Please note that the schedule may change.
|11:45 ♦ Registration|
|12:20-12:30 ♦ Welcome Note|
|12:30-13:20 ♦ (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.
|13:20-14:10 ♦ (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.
|14:10-14:30 ♦ (LT) 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:
|14:30-15:00 ♦ Coffee Break|
|15:00-15:20 ♦ (LT) reSIProcate Overview|
|Daniel Pocock, reSIProcate Maintainer, Switzerland|
|A high-level overview of each component of the reSIProcate project, including the SIP stack, Conversation Manager, repro SIP proxy, reTurn (TURN) server and client library, the XMPP gateway and Telepathy integration on the Linux desktop. Discussion of reSIProcate’s strengths, when to use it, how it works with other free software projects and how to contribute.|
|15:20-15:40 ♦ (LT) Tools For VoIP Security|
|Sandro Gauci, Founder Enabled Security, Germany|
|Summary of what tools, old or new, open source or proprietary, can be used for VoIP security checks and SIP fuzzing. What were the relevant VoIP security issues revealed during the past year, what can be done to protect better your VoIP systems.|
|15:40-16:00 ♦ (LT) Load Balancing With Congestion Detection|
|Julien Chavanton, Lead VoIP Software Engineer, Flowroute, USA|
|Last year, latency monitoring was introduced to Kamailio’s dispatcher module. Along with, a new load balancing algorithm is being developed: “relative weight based load distribution with congestion detection”. This one is able to greatly minimize the impact of congested nodes (gateways) by adjusting the load using a latency estimator. This talk is presenting how the algorithm is working and how to configure it in a SIP load balancer system.|
|16:00-16:40 ♦ (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 https://github.com/kamailio/kamailio-tests.
This workshop presents the unit testing framework, how to add new units and how it can help you test your Kamailio deployment.
|16:40-17:20 ♦ (Workshop) Migrating SIP Routing Logic To A KEMI Scripting Language|
|Daniel-Constantin Mierla, Co-Founder Kamailio, Germany|
|17:20-17:30 ♦ End of Workshops – Closing Remarks|
|09:00-09:30 ♦ 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.|
|09:30-10:00 ♦ 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.
|10:00-10:30 ♦ 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.|
|10:30-11:00 ♦ Coffee Break|
|11:00:11:30 ♦ Kamailio With Docker And Kubernetes: Scale In The Right Way|
|Paolo Visintin, Co-founder of Evosip.cloud, Italy|
|How to deploy a full containerized solution using Kamailio with Docker orchestrated by Kubernetes.
Topics to be approached:
|11:30-12:00 ♦ 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.|
|12:00-12:30 ♦ 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.|
|12:30-13:30 ♦ Lunch Break|
|13:30-14:00 ♦ SIP Is Dead – Long Live SIP|
|Nir Simionovich, Owner Cloudonix.io, 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 cloudonix.io, how we created a “Zero provisioning” and “Zero Battery” environment, while still maintaining a highly robust and efficient environment.|
|14:00-14:30 ♦ 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.|
|14:30-15:00 ♦ The VoIP Platform Architecture With AutoPilot Pattern|
|Simon Woodhead, Owner Simwood eSMS, UK|
|Each VoIP platform that grows has to go through the phase of being just not big enough for the problems of big orchestration to be worth the effort yet. Still, there is need to manage hosts numbering in the high 10s, plus containers and virtual machines running into the thousands, and they could be worldwide in multiple data centres. At Simwood, we rely on a technique called AutoPilot which essentially means a container is self-describing and self-managing, and thus runs the same anywhere. This talks is about the challenges we had to resolve, how we replaced the Docker network stack in order to make containers first class citizens on the network, enable anycast, and to put our devs in control of both network requirements and firewalling, as well as integration of applications for SIP routing, billing or database systems.|
|15:00-15:30 ♦ 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.|
|15:30-16:00 ♦ Coffee Break|
|16:00-16:30 ♦ 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.|
|16:30-17:00 ♦ 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.
|17:00-17:30 ♦ 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.
|17:30-17:50 ♦ Experimental Audio Apps Using WebRTC And WebAudio|
|Tim Panton, Co-Founder PIPE, UK|
|WebRTC and webAudio can be combined to create new (non-traditional call model) audio applications that fit specific niches. This talk describes 2 recent open source apps – (code is on GitHub). PodCall is a “serverless” tool for recording remote podcast interviews, it uses webRTC to minimise the requirements needed for the interviewee – they just use their smartphone browser. UnRupt is a “serverless” demonstration of a new conversation mechanism where you can speak over the other party and the system delays their voice until you finish. Machine assisted politeness if you will.
The talk is approaching the design decisions, walk through the code and describe the web features we use, the experience of Safari’s webRTC support, and as ever, I will attempt to demo the apps live.
|17:50-18:50 ♦ 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.|
|18:50-19:00 ♦ End of Day One – Closing Remarks|
|19:00-21:00 ♦ Cocktail Party – Social Networking Event|
|09:00-09:05 ♦ Welcome Note|
|09:05-09:30 ♦ 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.|
|09:30-10:00 ♦ 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.
|10:00-10:30 ♦ DEC112 – Austrian Text-To-112 Pilot|
|Wolfgang Kampichler, Frequentis AG, Austria|
|DEC112 (www.dec112.at) funded by netidee (www.netidee.at) 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.|
|10:30-11:00 ♦ Coffee Break|
|11:00-11:30 ♦ Extending Kamailio To Become A More Friendly SIP Routing Engine|
|Bart Coelmont, Netaxis Solutions, Belgium|
|This talk presents how we built our Session Routing Engine (SRE) around Kamailio, with an intuitive GUI that allows users to build very complex routing logics without knowing anything about Kamailio. It is even possible from within the GUI to test the logic constructed by the user. SRE is deployed as a class 4 routing engine into the core networks of IP telephony services. We will explore the architecture of the solution as well present some real life use cases.|
|11:30-11:45 ♦ (LT) WebPh.one – Connect Community Cellular Networks|
|Stefan Sayer, Lead Developer SEMS Project, Germany|
|Community Cellular (GSM) networks usually connect to the PSTN through SIP trunks that are connected via an Internet back-haul. For outgoing calls, PSTN termination costs have to be paid, and for incoming calls the caller usually tediously needs to enter the extension after connecting to a dial-in number, and it’s not possible to directly send text messages. This project uses WebRTC and Progressive Web App (PWA) technologies, implemented with the open source Kamailio+rtpEngine WebRTC gateway, SEMS, and an Angular app to connect people around the world on their smartphones and laptops directly to the users on the community cellular network for calls and texts. The open source webPh.one dialer is also an interesting technology base for other projects that need an App to connect to a SIP network. The talk gives a short intro to the two community cellular networks that have been connected (17 villages in Oaxaca, Mexico; PearlCel in Nicaragua), the problems that had to be solved and technologies used while doing so, and the solution this project created. It also gives our experience where today the limits of a pure PWA are, and shows our efforts to create docker-compose images for the infrastructure part.|
|11:45-12:30 ♦ Dangerous Demos|
|James Body, UK|
|12:30-13:30 ♦ Lunch Break|
|13:30-14:00 ♦ 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|
|Marius Corici, Dr., Deputy 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.|
|14:00-14:30 ♦ 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.
|14:30-15:00 ♦ 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?|
|15:00-15:30 ♦ 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.|
|15:30-16:00 ♦ Coffee Break|
|16:00-16:30 ♦ 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.|
|16:30-17:00 ♦ 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.|
|17:00-17:30 ♦ 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.
|17:30-17:50 ♦ 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, …
|17:50-18:00 ♦ End of Event – Closing Remarks|