We are working to build the schedule, meanwhile you can see a selection of presentations at the bottom of this page.

Pre-Conference Technical Workshops – Lightning Talks: May 06, 2019
11:45 ♦ Registration
12:20-12:30 ♦ Welcome Note
12:30-14:30 ♦ Workshops and Lightning Talks
14:30-15:00 ♦ Coffee Break
15:00-17:30 ♦ Workshops and Lightning Talks
17:30-17:45 ♦ End of Workshops – Closing Remarks
19:15-21:15 ♦ Social Event – Berlin City Boat Trip

Conference Day One: May 07, 2019
09:00-10:30 ♦ Conference Sessions
10:30-11:00 ♦ Coffee Break
11:00-12:30 ♦ Conference Sessions
12:30-13:30 ♦ Lunch Break
13:30-15:30 ♦ Conference Sessions
15:30-16:00 ♦ Coffee Break
16:00-18:30 ♦ Conference Sessions
18:30-19:00 ♦ End of Day One – Closing Remarks
19:00-21:00 ♦ Cocktail Party – Social Networking Event

Conference – Day Two: May 08, 2019
09:00-10:30 ♦ Conference Sessions
10:30-11:00 ♦ Coffee Break
11:00-12:30 ♦ Conference Sessions
12:30-13:30 ♦ Lunch Break
13:30-15:30 ♦ Conference Sessions
15:30-16:00 ♦ Coffee Break
16:00-17:45 ♦ Conference Sessions
17:45-18:00 ♦ End of Event – Closing Remarks

Next is a selection of sessions to be presented at Kamailio World Conference 2019.

(Workshop) Kamailio Extensions For Strong SIP Security
Henning Westerholt, Core Developer Kamailio, Germany
Kamailio developed over the years a large set of extensions targeting to allow building secure real time communications platforms. From old and venerable extensions such as pike, dns blacklisting or pipelimit, combined with well known htable, dialog or userblacklist, together with recent additions such as secfilter and sometimes blended with external tools such as fail2ban, there is no excuse to build a safe VoIP system to protect the service and the customers. This tutorials goes through the extensions offered by Kamailio in the security area, showing also severals configurations to protect against the common attack scenarios.
(Workshop) Kamailio – Ask Me Anything
Kamailio Devs
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) Optimizations For KEMI Scripting
Daniel-Constantin Mierla, Co-Founder Kamailio, Germany
Tips and tricks to have in mind when building SIP routing logic in KEMI scripting in order to get the best runtime performances. No matter you are using Lua, Python or other scripting languages, this session reveals the best practices to keep routing a high throughput of SIP messages while benefiting of flexibility and extensibility of external scripting languages together with built in features such as script reloading.
RTPEngine – Beyond RTP Relaying
Andreas Granig, CTO Sipwise, Austria
RTPEngine is known for its high performance RTP relaying capabilities, with its in-kernel forwarding mode scaling to over 10000 active sessions, as well as ability to encrypt and decrypt packets to gateway plain RTP to WebRTC and back. But there is a lot more that RTPEngine can offer: no downtime RTP session failover between active-standby nodes, repacketization, call recording, multiplexing and demultiplexing of RTP and RTCP, bridging between ICE-enabled and ICE-unaware user agents and even transcoding. This session given and up to date view of what RTPEngine can do and the plans for the future.
Advanced SIP: Communicating With Humans
Fred Posner, Owner The Palner Group, USA
Learning SIP was the easy part. Communicating with Humans requires the ability to adapt, improvise, and overcome. This presentation will discuss lessons learn and tips for how to better communicate with humans outside of the SIP protocol.
12 Years Journey With Kamailio
Yufei Tao, VoIP Consultant, PhD and PostDoc University of York, UK
From my 12 years experiences working with SIP and Kamailio, building SIP client software and implementing SIP servers, I have learned a lot from the work and the Kamailio community. With this sessions I am aiming to share some of my experiences and couple of nice tricks, which hopefully would be interesting and useful to the community, especially for new comers, but not only. They can be directly related to technical aspects of building VoIP systems with Kamailio and RTPEngine, such as SBCes, or to the social side of the open source community interactions.
Dealing With Outages
Simon Woodhead, CEO Simwood, UK
There’s two types of company: those who have had an outage and those who are going to have an outage. Their impact can be diminished through good architecture and planning but we are all likely to find ourselves responding to, possibly alone, a service affecting issue? What do you do? Through the real life account of Mr Black (name changed) we’ll look at what he did when in this situation, and what, eventually, the speaker lead him to do based on years of experience and lots of practice doing it wrong!
RTP Media Server Module
Julien Chavanton, Flowroute, USA
Recently a new module was introduced in Kamailio , RTP Media Server (RMS). In this presentation, I will show how the RMS modules was built using existing opensource libraries: MediaStreamer2 and oRTP, and explain why I believe we can leverage on the enormous amount of work needed to build both Kamailio and oRTP MediaStreamer2. Some examples of how to use Kamailio with the RMS module for playing files, bridging calls (currently under development), will be presented along with what media processing functionalities are provided (codec(s), resampling, dtmf, etc).
Operating Kamailio In Kubernetes
Seán McCord, Founder CyCore Systems, USA
The Kamailio Operator is a dynamic, flexible tool for deploying kamailio to kubernetes clusters. In this talk, we will discuss the various difficulties one encounters when deploying kamailio and voice systems to kubernetes, and then look at how the open source Kamailio Operator allows you to address them using a clean, declarative deployment definition and a fully-templated configuration system.
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)
What Is Going On In The World Of Asterisk?
Matthew Fredrickson, Manager Of The Asterisk Project, Digium, USA
The goal with this presentation is to talk a little bit about Digium’s recent acquisition and keep everybody informed on the current state of the Asterisk project. This will include what’s happened in the most recent major release of Asterisk, as well as what’s going on as we prepare for a new major release.
Designing A Scalable Billing System Using Blockchain Technology
Fabio Tranchitella, Evosip Cloud, Italy
Why and how created a real time scalable cloud native distributed billing platform for Kamailio with Docker and Kubernetes, with particular emphasis on security and consistency with hyperledger fabric blockchain technology for service providers.
Voice Over RTP != Video Over RTP
Tim Panton, CTO at |pipe|, UK
Many in the audience come from a VoIP (where V=Voice) background. Users however expect Video in their services and whilst both use the (S)RTP protocol on the wire, there are huge practical differences
between carrying voice and video. This talk describes the painful lessons I learnt when adding WebRTC Video to an existing technology stack. Hopefully this will make the transition easier for others. There will (of course) be demos and droids.
Making Kamailio More Inclusive And Easier To Adopt
Mack Hendricks, Founder dSIPRouter Project, USA
When the work on dSIPRouter started, the goal was (and still is) to create a UI that would make installing and configuring Kamailio much easier. We wanted to provide a prescriptive way to use Kamailio and as people became more comfortable they would use the Kamailio.cfg file we provide as a basis to create more sophisticated use cases or they would move to using Kamailio directly. In this session, I will give your an overview of the dSIPRouter project and walk you thru how people are learning Kamailio using this project and how carriers are using this project to help expedite scaling out their environment.
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 v5.2, and what else has been developed since then.
What Can You Do With Wazo And How Kamailio Can Help Us
Sylvain Boily, CTO Wazo, France
One of the typical problems nowadays is how to build a business oriented voice application to support both customers and partners, each with specific workflows, when the existing legacy solutions are too rigid, too complex, too expensive, too locked, too … Wazo is an open-source programmable platform for telecommunication that lets its users create value and build innovation on top without the traditional vendor lock-in. Our primary goal is to unlock your communication! We will show an example how a customer connected its existing business applications to our platform as well as how we provided an “infrastructure as code” for this particular case so that its service could scale and evolve easily.
Your Deployment On Stage – 5 Minutes 5 Slides
Markus Monka, Head of Infrastructure, Sipgate, Germany
Your chance as a participant to Kamailio World Conference 2019 to show what you are doing in the RTC space, what are your services and products, where and how Kamailio is used. You get 5 minutes to speak on maximum 5 slides and then let the discussions to continue during the breaks and social networking events.
PluSBC+ – SBC based on Kamailio and RTPEngine
Alexandr Dubovikov, Founder Homer Sipcapture, Plusnet, Germany
Introducing PluSBC+ from PlusNET, a streamlined and full-featured open source SBC based on Kamailio and RTPEngine featuring carrier-grade features for advanced routing, NAT and firewalling, transcoding, LCR, normalization rules and many more features implemented over a custom-built RTOS with a low memory footprint, designed to deliver brute session power. PluSBC+ comes ready with native monitoring and lawful interception capabilities via its advanced HEP integration and it is suitable as a production drop-in replacement for closed and expensive commercials platforms.
Event Routes, Timers And Other Mysteries
Olle E. Johansson, Owner Edvina, Sweden
Discover the hidden parts of Kamailio – from core or modules, event routes and timers can be the key for increasing vertical scalability of Kamailio or enable VoIP platform architects to add new capabilities via config to help operations team or attract more customers.
Building Your New Native Softphone With Web Technologies
Dan Jenkins, Founder of Nimble Ape, UK
Building native apps is a costly business; both in development time as well as developer costs. Web oriented web developers cost less than “pure” native developers for many reasons – but modern day web development techniques allow us to move super fast in development cycles. Want a new custom softphone for your VoIP business and don’t want to white-label an existing solution? Then you’re stuck building something that can handle all of Android and iOS’s nuances with integration into iOS’s CallKit and Android’s Connection Service as well as giving the user the best quality audio and call set up times experienced. The answer is to build your native app with Web Technologies using WebRTC and all of the libraries that bridge the gap between Web and Native. I’ll show you how those two worlds join and some of the things we have to think about.
SIP3 For Troubleshooting, Monitoring And VoIP-services Metrics Analysis
Konstantin Mikhailov, Co-Founder, Russia can help you monitoring and troubleshooting VoIP platforms without sacrificing your time and resources. Designed to scale as your business grows, SIP3 is a perfect fit for service providers, call centres and mobile operators. Originally created for helping support teams SIP3 has clean and simple user interface which makes troubleshooting any customer issue quick and easy. But that’s not all. SIP3 aims to not only reduce ticket’s resolution time but to also decrease the amount of customer support request. The product’s Dashboard gives you an insight on all collected service metrics. Keep your service uptime SLAs and track down accidents and VoIP service disruptions before they actually happen. Personalise your SIP3 Dashboard and monitor real-time key metrics that matter for your business. Meet us at KamaioWorld and find out more about the SIP3’s journey from troubleshooting to monitoring platform, all the biggest challenges along the way, and what has the team planned ahead.
What The Fuzz! Or Why You Should Really Fuzz Your RTC Code
Lorenzo Miniero, Co-Founder Meetecho, Italy
In the past few months, fuzzing has become a hot topic within the context of RTC projects. In fact, while in principle this is not strictly speaking a new or innovative approach (past Kamailio editions, and the latest one in particular, have already been home to some excellent talks on SIP fuzzing, for instance), its application to RTC projects, and WebRTC in particular, has always been limited up until very recently, when a blog post on Google’s Project Zero highlighted several key vulnerabilities in known and widespread multimedia applications. This sparked the interest, among WebRTC developers, on fuzzing their applications as well. Considering how complex the whole suite of WebRTC protocols is (ICE, DTLS, SRTP, RTP/RTCP, data channels, and even codec-specific payload access), there are several aspects that go beyond pure signalling and require dedicated considerations.

This talk will address some of the efforts most projects have taken in that direction, with a few additional details on how we tackled that in our Janus WebRTC server in particular. It will describe the process we followed with practical examples, and how we applied the same concepts to different protocols (RTP, RTCP, codec-specific payloads and SDP in particular) in order to make the implementation as robust as possible. Some considerations will also be provided on the steps we took to integrate fuzzing as part of the whole development process, on other fuzzing techniques that might be used, and on how this pattern could be deployed in existing scalable infrastructures to execute tests in a distributed way thanks to frameworks like OSS-Fuzz.

Scaling Your CGRateS Infrastructure Through Dynamic Partitioning
Dan Bogos, Owner ITSysCom, Germany
Growing Kamailio based networks require the billing side to cope with increased traffic and business rules for storing and handling user data in different locations. In this talk Dan will introduce the CGRateS Dispatcher Service, enhancing your installations with scalability and billing request dispatching over geographically distributed infrastructure. CGRateS is a battle-tested Enterprise Billing Suite with support for various prepaid and postpaid billing modes.
FusionPBX as ToolKit for SIP Services
Giovanni Maruzzelli, Founder OpenTelecom.IT, Italy
FusionPBX is the web interface for FreeSWITCH configuration and management. It strives to be faithful to FreeSWITCH power and flexibility, adding features on top of it, and hiding nothing. So, you can use FusionPBX to build any kind of SIP B2BUA services, complete of users and roles management, dialplan, php and lua scripting, db transactions audit, CDR visualization, call recording, etc. We’ll also see how to build a new, custom FusionPBX module to deal with the specifics of a new, custom call bridging service.
(Workshop) Homer Seven
Lorento Mangani, CEO QXIP, Netherlands
Learn how to install, configure and deploy the latest and greatest HOMER Sipcapture anywhere – Master the HEP stack troubleshooting and time-series generation features to capture hundreds of thousands VoIP packets and RTC events per second, monitoring simple and complex infrastructures in real-time, integrating with big-data platforms and why not – even adding a pinch of machine learning magic watching over VoiP and RTC Traffic and Metrics passing through Kamailio and towards your favorite Open-Source systems.
The Various Ways Your RTC May Be Crushed
Sandro Gaucci, Owner Enabled Security, Germany
From the author of SIPVicious (aka friendly-scanner), this presentation is taking a look at how attackers may bring to its knees an RTC system running on a decent setup (i.e. a medium sized telco, or a large RTC platform). It will cover the various attack vectors that have been explored in our past work, including the obvious (e.g. SIP, websockets) and the less obvious (e.g. recording systems). Additionally, it will discuss how rate limiting and other security solutions may be bypassed, subverted and become harmful in certain situations. Demos will be included to illustrate the beauty of it all.
Kamailio As An SBC: Definitive Answers
Alex Balashov, Owner Evariste Systems, USA
The question is frequently asked in the larger user community: can Kamailio serve as a Session Border Controller (SBC)? vThe aim of this presentation is to convince you that while the answer is rather generally “yes”, this is the wrong framing of the question. We will briefly explore the role of SBCs in the service provider voice network and where Kamailio fits in, and how the movement of RTC service providers to the cloud is shaping the next iteration of folk wisdom on service provider network architecture and best practices.
Replication Mechanisms For Multi-Node Kamailio Platforms
Daniel-Constantin Mierla, Co-Founder Kamailio, Germany
Kamailio offers several mechanisms to replicate data and states between different nodes in a VoIP platform. One can choose from low lever TM based replication functions, to PubSub architecture, database layer mirroring or going with the all-in-one solution DMQ (distributed message queue) with node auto discovery. This presentation shows all of them, when one suits better than the others and what are the aspects to be taken in consideration for various use cases, such as replication for user location, active calls or in memory hash tables.
back to top