Schedule

12:15 ♦ Opening Attendee Registration

12:45-14:45 ♦ sip:provider CE: How to deploy WebRTC, Class5, Class4 and SBC Services within minutes

Coordinator: Andreas Granig, Co-founder Sipwise, Austria
Description: By leveraging the power of Kamailio and SEMS, accompanied with flexible billing and provisioning frameworks and APIs, the sip:provider CE is a versatile turn-key open source telephony appliance to cover a wide range of use cases for tens of thousands of end users. In this work-shop, Sipwise engineers will guide you through the concepts of the appliance and help you deploy whatever service you would like to build on top.

Follow the instructions at this link if you want to actively participate during this workshop.

14:45-15:00 ♦ Coffee Break

15:00-15:45 ♦ Kamailio as an IMS Core

Coordinator: Carsten Bock, Owner NG Voice, Germany
Description: Hands on tutorial of how to deploy Kamailio as an IMS core platform.

15:45-16:30 ♦ Kamailio-based Prepaid Engine

Coordinator: Carlos Ruiz Diaz, VoIP Consultant NG Voice, Paraguay
Description: Configuring Kamailio as a SIP prepaid engine, setting limits for duration, number of active calls as well as using same credit account for charging in real time.

16:30-17:30 ♦ Getting Started with SEMS

Coordinator: Stefan Sayer, Lead developer SEMS, Frafos, Germany
Description: How to configure a SEMS instance for offering common media services such as announcements, voicemail, audio conferencing and IVR menus.

08:30 ♦ Opening Attendee Registration

09:00-09:10 ♦ Welcome!

Speaker: Daniel-Constantin Mierla, Co-founder Kamailio, Asipto, Germany
Description: Getting together in the conference room and overview of the event.

09:10-09:25 ♦ Kamailio: State of the project and the infrastructure behind it

Speaker: Elena-Ramona Modroiu, Co-founder Kamailio, Asipto, Germany
Description: From 2001 to 2014, SER to Kamailio, 14 major versions and over 60 releases, hundreds of developers contributing over 800k lines of code. What keeps us running? Whose shoulders, when and where sustained the project and provided resources for continuous development and to maintain high standards of quality without losing the focus on innovation.

09:30-09:55 ♦ #MoreCrypto – We can do better and we will!

Speaker: Olle E. Johansson, Owner of Edvina, Sweden
Description: During the last year, the world woke up to pervasive monitoring. Every network engineer needs to do better, to protect the users. Every application designer, sysadmin and open source developer needs to stand up to the challenge and ask ourselves: Can we do better? The answer is simple: Yes, we can.

The world of realtime communication has had a type of customer not really bothering with security in the sense of data communication security. As we move out on the Internet with SIP over WebRTC and WebSockets, softphones on every laptop and open federations we need to do better. It’s time for the Open Source developers to take the lead and show the users how it can be done and improve our own applications.

This talk outlines a few items where we can improve our Kamailio installations as well as the Kamailio source. It suggests some work items to focus on until next Kamailio world as well as hints at improvements in other SIP platforms, clients as well as servers

Olle E. Johansson is active in many Open Source communities as well as in the IETF and ISOC-SE. He has promised to improve his security skills and challenge you to do the same.

10:00-10:25 ♦ Kamailio: The platform for Interoperable WebRTC

Speaker: Peter Dunkley, Technical Director of Acision, UK
Description: WebRTC will revolutionise the telecommunications market. There are many silo’d WebRTC use cases which do not require interoperability, but there are also many use cases in which interoperability is key. Kamailio (with it’s websocket and auth_ephemeral modules) is the ideal platform to build interoperable WebRTC services on. This presentation will provide a high-level introduction to WebRTC and SIP over WebSockets, describe some example use cases which require interoperability, and show how to configure Kamailio for this.

11:00-11:25 ♦ Tearing down walls: WebRTC and IMS – lets re-unite!

Speaker: Carsten Bock, Owner of NG-Voice, Germany
Description: IMS is the the “Best of the 90’s” and often associated with huge Telcos, while WebRTC is often seen as the next big game changer in agile IP communications. How does that fit together? We will shed some light on merging the best of both worlds: We will discuss the benefits of a real converged infrastructure of IMS and WebRTC and how WebRTC fits into an IMS-best-practice-architecture. I will explain, how Kamailio and other open-source components can help you to achieve that goal.

11:30-11:55 ♦ VoIP Fraud Analysis

Speaker: Simon Woodhead, Owner of Simwood, UK
Description: A presentation of the most common VoIP fraud mechanisms along with solutions to prevent and protect your business and your customers from dramatic financial loss. The content is based on doing some analysis of three years of data collected operating the Simwood SIP honeypot, which represents over 60m related events – the results uncovered some amazing and shocking things but we also have some good news!

12:00-12:25 ♦ Kamailio and OpenStack – Together to build a truly scalable solution

Speaker: Ruben Sousa, CTO of ITCenter, Portugal
Description: The challenge was to build a full featured VoIP system, highly scalable, highly resilient, geographically distributed, with low bandwidth consumption and without a single point of failure. Presentation is how Kamailio and OpenStack was used by ITCenter to achieve that.

12:30-13:30 ♦ Lunch

13:30-13:55 ♦ Continuous Integration: the new Kamailio Build Process

Speaker: Andreas Granig, Co-founder and CTO Sipwise, Austria
Description: With over 130 people actively contributing to Kamailio, code review, regular builds and automated testing becomes crucial for a software project at that scale and scope. This presentation will describe the new build process running behind the scenes of Kamailio to produce stable nightly builds and release snap-shots for Debian, and the proposed work-flows for improving code quality.

14:00-14:25 ♦ Automatic Kamailio Deployments with Puppet

Speaker: Giacomo Vacca, Senior Network Applications Developer at Truohone, Italy
Description: Kamailio is a robust, powerful application; virtualization and cloud computing make it extremely easy and quick to setup Kamailio-based solutions. Scaling, geographic distribution and multiple environments (e.g. development, staging/QA, production), though, present some deployment challenges. Puppet provides a solution that dramatically cuts deployment time, reduces occurrences of errors, while at the same time documenting the configuration status. This presentation describes what we did in Truphone Labs to move from a freshly created virtual machine to a running Kamailio instance, automatically and in minutes. Firewall, nagios, syslog, monit, sec, and other related services are also automatically configured.

14:30-14:55 ♦ Asterisk’s PJSIP channel driver: a SIP architecture for the future

Speaker: Matt Jordan, Engineering Manager for the Open Source Software team at Digium, USA
Description: In Asterisk 12, the Asterisk Developer Community replaced Asterisk’s venerable yet aging chan_sip channel driver with a new SIP stack and channel driver based on Teluu’s PJSIP project. The new PJSIP stack has a dramatically improved architecture and design when compared against the legacy chan_sip channel driver, which makes it attractive for deployments of all sizes and types. In this presentation, we’ll look at an overview of the new PJSIP architecture in Asterisk, as well as how it can be effectively deployed with Kamailio.

15:00-15:25 ♦ Jitsi Videobridge and WebRTC

Speaker: Emil Ivov, Founder Jitsi Project, Bluejimp, France
Description: Jitsi Videobridge is an open source video router based on libjitsi and developed by the jitsi.org community (home of the Jitsi rich client). While it initially started as a backend for Jitsi video conferences, Jitsi Videobridge recently met WebRTC and decided they should play together! Today Jitsi Videobridge is a viable open source alternative to solutions such as the VidyoRouter that Google use to power their hangouts. Emil is going to walk us through the history and specifics of the project, the principles of video conferencing and an overview of how easy it is to integrate the bridge in VoIP and Web projects.

15:30-16:00 ♦ Coffee Break

16:00-16:25 ♦ Dynamically scaling Kamailio/Asterisk based queues

Speaker: Nir Simionovich, Owner of Greenfield Technologies, Israel
Description: Call queues – we all hate them, but can’t live without them. Recently, we’ve finished a project involving dynamically allocated call queues, where Kamailio and Asterisk were used to implement a whole new style of queuing system – highly scalable, cloud ready and highly efficient. This talk will illustrate the challenges that were introduced, how these were mitigated, what was done and what still remains.

16:30-16:55 ♦ The poor state of SIP endpoint security

Speaker: Henning Westerholt, Head of IT Operations Access at 1&1 Internet AG, Germany
Description: SIP endpoint devices like CPEs or SBCs are common and necessary infrastructure elements in any voice over IP network. The talk gives an overview about the poor security state of these devices: what you should expect from a device vendor with regards to security responses; how to protect your infrastructure against common attacks and be prepared when security incidents happen. Different stories from the field and several best practices techniques from past experiences running one of the largest Kamailio back-ends.

17:00-18:00 ♦ Open Discussions Panel – VUC Visions

Moderator: Randy Resnick, Founder VoIP Users Conference, France
Description: The state of real time communications and their future.

18:00-18:10 ♦ Day One Closing Remarks

Moderator: Organizers

19:00-22:00 ♦ Social Networking Event – Cocktail Party

Description: Make new business connections and discuss what is new in real time communications lately while enjoying prestigious German beer, excellent wine and tasteful food.
Location: Atrium area – next to conference and exhibition space.

09:00-09:25 ♦ An Erlang Integration for Real-Time Call Cost Control

Speaker: Seudin Kasumovic, Software Developer, Bicom Systems, Bosnia and Herzegovina
Description: The presentation will show the integration of Kamailio with a real-time telephony billing application using Erlang, describing the design and data model built for entity abstraction and call cost control for post/pre paid subscribers, vendors (SIP trunks) and hosted PBX as well as the challenges to develop a custom Kamailio module for real time interaction with the billing application.

09:30-09:55 ♦ WebRTC signaling: alternatives

Speaker: Antón Román Portabales, CTO of Quobis, Spain
Description: Presenting how Quobis used Kamailio to setup the WebRTC demo platform and connecting it to PSTN, then moving to show a comparison of the alternatives for signaling found in different WebRTC solutions, and other interesting topics like the management of trickle ICE in terms of signaling.

10:00-10:25 ♦ Forgetting about legacy

Speaker: Dragos Vingarzan Senior Researcher, Dr., FhG Fokus, Germany
Description: Providing 2G/3G CS-services together with VoLTE/VoWiFi PS-based voice on single telephony core — the OpenEPC project is a comprehensive implementation of all 3GPP Rel.12 Mobile Core Network elements. Started in 2008 at Fraunhofer FOKUS by the same team that created also the Open Source IMS Core project, OpenEPC has moved since late 2013 into its next phase. Core Network Dynamics, as a company spun-of Fraunhofer FOKUS, seeks to enhance the very successful test-bed prototypes with carrier-grade features and integrations with existing telecom infrastructure. While VoLTE/VoWiFi are basic features, the talk seeks to explore the issue of integrating PS-based voice with 2G/3G networks where CS-voice is still deployed for legacy with existing terminals. The OpenEPC project has prototyped a radical approach MSC components, which is capable of translating early between GSM-Layer 3 signaling and SIP. This in effect emulates a SIP/IMS User Endpoint for every legacy 2G/3G terminal in the network, effectively eliminating the need to continue support for SS7/INAP/MAP/a.s.o signaling, while serving all voice/SMS facilities from a unified NGN/IMS voice core. Additional benefits are then provided by the simplification of SRVCC issues and the use of a single service platform for both CS-anchored and PS services.

10:30-10:55 ♦ Coffee Break

10:55-11:20 ♦ RTC Beyond Borders

Speaker: Alan Duric, Co-founder Telio Holding ASA, Norway
Description: A perspective over the past months events in RTC market and what to expect in the near future.

11:25-11:50 ♦ 10 years of working with Kamailio at sipgate

Speaker: Krischan Udelhoven, Software Developer and VoIP Engineer, Sipgate, Germany
Description: In the past 10 years sipgate evolved from a VoIP startup to one of the largest VoIP providers in Europe. The talk will present the most important steps in the development of sipgate’s Kamailio based core routing system, that helped making this progress possible.

11:55-12:45 ♦ (Dangerous) Live Demos – Interactive Session

Moderator: James Body, Head of Research and Development at Truphone, UK
Description: Participants will each have no more than 5 minutes to present their offering, which can consist of any clever and imaginative concept – as long as it has Kamailio or open source VoIP/RTC software embedded somewhere within the mix! Marking will give points for originality, usefulness, excitement, amusement value and technical risk! The winner(s) will receive a prize donated by Truphone.

12:30-13:30 ♦ Lunch

13:30-13:55 ♦ Kamailio and large-scale networks: Using it as a front-end for FreeSWITCH

Speaker: Karl Anderson, CTO of 2600hz, USA
Description: A presentation of how 2600hz has used Kamailio’s flexible and in-depth module system, showing how we built our own module and the mechanisms for tying it into our core AMQP architecture for Kazoo platform. This helped to offload heavy SIP traffic volume from FreeSWITCH while allowing us to ramp-up automatic server distribution.

14:00-14:25 ♦ SIP and SS7 interworking

Speaker: Torrey Searle, Senior Voip Architect at Voxbone, Belgium
Description: Integrating SIP and SS7 networks can be quite complex. This presentation will explain how the new SIP-T module in Kamailio can be used to to extract information from SS7 headers as well as being able to modify the ss7 headers to implement advanced SIP to SS7 call scenarios.

14:30-14:55 ♦ Modern Performance Testing with Open-Source Tools

Speaker: Robert Day Lead Developer SIPP, Metaswitch Networks, UK
Description: This session would discuss the options and strategies available today for using open-source test tools to test the performance and functionality of Kamailio and other elements of a VoIP network. My main focus will be on the SIPp testing tool widely used in the industry, presenting some advanced strategies for creating more complicated test scenarios, the options available for media testing with RTP (including some recent high-performance developments), and how to build custom automation and reporting frameworks around SIPp. Other tools will be briefly approached, such as Quaff Ruby library for creating simpler, more scriptable SIP test scenarios, and the Seagull test suite for testing a wider variety of protocols (particularly Diameter, the protocol Kamailio uses for its IMS Cx and Ro interfaces).

15:00-15:25 ♦ Homershooting: Troubleshooting Voice in Real-Time

Speaker: Alexandr Dubovikov, Founder of Homer Sipcapture Project, QSC, Germany
Description: HOMER is a de-facto standard SIP capture and monitoring tool for large and small IP voice networks alongside other well known complementary tools, although most users barely scratch the surface of its great potential. The audience will be presented with some advanced troubleshooting scenarios and configuration examples for attack/abuse or fraud alarming and detection, UA QoS reports monitoring and interaction with other flexible tools such as sipgrep.

15:30-16:00 ♦ Coffee Break

16:00-16:25 ♦ Kamailio: Tips, Tricks, and Notes From the Field

Speaker: Alex Balashov, Owner of Evariste Systems, USA
Description: An eclectic collection of different route script tricks and particularly interesting, but under-emphasized modparams, as well as some notes on common pitfalls.

16:30-16:55 ♦ Real-time Toll Fraud Detection with Automated Mitigation using CGRateS

Speaker: Dan Bogos, Owner of ITSysCom, Germany
Description: Recent research has shown a continuous increase of Toll Fraud, becoming a multi-billion global threat with monetary damages more than double that of Credit Card Fraud. In the future it is expected to become an even more sensible subject especially with widening the adoption of VoIP in our day to day communication. In this talk Dan will walk the audience through various techniques of detecting and real-time blocking the Toll Fraud in the billing component of a carrier network.

17:00-17:15 ♦ SEMS as open source SBC and call stateful SIP toolbox

Speaker: Stefan Sayer, Lead Developer SEMS Project, Germany
Description: The SIP Express Media Server, originally a media server to complement SIP based VoIP networks with services such as voicemail, conferencing and IVR services, with the introduction of a B2BUA and especially the SBC module can be useful as call stateful control element in the operator core, for specific applications, or as a full Session Border Controller. In upcoming SEMS 1.6, again a whole lot of useful functionality has been and will be added, among them registration handling, tcp stack and more NAT traversal options, transcoding, multiple interfaces, bandwidth limiting etc. Further, a new extended call control interface which even can be scripted with the super simple DSM state charts language not only supports more complex, PBX type call flows, but also makes SEMS a toolbox useful for most situation where a call stateful element is needed.

17:20 – 17:35 ♦ Building application servers for IMS

Speaker: Carlos Ruiz Diaz, VoIP Consultant, NG Voice, Paraguay
Description: While the Kamailio IMS-core is quite stable and reliable, we will look on building application servers for an IMS setup. We will start on discussing different types of application servers and we will look, how different open-source projects can serve for the different types of an application server in an IMS world. We will look at a basic MMTel for Telephony features, at a presence-server and on more advanced topics, such as adding a video-MCU to an IMS-setup.

17:40 – 17:55 ♦ Asynchronous SIP routing with Kamailio configuration scripting

Speaker: Daniel-Constantin Mierla, Co-Founder Kamailio, Asipto, Germany
Description: There could be interesting features that, even not a critical service for customers, can make the difference on the market, like notifications to other social or rtc networks, information discovery and mid session updates, statistics collectors. But they might be blocking operations and that can have big impact in SIP routing performances. Kamailio has a comprehensive set of tools in configuration file to handle such task in an asynchronous way. The talk will detail them, showing some interesting examples as well.

18:00-18:10 ♦ Closing Session