for the management of the system for users. Currently it is still being migrated, with about 800 users migrate to VoIP.
Norm, with over 30 years of experience building computer hardware and software systems, has architected and delivered advanced communication solutions. The results were new business opportunities, lower OPEX, and more satisfied customers. Norm has been called upon by some of the largest corporations in the world to help manage and integrate their VoIP networks. Norm has also helped startups and mid-sized companies realize their goals of providing advanced communication solutions.
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'
Since 5 years I am working as VoIP Administrator at sipgate, a carrier with focus on VoIP business products in Germany. Since we try to make as much as possible by our selves, we are using mainly open source components like kamailio, asterisk, yate, homer etc. Before my time at sipgate I was involved in the development of Siemens VoIP phones, FMC and trunking equipment as system test engineer.
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.
William is the Chief Architect of Flowroute, responsible for strategic tech direction, tech culture and product architecture. William comes to Flowroute as a long time customer, having built countless inbound and outbound cloud communications services, contact centers and hosted PBX systems integrated with Flowroute APIs. During the 11 years running his consulting company prior to joining Flowroute, William redeveloped architecture for high scale geographically dispersed telecommunication networks for clients including Silent Circle and Portugal Telecom and used his breadth of technical expertise to optimize systems- and human-level processes for businesses in the telecommunications, genomics, high performance computation, and distributed processing industries.
Passionate about new technologies, Open Source, WebRTC, modern Web development, Node.js, C++ and, above all, Real-Time Communications. Projects, experience and talks here: https://inakibaz.me
David works with the Worldwide Asterisk Community for Digium, and is an Asterisk enthusiast in addition to being a Chartered Engineer, globally experienced trainer and public speaker. His experience includes Air Traffic Control communications, Wireless Local Loop, Mobile Networks, Computer Telephony, Voice over IP and Asterisk specifically. In addition to many web articles, David's publications include Asterisk 1.4: The Professional's Guide (Packt, co-author) and the contribution of a chapter (on Internationalisation) to Asterisk: The Definitive Guide (O'Reilly). David is editorially responsible for AstriCon (THE annual global Asterisk event) where he also introduced the 'Fastest Dude to the Dialtone' contest some years ago. He is a frequent speaker at AsterConference Asia (David has also MC'd at this event), IT Expo (East and West) and a number of corporate events. He has also spoken at numerous other conferences - VoIP Developer, Speech World, CT Expo and UC Expo to name a few.
Having over 10 years experience in the SIP world, Pete has helped to build a number of large scale Wholesale, Retail and DID management systems. He now owns Sourcevox Ltd, a company supports and helps businesses to develop and improve their SIP based VoIP systems.
M.Sc. in physics and radio-physics from Kiev University. Long term open-source advocate and contributor to numerous open-source projects including FreeBSD OS, SER, OpenSIPS and Asterisk. Author of few projects of his own, such as RTPProxy, Sippy B2BUA and RTP Cluster.
David Casem is the CEO and co-founder of Telnyx, a wholesale VoIP provider that gives its customers the power to Be Your Own Carrier® through its innovative multi-tenant Mission Control platform and RESTful API. He believes in democratizing the Public Switched Telephone Network through application-centric solutions. Prior to his current role, David founded the Slingshot Consulting Group, specializing in VoIP leveraging Cisco and open source technologies such as Asterisk, SER and Freeswitch. He holds a Bachelor of Arts degree in economics from Swarthmore College.
Jason Craft ( CCIEx2 #37524 - Route/Switch and Service Provider ) is a network engineer focused on architecture, design, and implementation of all things networking at Telnyx. Jason has spent the majority of his career working for financial organizations and service providers with a focus on high availability and low latency. Jason's primary interest today is building traditional IP backbones that blend with NetDevOps and SDN initiatives.

Event schedule 2017

The Amsterdam Summit is confirmed for May 2017!

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.

IRC #OpenSIPS
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

  Advanced Design Clinics
book your seat for extra day

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

Summit Sponsors

If you are interested in sponsoring or participating the event, please contact us or consider a kind Donation.

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Event Location


OpenSIPS Summit 2017

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