At this moment we are working on finalizing the first version of the schedule and we expect to have it ready very soon. For now, next are listed the structure of the event and a selection of workshops and presentations.

Pre-Conference Technical Workshops: May 14, 2018
12:00 ♦ Registration
12:20-12:30 ♦ Welcome Note
12:30-17:20 ♦ Technical Workshops
17:20-17:30 ♦ End of Workshops – Closing Remarks

Conference – Day One: May 15, 2018


08:30-09:00 ♦ Registration
09:00-09:10 ♦ Welcome Note
09:10-12:30 ♦ Conference Presentations
12:30-13:30 ♦ Lunch Break
13:30-18:00 ♦ Conference Presentations
18:00-18:30 ♦ End of Day One – Closing Remarks
19:00-21:00 ♦ Cocktail Party – Social Networking Event

Conference – Day Two: May 16, 2018


09:00-09:10 ♦ Welcome Note
09:10-12:30 ♦ Conference Presentations
12:30-13:30 ♦ Lunch Break
13:30-17:00 ♦ Conference Presentations
17:00-17:30 ♦ End of Event – Closing Remarks

Selection Of Sessions

Pre-Conference Technical Workshops: May 14, 2018
(Workshop) Kamailio For IMS – VoLTE
Carsten Bock, CEO NG Voice, Germany
The IMS extensions are one of the most dynamic parts of Kamailio, evolving a lot year over year. Working in the areas of 4G and VoLTE, wanting to be ready for 5G, planning IoT or IIoT deployments over mobile networks? You can get a lot for your core network and infrastructure with open source and Kamailio.

This workshop goes through the extensions offered by Kamailio for building IMS and VoLTE core platforms, with examples for various use cases, different IMS node types (I-CSCF, P-CSCF, OCS, ISC, S-CSCF, …) and integration with WebRTC.

(Workshop) Kamailio – Ask Me Anything
Victor Seva, VoIP Consultant, Spain
An interactive session allowing the audience to ask any question about using or developing Kamailio. Prepare your questions about scalability, security or anything else you need to build RTC systems with Kamailio.

The panelists will be several prominent Kamailio developers and community members.

(Workshop) Kamailio – Unit Testing Framework With Docker
Giacomo Vacca, Owner RTCSoft, Italy
A simple unit testing framework has been built to help consolidating Kamailio development, detect regressions and integrate with releases management. It should allow developers to create tests for their new features as well as reproduce easily reported issues from the community.

The framework relies on Docker to create an isolated environment for testing as well as known tools for SIP such as sipsak or sipp. It is available at

This workshop presents the unit testing framework, how to add new units and how it can help you test your Kamailio deployment.

(Workshop) Migrating SIP Routing Logic To A KEMI Scripting Language
Daniel-Constantin Mierla, Co-Founder Kamailio, Germany
Attend this tutorial to learn what you should take in consideration when migrating from native scripting language of kamailio.cfg to a KEMI language such as Lua, Python, JavaScript or Squirrel. What are the differences, the benefits and drawback for each of these options, do some hands on sessions showing how changes in SIP routing logic can be reloaded by Kamailio without restarting as well as integrate with external extensions provided by these rich scripting languages.

Conference Days: May 15-16, 2018
Remove any Single Points of Failure in your VoIP Stack – 2600Hz Story
Miriam Libonati, Head of Marketing at 2600Hz, USA
In this session we will deep dive into the inter workings of the SBC, media server and Class 5 relationship and the technical challenges of building a TRUE distributed cluster. In this journey we will discuss how Kamailio is a key component that allow KAZOO to be a truly geo-redundant, distributed infrastructure that removes any single points of failure. Scaling an Open-Source telco switch has never been so easy.

As a fun surprise our lead engineer will be available for live Q&A following the session, so you can learn more about the intricate workings of our infrastructure and why its so unique.

Kamailio With Docker And Kubernetes: Scale In The Right Way
Paolo Visintin, CTO and Co-founder of TimeNet S.r.l, Italy
How to deploy a full containerized solution using Kamailio with Docker orchestrated by Kubernetes.

Topics to be approached:
– containers with public IP network
– htable and other data sharing between different instances of Kamailio
– recover dialogs of dead instances
– failover issues
– caching

The Nexus Between Blockchain And Telephony
Michael Iedema, VoIP/RTC Consultant, Kapsulate, Spain/USA
Blockchain technology solves a fundamental problem in computer science. It allows two parties to establish trust without the need of an intermediary. Blockchain’s validation and distribution protocols establish this trust algorithmically.

There are many intermediaries present in the PSTN. Placing a call between two offices usually involves: the caller’s phone, local PBX, originating provider, terminating provider, remote PBX and finally the callee’s phone.

If trust can now automatically be negotiated between any subset of these parties or elements, how could network architectures change?

In this talk I will provide a technical overview of blockchain and give practical examples of how it could evolve the way we design, implement and deploy telephony services and infrastructure.

Kamailio With Redis Backend
Andreas Granig, CTO and Co-founder Sipwise, Austria
Redis is well know for being fast and reliable key-value storage system. Kamailio introduced a configuration file connector module (ndb_redis) several years ago, allowing to store/retrieve/delete data to/from Redis servers. At the beginning of 2018, Sipwise developed and contributed a database connector module for Redis, named db_redis, enabling instantly support for Redis storage to modules such as usrloc, auth_db, acc, presence, etc. This presentation shows the benefits of using Redis with Kamailio in your VoIP/RTC deployments, the gains in performance and where you should pay attention for getting the best out of the two applications when used together.
Kamailio In The ITSP: The Changing Winds
Alex Balashov, Evariste Systems, USA
The Internet Telephony Service Provider (ITSP) industry, a major consumer of Kamailio, is rapidly embracing cloud architecture and searching for new ways to differentiate products beyond POTS (Plain Old Telephone Service) replacement. This comes with unique challenges for real-time communications because many major cloud service providers are not designed for that from the ground up. In this presentation, some of the technical aspects of this will be explored, as well as the additional infrastructure layers required to make Kamailio work in the brave new world taking shape in the service provider market. Some other changing commercial requirements and ways to address them with Kamailio will also be discussed.
Modular And Test Driven SIP Routing With Lua
Sebastian Damm, Sipgate, Germany
A talk about leveraging Lua (via app_lua module) for flexible SIP routing with Kamailio in a modular way, no longer creating just one big script, but splitting in Lua modules and using the Busted Framework for testing the small functions used in the configuration.

Historically, Kamailio was configured by using core and module commands in kamailio.cfg with a self brewed scripting language. In bigger installations, with extended set of features, this led to huge Kamailio configuration files. app_(lua|python|jsdt) and the KEMI framework brought the possibility to configure your Kamailio in a language of your choice, one of the immediate benefits being the possibility to do test-driven building and maintenance for Kamailio configuration with existing tools, ensuring more stability and reliability before switching to the new configuration in production.

DEC112 – Austrian Text-To-112 Pilot
Wolfgang Kampichler, Frequentis AG, Austria
DEC112 ( funded by netidee ( aims to provide an easy, reliable and secure way for deaf or hearing-impaired people to text for help in an emergency through a simple and intuitive interface and supporting service entities based on open source. A key element, the Emergency Services Routing Proxy, is based on Kamailio and a newly developed LoST (location to service translation) module that allows querying a LoST server to retrieve SIP routing information. The presentation provides insights into the Austrian Text-To-112 Pilot and the newly developed Kamailio module. Further, it gives an update on current European standardization activities, e.g. ETSI SC EMTEL recently started a work item covering a technical specification on NG112 core services and interfaces and finally closes with a live demonstration.
Research And Innovation In RTC With 5G
Thomas Magedanz, Prof. Dr. (Professor at Technische Universität Berlin and Head of NGNI at FhG Fokus Institute), Germany
Fraunhofer Fokus, the place where Kamailio was started as SIP Express Router (SER) project back in 2001, continued for the past two decades to be one of the leading world wide research institutes in real time communications, influencing and shaping the technologies and markets. From bare SIP and VoIP, to IMS, EPC, Fokus continues to explore how to evolve, optimize and scale communications between humans as well as machines. With the reach experience of successful projects at all layers of communication stack, the team of researchers at Fokus is now leveraging SDN, NFV and Cloud to offer a complete 5G playgrounds to industry players. The talk is revealing the current hot topics in research and gives a perspective of what to expect in the near future in the RTC evolution.
Asterisk: Where Is It Going This Year?
Matthew Fredrickson, Manager Of The Asterisk Project, Digium, USA
This is an opportunity for people that use Kamailio in conjunction with Asterisk to get a project level update on what’s happening in Asterisk development – new features being developed, and intended plans for the next Asterisk release. This will help listeners to better plan for the future in how they can better use Asterisk in their suite of telephony infrastructure.
Dynamic SIP Routing And Configuration Management With Consul
Mathias Pasquay, CTO Pascom, Germany
This talk explains how consul can be used as a cluster configuration manager and service discovery tool. Dynamically enable/disable features in Kamailio and other services in your cluster. Use it as a central service registry and fully dynamic dns service. Leverage consul-template to render configuration files for various services. Combine a event based architecture with a stateful key/value store.
Fixing VoIP Quality Over Commodity Connectivity
Simon Woodhead, Owner Simwood eSMS, UK
The ISPs blame the VoIP provider. The VoIP provider blames the ISP. The web developer blames them both because it is “in the cloud”. The end-user hates you all because he can open Google, unless he’s an Enterprise and then it is much safer to just pay the incumbent. Let’s fix this!
CI/CD And TDD In Deploying Kamailio
Aleksandar Sošić, ICT Consultant and Software Engineer, Kinetic, Croatia
A method on how to do continuous integration and deployment of Kamailio instances using test driven development with Jenkins, creating custom testing routes in Kamailio and running multistage test calls with pjsua or sipp.
Another Year Over – What’s Up With IMS, VoLTE And Kamailio?
Carsten Bock, Owner NG Voice, Germany
During the last year a lot has happened with Kamailio’s IMS implementation and surrounding systems. During MWC 2018, we released both our Management-Interface as well as our HSS and REST-API as open-source making the environment complete for IMS and VoLTE deployments, both for mobile and fixed line deployments.

In this talk, I will shed some light on what’s new in IMS based on Kamailio and describe the new features added over the year. Especially, I will highlight the design principles of our new HSS (which is based on Kamailio and other components) and show, how easy it is to scale the setup up and down, based on demand.

Open Source Software Myths
Fred Posner, The Palner Group, USA
Kamailio is open source software that can carries millions of calls every day for thousands of deployments world wide. So, what are some myths that give some enterprises concern about using open source software?
Fuzzing The Kamailio SIP Server
Henning Westerholt, Kamailio Project, Germany
Fuzzing is an automated software testing technique that involves providing invalid, unexpected, or random data as inputs to a computer program. The program is then monitored for exceptions such as crashes, or failing built-in code assertions or for finding potential memory leaks.

I will present in this talk my results in fuzzing the Kamailio SIP Server. This includes an overview of the fuzzing tool chain and motivation for applying this quality assurance method. I will describe the fuzzing setup and the necessary changes in the core for the fuzzer to interact with the server. Furthermore I will give an overview of the results and and describe possible future extensions.

Security, Authentication And Privacy: The WebRTC Challenge
Lorenzo Miniero, Co-Founder Meetecho, Italy
As a technology, WebRTC was conceived from the very beginning with security and privacy in mind. SRTP was mandated from the very beginning, and DTLS-SRTP soon followed, in place of the more widespread, but much less secure, SDES-SRTP instead. With the proliferation of server-based WebRTC services, though, the peer-to-peer nature of WebRTC is often bent and replaced with different topologies, which can in turn result in the loss of some of the properties the security mechanisms of WebRTC can provide. Even in peer-to-peer scenarios, though, these assumptions may be broken, especially if authentication is important.

The talk will try and describe the challenges these new communication patterns entail in the WebRTC world, explaining how the current protocols provide secure communication channels, and how authentication may be added with approaches like Identity Providers. The talk will then analyse the impact of WebRTC servers, how new standardization efforts like PERC aim at implementing privacy and end-to-end authentication/encryption across potentially untrusted components, and possibly even help in replicating DRM solutions for WebRTC broadcasting. Some considerations on real proof-of-concept integrations will be provided during the presentation, by referring to some work done in that direction on Janus as a reference WebRTC server implementation.

Homer 7 – Time To Born
Alexandr Dubovikov, QSC AG, Germany
A talk about the next generation version of Homer, the SIP capturing platform. Version 7 is a result of evolving based on user feedback and most common used cases, facilitating an easier integration within existing platforms, exposing more APIs and improved user experience.
Kamailio – Past, Present And Future
Elena-Ramona Modroiu, Co-Founder Kamailio, Asipto, Germany
A walk through the most relevant events of Kamailio project, with a special focus on the development during the last year and the plans for the future. Details about what is new in the latest stable release, Kamailio v5.1, and what else has been developed since then.
Open Discussions Panel – VUC Visions
Randy Resnick, VUC, France
What is new and exciting in the real time communications? What will be there in one or two years from now? Open discussions with a selected group of guests.
Dangerous Demos
James Body, UK
Interactive session:

  • Live and interactive ‘Dangerous Demos’ session which can be done by any of the participants at the event, with subjects containing material that is exciting, educational, entertaining, energetic and potentially explosive, of course, all harmless and related to anything Real Time Communications.
  • Among prizes:
    • Most Entertaining Demo (selected by judges/panel)
    • Audience Choice (selected using live (dangerous) voting system)
    • Most Ridiculously Risky Demo (aka the demo that crashes and burns most spectacularly)
Scaling Out FusionPBX With Kamailio
Mack Hendricks, RTC/VoIP Entrepreneur, USA
I love the fact that FusionPBX has multi-tenant support for SIP Domains. But, figuring out how to leverage the power of Kamailio to scale out that environment in a consistent and repeatable fashion can be a challenge (especially for newbies). This talk will focus on using Kamailio to scale out FusionPBX with a focus on domain registration and provisioning. I’ve taken this problem space and a few other common problems and wrapped it into a project called dSIPRouter. dSIPRouter allows you to quickly turn Kamailio into an easy to use SIP Service Provider platform, which enables two basic use cases, SIP Trunking services and Hosted PBX services.
Kamailio And FreeSWITCH For Video, Chat or Conference Service With Pure SIP
Giovanni Maruzzelli, Owner OpenTelecom.IT, Italy
SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. On top of traditional SIP deskphones and softphones, WebRTC has now universal acceptance and let us extend the reach of SIP on all browsers, and in apps for all smartphones (apps are not listening for incoming traffic, they can be pushed to wake from the server, and save the battery). Kamailio has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients.
Using CGRateS As A Database Backend For Kamailio
Dan Bogos, ITSysCom, Germany
Most of the billing solutions will require admins maintaining two sets of user data: one in Kamailio and second on billing side. In this talk Dan will introduce the CGRateS Attributes subsystem, a performance driven key-value store which can be used to unify data management, eliminating the need of using any database on proxy side. CGRateS is a battle-tested Online Charging System with support for both Prepaid and Postpaid billing modes.
Kamailio, Janus, BareSIP And MQTT – What A Lovely Platform!
Olle E. Johansson, Owner Edvina, Sweden
A wild brainstorm and inspirational talk about how to use SIP and MQTT hand in hand. Why I built an MQTT event handler for Janus and how I failed building the MQTT module for Kamailio, with a touch on using BareSIP project for a flexible client side toolkit.
SIP Is Dead – Long Live SIP
Nir Simionovich, Owner, Israel
Over the course of the past 15 years SIP has become the de-facto standard for VoIP Services. While it is by far superior to other protocols, it still suffers from its own caveats. Putting aside NAT issues, SIP isn’t the best protocol to use for Mobile VoIP services, and thus, something needs to change. This talk will describe how we changed the paradigm of SIP with a new RTC platform at, how we created a “Zero provisioning” and “Zero Battery” environment, while still maintaining a highly robust and efficient environment.
KamInboundSIP – Inbound DID Routing Platform
Surendra Tiwari, Plivo, India/USA
KamInboundSIP is an open source VoIP inbound DID call routing platform, leveraging Kamailio, RTPEngine and noSQL to provide flexibility and high performances. The project comes with a web GUI to manage the rules to route calls based on DIDs to customer’s telephony system. It has an extensive set of features to facilitate the SIP routing interconnects, security and accounting, among them:

  • inbound termination with carrier IP validation
  • carrier LCR for DID/TFN to PSTN forwarding
  • inbound calling abuse block
  • push CDRs to MongoDB
  • integration with IPTables to block SIP scanners and malicious attacks
  • use of RTPEngine for media streams relay
  • use of RedisDB for a fast storage
Kamailio – Least Cost Routing Engines
Daniel-Constantin Mierla, Co-Founder Kamailio, Germany
Kamailio offers a couple of options for building least cost routing engines. Straightforward options are dedicated modules such as lcr, carrierroute or drouting, but the beauty and flexibility comes when combining several modules to get innovative solutions that meet your custom requirements for a competitive service.

Besides presenting the benefits and the difference between the dedicated LCR modules, this talks shows few other options of building a LCR system by combining other modules such as mtree, dialplan, htable, dispatcher, …

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