Schedule

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 8, 2017
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 9, 2017

 

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 10, 2017

 

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



(Workshop) Kamailio For IMS – VoLTE
Carsten Bock, CEO NG Voice, Germany
An overview of 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) Homer: From Zero To Hero
Giacomo Vacca, Owner RTCSoft, Italy
Homer is an Open Source tool for real-time analysis and monitoring of VoIP and RTC platforms. It supports all the major OSS voice platforms, it’s modular, easy to install and scales to carrier-grade infrastructures. Homer goes beyond collecting and correlating signalling and logs, and can also capture RTCP reports, QoS reports, and other events. Through an ElasticSearch endpoint, Homer supports BigData analysis of traffic.

This workshop focuses on the deployment of a multi-node Homer framework with various approaches: bash installers, Docker containers, Puppet.

We’ll see how to configure Kamailio, FreeSWITCH (including the ESL interface), RTPEngine, Janus gateway (Events API), to collect signalling, RTCP reports, app-specific events and have them correlated and presented in a user-friendly GUI.

For advanced users, we’ll present the installation of captagent, the standalone capture agent, hepgen.js to generate test traffic, and a Wireshark dissector to have full visibility of data flows.

(Workshop) Load Testing Of SIP and WebRTC Infrastructures
Lorenzo Miniero, Co-Founder Meetecho, Italy
Several tools exist for the stress and load testing of SIP infrastructures. Whether the purpose is to only test signalling, or involve media as well, there usually is a solution that can help achieve the desired result. When the SIP infrastructure is WebRTC based, though, this becomes a less trivial problem. In fact, generating a lot of legitimate WebRTC traffic is complex, and simply replaying previously captured traffic would break the security constraints imposed by all the involved protocols (ICE, DTLS, SRTP). This talk aims at providing an overview of the solutions already existing and widely used for SIP/RTP, and show how the approach must be refocused once WebRTC comes into the picture.
(Workshop) Kamailio Configuration Optimizations
Daniel-Constantin Mierla, Co-Founder Kamailio, Romania
A practical tutorial about the tools and mechanisms offered by Kamailio to optimize the execution of SIP routing scripts: caching systems and fast indexing of large data records, tips and tricks about SIP routing logic, smart execution control, benefits and drawbacks of various optimization options.
Kamailio And Next Generation Emergency Services
Wolfgang Kampichler, Frequentis AG, Austria
Next Generation Emergency Services are designed to close the gap between the quickly evolving technologies (fixed and mobile IP-based communications) and the more conservative approaches required by the emergency communications industry. This presentation provides a comprehensive overview of the architecture envisioned and standardized for NG emergency services. An important functional element of this architecture is the emergency services routing proxy (ESRP), which is the SIP entity that makes decisions about the call routing by using service URNs, location information and domain specific routing policies. Kamailio, as feature rich, reliable and performant communications platform is well suited for such scenarios, therefore, this contribution gives also a deep insight into the process of next generation emergency call routing utilizing Kamailio.
Billion Calls With A Modern Stack
William King, Chief Architect, Flowroute, USA
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.
The Future Of RTC And 5G Services
Thomas Magedanz, Prof Technical University Berlin, FhG Fokus, Germany
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, machine to machine communications and IoT, offering playgrounds for IMS, EPC and 5G. The talk gives an overview of current research directions that can impact the way we communicate tomorrow, what 5G is targeting for the interactions of humans, machines and everything in between.
Resilient Kamailio Multi-Node Infrastructure For Carrier Services
Pablo Quiroga, NexVortex, USA
Showing the architecture and topology of a carrier specific platform, with details on database replication mechanisms, inter-node interactions and elasticity, implementing E911/112 services, different roles with Kamailio across the network as well as how automatization with Ansible helps the operations and scalability.
Why IVR Is Essential To RTC
Allison Smith, The IVR Voice, Canada
IVR is often seen as a necessary evil; an often abrupt and unwelcoming gateway to a company. Industries are realizing that their IVR sets the tone for callers — it tells them everything they need to know about the company and what it will be like to deal with them. It is essential that the outward-facing IVR on your RTC system (with Kamailio or without), is intuitive, easy to use, and presents your company in the most positive light.
RTPEngine Media Processing For Fun And Profit
Andreas Granig, Sipwise, Austria
The Sipwise rtpengine for Kamailio provides various ways to analyze and process RTP flows for ongoing calls. This talk describes the concepts to report the quality of calls in real-time to Homer, to store raw RTP packets to PCAP files, and to record plain and encrypted audio call streams for standard SIP and WebRTC calls in WAV and MP3 format.
Why I Love Kamailio
Fred Posner, LOD, USA
From large scale deployments to Raspberry PI systems, there isn’t much that Kamailio can’t cover. The software has come a long way, and yet there’s so much more we can do as a community to give back to the product.
Having Troubles With Fraudsters? Kamailio HTable To The Rescue!
Sebastian Damm, Sipgate, Germany
Telephony fraud is still on the rise. Everybody running a telephony system in the internet should be aware of possible attacks. This talk will show some easy-to-implement examples on how to detect and prevent fraud.
Kamailio As A Stateless, Containerized Session Border Controller
Jan Lorenz, Pascom, Germany
This specific use case presentation demonstrates how Kamailio can be used as a containerized, stateless session border controller whose main mission is to protect and defend an existing Asterisk infrastructure from the inherently dangerous and nefarious internet.
The Strategy And Technical Mechanics Of Building A VoIP Global Network
David Casem, Telnyx, USA
The internet is great for non-RTC based web applications. When it comes to telephony however, the fact that packets take the cheapest path – not the shortest – means that latency and packet-loss sensitive packets are exposed to not only degradation, but possible interception.

By leveraging cloud infrastructure, containerization, and a highly efficient VoIP PoP design, Telnyx has successfully shortened the length of the “last mile” by quickly taking packets off the internet and putting them on an MPLS-TE backbone. In doing so, not only we protect better users’ packets, but ensure uptime through geographic diversity, redundancy and resiliency.

In this presentation, you’ll learn why and how Telnyx built a global, cloud agnostic network and the tooling used to deploy and manage it.

Scaling VoIP – The AWS Advantage
Nir Simionovich, Owner Greenfield, Israel
Building a VoIP network isn’t an easy thing to do, building a VoIP network that scales at ease using commodity cloud offerings is far more complex than it sounds. This presentation will introduce one of our projects and use cases, for deploying a world-wide wholesale carrier, using Kamailio and Amazon AWS services.
Listening By Speaking – An Under-Estimated Security Attack On Media Servers And RTP Relays
Sandro Gauci, Enable Security, Malta/Germany
In this presentation we take a look at a vulnerability affecting a number of media servers, some of them being often used together with Kamailio. The vulnerability allows hijacking of the RTP stream without a typical man-in-the-middle attack. We include demos, packet captures and other fun and practical ways to explain the vulnerability. Then we will assess the impact of such a vulnerability as it affects traditional media gateways for VoIP systems as well as the WebRTC infrastructure. Finally we discuss potential solutions and hopefully have a bit of a Q&A too!
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)
Routing On The Host And Anycast SIP
Simon Woodhead, CEO Simwood, UK
The network is a strategic asset that can make a difference to your end-users’ communication experience. While most of VoIP platforms rely on unicast routing, here we show the advantages and how to build a SIP anycast routing for optimum node discovery, redundancy and scalability.
What Is New In Asterisk
Matthew Fredrickson, Digium, USA
A few months ago, we closed out the year and Asterisk 14 was released. This talk is to take a look at Asterisk’s development, what’s going on there, and where the project is heading from a technology perspective, what are the new features and tools that can be used to leverage Asterisk for VoIP platforms and integrate it with other RTC components.
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.
Jitsi: Open Source Video Conferencing
Saúl Ibarra Corretgé, Atlassian, Spain/The Netherlands
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.
We Have Got The Wizardry, What Now?
Dan Jenkins, Nimble Ape, UK
Why build any experience that isn’t magical for your user? Building anything truly extraordinary used to be really difficult; but today we can do it in a few hundred lines of code. But can you truly say you’re offering your users the most spellbinding experience possible? It’s doable today but we’re not doing it everywhere yet.

Natural Language processing, Real Time Image processing and speech recognition APIs; they’ve all moved forward at a rate of knots in the past couple of years. What used to require huge investment in hardware, software and talent is now available to everyone who can write a few hundred lines of code. Even though we have huge power at our fingertips with cloud and browser based APIs, we aren’t truly utilising this technology to make a meaningful difference to our customers, users and our businesses today. Building for tomorrow is difficult when we’re still thinking about yesterday. In this session you’ll learn about the APIs available to you today and how you can make a difference to what your applications are doing to build relationships with your users. Developing that connection with the user can be magical. Give your user magic.

5G Platforms And Playgrounds
Marius-Iulian Corici, FhG Fokus Institute, Germany
An overview of the testbeds and platforms for 5G offered by Fraunhofer Fokus Research Institute, generally available for industry or academic entities, what they offer and how they can be used to plan, test and assert feasibility of new ideas for the future of communications.
Centralized VoIP Resources Allocation Using CGRateS
Dan Bogos, ITSysCom, Germany
Modern business models require us enforcing various limits for the channels/data streams passing through our infrastructure. In this talk Dan will introduce the ResourceLimiter component of CGRateS, adding complex channel reservation capabilities to a pool of Kamailio servers. There will be exemplified advanced usage scenarios such as channel reservation per destination, time based or traffic load related.
CGRateS is a battle-tested Online Charging System with support for Prepaid, Postpaid, Pseudo-prepaid and Rated charging modes.
Cloud API For SIP Devices Management
Sherman Scholten, Obihai, USA
A presentation about device management cloud APIs and the relevance to network agnostic deployments of VoIP endpoints (phone or fax adaptors and IP phones). How it can improve user experience, ensure seamless firmware upgrades for security and new features, and facilitate operations for large VoIP services.
RTC-TIE: Building an RTC Threat Intelligence Exchange
Alexandr Dubovikov, QSC, Germany
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.
Getting Started With WebRTC
Chad Hart, Voxbone, USA/Belgium
Starting with an overview and brief history of WebRTC, a 5-year-old real time communications technology, this presentation will look at WebRTC use cases in modern communications. Chad will show how you or your company can get started with WebRTC and implement APIs while being aware of the back-end considerations and challenges.
Diameter Processing with Kamailio
Carsten Bock, NG Voice, Germany
While SIP is a common protocol for Kamailio, Diameter is not. Diameter Processing was heavily limited to be client-side operations in Kamailio, e.g. acting as Client for an Online-Charging System or for requesting QoS in an VoLTE network. Until now. Apart from our Online-Charging-Server module (which we added last year), we recently added a generic Diameter-Processing Module to our beloved and long-term friend and SIP-Proxy Kamailio. Diameter-Processing in Kamailio? In this talk I will sched some light on the development, the goal and capabilities of this development.
Building A Modern Doorbell With A $9 Device And WebRTC
Tim Panton, Pipe, UK
Learn how to build a doorbell using the Getchip $9 Linux machine, solving the following problems:
1) setup and config – how do you get a wifi password onto a device with no keys?
2) pairing – how do you get a device to identify itself and it’s owner?
3) network transparency – how do you get through a complex NAT?
4) notifications – how to wake a smartphone
5) media transfer – how to talk with your visitor.
As usual there will be a stupidly dangerous demo and lots of VoIP tools will be (mis) used.
IMS And EPC – The Core Layers For 4/5G Services
Dragos Vingarzan, Core Network Dynamics, Germany
Leveraging the work done during the past decade with IMS and EPC in research as well as industry, this presentation is highlighting how these technologies can enable the operators to extend their offers beyond the traditional telecom services.
Kamailio On Air! How Sveriges Radio Use Kamailio In Live Radio Broadcasts
Olle E. Johansson, CEO Edvina, Sweden
EBU, the European Broadcast Union, has standardised on using SIP as an infrastructure for radio broadcasting. The Swedish national public radio has built a next generation broadcast platform based on Kamailio, BareSIP and an in-house platform for managing the system. The talk will describe how this new platform is built, how it is used on in-house networks as well as mobile and remote Internet connections. The talk also describes how the new platform has changed how radio is produced – the transition from old ISDN technology to a IP based system with built-in support for mobility.
Meeting Evolving Class 4 Demands With Kamailio
Alex Balashov, Owner Evariste Systems, USA
We built our CSRP product as the 2000s drew to a close. Since then, the IP telecoms market has evolved considerably, and with it, so has the nature of customer demands. While SIP trunking is as popular as ever, the “peak cultural moment” of billing conversation minutes has arguably passed; at any rate, it’s receding into the background in the USA and in other developed economies. It’s not 2008 anymore. We will discuss the changing shape of customer demands and expectations of a Class 4 switch platform as we get into 2017, and the ways CSRP, as a Kamailio-based platform, is evolving to meet them.
2001-2010-2100: 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 v5.0, and what else has been developed since then.
Kamailio.cfg Scripting Languages
Daniel-Constantin Mierla, Co-Founder Kamailio
Since v5.0.0, Kamailio allows several scripting languages to be used for building SIP routing logic. From the old-good native scripting language, to Lua, Python or JavaScript, one can choose the language that is more suitable for own needs. Besides the flexibility and extensive list of extensions coming with Python, Lua or JavaScript, with some of them (Lua and JavaScript) it is also possible to reload the config routing script without restarting Kamailio. The talk will reveal the benefits and limitations for all these scripting languages that can be used inside kamailio.cfg.
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