Search results for “Bezeq

Speakers

January 6, 2013 at 8:07 pmCategory:

The list of submitted bios for confirmed speakers in alphabetic order.

Alan Duric

He co-founded Telio Holding ASA (formerly Telio Telecom AS, Oslo stock Exchange: TELIO) in 2004 and served as its Chief Technology Officer. Alan is an early pioneer of VoIP with over a 12 years of active contribution in multi modal communications, through his software development and standardization work (as a co-author and/or contributor to number of IETF, ETSI and ITU standards). As a developer and senior systems architect at Ericsson, he took part in the early deployment and development of the world’s largest VoIP networks such as deltathree. At Global IP Sound (creators of Skype´s audio), Alan was Senior Systems Architect, where he steered standardization efforts and was responsible for iLBC (internet Low Bitrate Codec), an IETF and CableLabs standard codec for VoIP. Alan is also Co-founder of Sonorit/Camino networks and was Director of its Board of Directors from its inception until it was acquired by eBay/Skype. Currently serves as Director of Sipfoundry.org BoD and is an advisor to a number of industry companies and financial institutions in M&A processes.

Alexandr Dubovikov

He is employed as Senior Voice Engineer for QSC AG, one of the major German voice and data providers. Alexandr holds a diploma in physics of Odessa State University. He has 20 years of experience in telecommunication techniques and has contributed to many OpenSource projects like FeeeSwitch, SER, Kamailio, SEMS, Asterisk, SIPP, Wireshark. Alexandr is the main developer of Homer SIP Capture project. He is also founder of IRC RusNet Network, one of the biggest national IRC networks in the world.

Alex Balashov

He is principal of Evariste Systems, an Atlanta-based consultancy specializing in open-source focused SIP service delivery solutions for wholesale and retail ITSPs and competitive carriers (CLECs). Alex is a member of Kamailio project management board, involved in application development and community engagement.

Andreas Granig

He holds a degree in IT/IT-Marketing from University of Applied Sciences in Graz, Austria. He is involved in VoIP development since 2000 and later co-founded Sipwise, a company dedicated to provide highly available and scalable SIP soft-switch appliances based on open-source building blocks like Kamailio and SEMS for medium to large-size ISPs/ITSPs. A well know product released by Sipwise is the SIP:Provider IP telephony platform. Andreas is member of management board of Kamailio project.

Anton Roman Portabales

He holds a degree in Telecommunication Engineering (University of Vigo) and finished the pre-PhD program in Telematics Engineering (University of Vigo). He started in VoIP In 2006   as developer for the Motorola IMS development group in Madrid. In 2008 joined Quobis  as VoIP engineer. Quobis is a privately-held Spanish company founded in 2006 dedicated to developing products and consulting services, based on open standards, in emerging unified communication technologies. Since 2011 he is the CTO of the company and actively participating in the expansion strategy to introduce Quobis products and servicies in emerging markets.

Björn Schwarze

From 1998-2001 he was working as venture manager at Lucent Technologies Bell-Labratories identifying value from new technologies. In 2002 he founded his own venture ADDIX Internet Services in the telecommunications industry, developing an Asterisk based PBX in 2005. Since 2012 he has been developing different business opportunities with WebRTC. Today he acts as CEO of Junghanns Communications and is Vice President of Schleswig-Holsteins IT Cluster Digitale Wirtschaft.

Carsten Bock

He is a member of the development and management teams of Kamailio project and founder of “ng-voice GmbH“, a company with focus on providing consulting services about Voice, IMS and Open-Source. Previously, he was working for Telefonica Germany in Verl & Hamburg on the deployment and operation of the new IMS-platform at Telefonica. He was among the leading technologists to develop and provide the first IP telephony solution as a real analog line replacement for Telefonica/O2 in Germany. Carsten has a Bachelor’s Degree in Computer Science from the University of Applied Sciences of Hamburg, Germany.

Dan Christian Bogos

He is the founder of ITsysCOM, experienced communications architect and VoIP specialist. Dan is a double graduate of Politechnica University, Timisoara, with post-graduate specialization in Communication Protocols and Software Development. For the past couple of years, he has focused mainly on Cloud Computing Technologies interoperability, subject of his PhD thesis research. A frequent and well-known contributor to the Open Source community, most noticeably being the co-founder of CGRateS Project, Dan is a firm believer in merging the very best production-ready software to create high-quality, scalable and cost-effective communications solutions.

Daniel-Constantin Mierla

He co-founded Kamailio in June 2005, aiming to build a solid SIP server project where openness to community and contributions has an important role, previously being core developer of SIP Express Router (SER) from its early beginning in 2002. He has a Master degree in Computer Science and Engineering from the Polytechnics University of Bucharest. His experience was accumulated working as consultant for Orange Romania, branch of French Orange mobile operator, and researcher in network communications at Fraunhofer Fokus Institute, Berlin, Germany.

His activity is done at Asipto, a company targeting to offer and build reliable services and solutions that benefit at maximum from Kamailio’s flexibility and features, sharing knowledge and expertise via professional trainings and consultancy. Daniel is leading the development of the Kamailio project and he is member of its management board.

David Duffett

He is Director, Worldwide Asterisk Community at Digium. David has been in telecoms for longer than he likes to remember, with positions in the Civil Aviation Authority (air-ground comms), a telco, a mobile provider and a computer telephony card manufacturer prior to getting in VoIP in general, and Asterisk in particular.
Having spent several years working in and around the Asterisk Community, David was delighted to be appointed as the Community Director in 2012. This involves upholding the interests of community members aswell as growing and enhancing the adoption of open source Asterisk. David’s favourite thing about Asterisk is that it can handle all sorts of technology (including WebRTC), without being contstrained to any one of them.

Dean Bubley

He is the Founder of Disruptive Analysis, an independent technology industry analyst and consulting firm. An analyst with over 20 years’ experience, he specializes in wireless, mobile, and telecoms fields, with further expertise across the broader technology industry. His present focus is on wireless technology, especially the evolution of mobile broadband, service provider business models, next-generation voice & messaging services, mobile device architecture & software, applications ecosystems and enterprise mobility. Dean provides clients with advice and analytical opinion on topics such as business models validation, technology innovation and go-to-market strategies, “addressable market sizing”, planning and due diligence.

Dragos Vingarzan

He graduated at the “Politehnica” University of Bucharest in February 2005 with a Diploma Thesis on the subject of service triggering in IMS, with a practical prototype implementation. During the 5-year study at the Faculty of Automatic Control and Computers he followed the Computer Engineering program on base software, compilers and computer networks. Between 1999 and 2004, he gained valuable hands-on experience as a full-time employee at various national ISP and Telecom Operators. During this period he acted as software analyst/developer as well as project development leader in various projects for network monitoring, provisioning, management, billing and VoIP deployments. During 2004 he worked on his Diploma Thesis at Fraunhofer FOKUS, which represented the first milestone of the Open IMS Playground.

Since 2005 he is employed as a Scientific Researcher as the same institute, where he continued his work into the Open Source IMS Project, as well into feasibility and performance studies on NGN architectures. He is an active member of various IMS working groups and leader of OpenEPC project.

Elena-Ramona Modroiu

She is a co-founder of Kamailio project, being also member of its management board. Her involvement in VoIP started by joining SER (SIP Express Router) in the spring of 2003, just after several months since the project was publicly released. She became one of the most active contributors, with key modules like avpops, diameter support, pdt, speeddial, uac, xlog. Elana-Ramona has a Master degree in Computer Science and Engineering from the Polytechnics University of Bucharest, Romania, completing the studies at Politechnics University of Valencia, Spain, and Fraunhofer Fokus Institute, Berlin, Germany.

Nowadays she works as VoIP and Kamailio consultant at ASIPTO, focusing on innovative solutions and integration of SIP and VoIP with web2.0. She manages Siremis Project – an open source web admin application for Kamailio – and has authored many tutorials about the Kamailio project, including Devel Guide, Pseudo-Variables Cookbook or Radius Integration.

Emil Ivov

He is the founder and current lead of the Jitsi project, a cross platform open source rich communication softphone (voice, video, instant messaging, presence, etc.). He is also involved with other initiatives like the icej4.org, libjitsi and JSIP projects, and contributes to IETF working groups like MMUSIC, AVTEXT and RTCWEB. Emil obtained his Ph.D. from the University of Strasbourg in early 2008, and has been focusing primarily on Jitsi activities ever since.

Henning Westerholt

He is one of the core developers of the Kamailio project and member of its management board. Henning is responsible at 1&1 Internet AG for the operation of their Voice over IP and DSL platform. This includes the maintenance and deploying of extensions of the Kamailio, Radius, ACS and order middleware systems of 1&1. He is in charge with one of the biggest VoIP deployments out there, using Kamailio (OpenSER) as core routing system: over 3 000 000 subscribers, over 7 000 000 phone numbers and more than 1 500 000 000 routed minutes per month.

Henning is a long term Linux user and contributer to several Open Source projects, participating to world wide events, presenting about Kamailio and open source. He holds a master degree in applied computer science from the University of Siegen.

James Aimonetti

He is the senior distributed systems engineer at 2600hz. What was a rather happenstance introduction to Erlang quickly developed into a huge passion that lasted for the past 5 years. Previously, he served as senior developer of insurance agency management software used by multiple independent insurance agents where he created a global address and social network service and an exemplary text-messaging service. James graduated from Principia College in 2004 with a degree in Computer Science and a minor in Mathematics. He is an avid country music fan.

James Body

He is the Head of Research and Development at Truphone, being nominated in 2007 among the top 20 Influencers in VoIP by VoIP news. James is well known a source of new, disruptive and innovative thinking is the telecomms space. Luckily, he managed to put these and a number of other disruptive skills to good use as a Regular officer in the British Army. Following 20 years service in the British Army, where he graduated with degrees in Electronic Engineering and Design of Information Systems, he became involved in the pioneering of UK based VoIP, setting up SIPCall and Gossiptel, leading UK SIP<>PSTN services. A founding member of the UK Internet Telephony Providers Association, he has consistently been involved with getting things done at the cutting edge of the European VoIP community. As Research Director at Truphone, he is responsible for new and exciting technologies, ranging from the handset back into the PSTN. An enthusiastic supporter of Open Source software, he sponsors the development and rapid deployment of some of the leading telecommunications applications.

Lukas Macura

He is VoIP and network specialist at CESNET, lecturer and network administrator at Silesian University in Opava. In the past, Lukas was chef developer at Phonyx, a PBX system for ISPs based on Asterisk. Today, he is member of BESIP team which develops embedded SIP communication servers based on Kamailio, Asterisk and OpenWrt. He aims to be “open-source extremist”, the motto that guides his activity being “Nothing is impossible”.

Markus Lindenberg

He is a technical consultant at GONICUS GmbH since 2008, where he is responsible for design, architecture and implementation of enterprise VoIP PBX systems based on Asterisk and FreeSWITCH. Apart from VoIP, Markus works on scaling HTTP and database based systems and deploying and operating Unix setups in data center environments both
as a software developer and systems administrator. He is using Linux professionally since 1998 and has been an avid Open-Source supporter and Pytonista ever since.

Peter Dunkley

He is the Technical Director at Crocodile RCS.  Peter graduated from The University of Edinburgh in 2000 with a BSc (Hons) in Computer Science. After graduation Peter worked on a PSTN switch developing signalling stacks for SS7, ISDN and similar protocols and creating advanced routing and service applications. Since 2005 he has worked mainly with SIP first leading a team developing a PSTN gateway and then managing the development of a SIP Application Server. Peter joined Crocodile RCS in September 2010 and has made numerous contributions to the Kamailio project (particularly in the areas of presence, WebSocket, MSRP, and SIP Outbound) since then. Peter is one of the authors of the MSRP over WebSocket draft (draft-pd-dispatch-msrp-websocket).

Randy Resnick

He is the producer of VoIP Users Conference (aka VUC). VUC is a weekly live discussion about VoIP, SIP and all kinds of telephony-related topics. The conference has been running for nearly six years, happening every Friday at 12:00 noon US Eastern Time. Many Kamailio developers are frequent participants at VUC, several of them were invited guests in the past, presenting the project and typical use cases.

Simon Woodhead

He is founder and Managing Director of Simwood eSMS Limited, a wholesale VoIP enabler. Simwood operates a national UK IP network, VoIP stack and SS7 interconnects, all of which can be leveraged by customers using its leading API. In 1996 Simwood developed ‘eSMS’, the world’s first global gateway between the Internet and Mobile phones, enabling cross-network SMS and SMS<>e-mail. Prior to that Simon worked in finance and was the youngest qualified member of the Securities Institute at the time and through subsequent start-ups was named Barclay’s Young Business Person of the Year. No longer young he combines commercial experience with deep technical knowledge and brings a unique perspective to today’s challenges. Outside of work he has been a Mountain Rescue Team member for over 10 years and played a part in hundreds of incidents, rounding his outlook on life and shaping his approach to leadership.

Stefan Wintermeyer

He is CEO and founder of AMOOMA, author of an Asterisk book ( http://www.the-asterisk-book.com ) and the creator of Gemeinschaft PBX. Before working with VoIP solutions he was CEO at OTRS which does open-source trouble-ticket-systems. Stefan started his open-source career as a vice president for SuSE Linux in 1998. He is a very open-source minded guy who even published his book under the FDL. AMOOMA is a German company which offers mainly VoIP consulting in Europe.

 

Thomas Magedanz, Prof. Dr.

He is professor of the chair for Next Generation Networks (AV – Architektur der Vermittlungsknoten in German) in the electrical engineering and computer sciences faculty of the Technische Universität Berlin, Germany, where he is educating masters and PhD students in the fields of multimedia Service Delivery Platform technologies on top of converging fixed and mobile networks, Next Generation Networks, and the Future Internet. In addition, he leads the Next Generation Network Infrastructures (NGNI) Competence Center at Fraunhofer Institute FOKUS in Berlin, Germany, where he is responsible for the performance of major international R&D co-operations and related projects in the context of next generation telecommunications infrastructures. In this context he is a globally recognized pioneer of the development and delivery of advanced network and service technology testbeds and related software tools in the fields of Next Generation Networks and the emerging Future Internet for both academia and industry. Well known examples include the Open IMS Playground and the Open SOA Telco Playground.

Uri Shacked

He is employed as senior architect for added value services at Bezeq, the largest Israeli telecom company. Uri holds a diploma in Electric Engineering, M.B.A and he has 10 years of experience in developing and deploying large scale added value systems based on open source and other vendors platforms. For the last year, he is responsible for the deployment and development of the largest IN system in Israel based on Kamailio applications. Uri is 38 years old, a father of two and addicted to kite surfing.

Victor Pascual Avila

He is a technology strategist, holding a Master’s Degree in Telecommunication Engineering. Victor is involved in standardization activities, mainly focusing on SIP, WebRTC, and Diameter, co-chairing the STRAW (Sip Traversal Required for Applications to Work) Working Group at the IETF, serving as Expert Reviewer for the European Commission and several technical or research conferences.

Vladimir Broz

He is Director of Engineering at Frafos. Vladimir holds a MSc. degree from CVUT, Prague. Before joining Frafos, he was responsible for engineering of SS7 and SIP products in Sitronics and Tekelec. Being one of the Frafos Day-1 employees, Vladimir is responsible for company’s ABC SBC product from inception to delivery.

Schedule

November 1, 2012 at 8:27 pmCategory:

Please note that there can still be updates to the timelines of the schedule. Details about accepted speakers are presented in a dedicated page.

 

Day 1: April 16, 09:00 – 18:00

 

09:00 – 09:05 – Welcome!

09:05 – 09:15 – Kamailio – State of the Project

Speaker: Daniel-Constantin Mierla (Asipto, co-founder of Kamailio)

Description: the summary of what happened recently in the development and the environment around Kamailio project, its newest features and use cases. An overview of the development and management process, how you can get involved and help the project evolve the way you would like.

09:15 – 09:25 – Kamailio – Business Report

Speaker: Elena-Ramona Modroiu (Asipto, co-founder of Kamailio)

Description: how and where is Kamailio used, what are the successful business models and the trends to watch for future communication systems, statistics on performances and unusual use cases

09:30 – 10:00 – The future of NG RTC platforms and services

Speaker: Thomas Magedanz, Prof. Dr. (Professor at Technische Universität Berlin and Head of NGNI Competence Center at FhG Fokus)

Description: 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 and machine to machine communications. The talk gives an overview of current research directions that can impact the way we communicate tomorrow.

10:05 – 10:35 – Multimodal Real Time Communications – the impact of becoming omnipresent and continuous

Speaker: Alan Duric (Co-founder Telio Holding ASA)

Description: outlook and analysis of the trends in (Real Time) Communications industry – impact on the existing business models – role of OSS (Open Source Software) and Kamailio in shaping the RTC landscape

10:35 – 11:00 – Coffee Break

 

11:00 – 11:30 – WebRTC and VoIP: bridging the gap

Speaker: Victor Pascual Avila (Technology Strategist)

Description: A look at whether it is possible and how to blend classic VoIP with WebRTC, what are the common elements and incompatibilities between the two technologies.

11:35 – 12:05 – Start Your Own Telephony Service with SIP:Provider

Speaker: Andreas Granig (CTO of Sipwise)

Description: how to become a telephony service provider in less than one hour using SIP:Provider platform. Bundled as out of the box system for running a VoIP operator service, SIP:Provider offers everything needed to take care of security, scalability, media services and billing. Built using many open source applications, wrapped by web administration and customer portals, SIP:Provider gives the valuable flexibility, pretty unique in the telephony market, to add over the top new features without vendor lock.

12:10 – 12:40 – Asterisk Update

Speaker: David Duffet (Director, Asterisk Worldwide Community)

Description: Although Asterisk has now been around for more than 12 years, Digium and the Community continue to refine and optimise it, and add support for new features and technologies. This is what keeps Asterisk as the most popular and well established open source communications platform in the world. Attend this session to see some of the latest additions to Asterisk, and to hear about what’s coming down the line in Asterisk 12.

12:40 – 13:40 – Lunch Break

 

13:40 – 14:10 – Large IP Telephony Deployment Scenarios in Real World

Speaker: James Body (Head of research and development at Truphone)

Description: how to get most of Kamailio’s scalability, flexibility and reliability and its role in building innovative real time communication services. A trip through an over decade of deployments to disturb the telecom services.

14:15 – 14:45 – SIP and MSRP over WebSocket in Kamailio

Speaker: Peter Dunkley (CTO of Crocodile RCS)

Description: bridging the present and the future of communications using SIP – call your contacts from browser via websockets, text chatting, advertising presence states and transferring files, at the same time, using same device.

14:50 – 15:20 – Deploying IMS Platforms

Speaker: Carsten Bock (CEO of NG Voice)

Description: experiences while deploying IMS and the benefits brought by relying on Kamailio and open source for building flexible IMS platforms. Not many took the challenges of developing IMS extensions, even fewer could keep it going, this talk comes to present that IMS implementation is mature, with key deployments world wide, at a moment when this technology has to become the core of 4G+ networks.

15:20 – 15:50 Coffee Break

 

15:50 – 16:20 – Embedding Kamailio into BESIP and OpenWrt

Speaker: Lukas Macura (Lecturer at Silesian University in Opava, VoIP and Network Specialist at CESNET)

Description: The aim of the BESIP project is the development and implementation of open source based embedded SIP communication server with an easy integration into the computer network. BESIP should be able to act as a small PBX, edge proxy or SBC. We used Kamailio as core SIP router and our goal is to configure it using NETCONF protocol. We ported several packages into OpenWrt (e.g., Kamailio3, Asterisk11) and we prepared experimental feeds (e.g., kamailio4, mediaproxy), which are waiting to be accepted by community. Presentation will be focused to explain basic goals of BESIP project and interaction with Kamailio.

16:25 – 16:50 – Roll Your Own VoIP Cloud in 30 Minutes or less

Speaker: James Aimonetti (Distributed Systems Architect at 2600hz)

Description: Discussion of the 2600hz implementation of Kamailio within the Kazoo stack, the features of the platform, how to get it up and running quickly and mange it in production.

16:55 – 18:00 – VUC RTC and RCS Panel – Open Discussions

Moderator: Randy Resnick (Producer of VoIP Users Conference)

Participants: James Body, Olle E. Johansson, Simon Woodhead, Thomas Peterseil, Paul Fermedge, Alex Kinch — actually everyone in the room is invited to join the discussion

Description: debate about present and future of real time communication technologies and services

18:00 – 18:15 – Closing Session

 

Evening – Cocktail and Social Networking Party

Time and Location: 18:30 – 22:00 in the Atrium Area inside the conference building.

Day 2: April 17, 09:00 – 18:00

09:00 – 09:05 – Welcome!

09:05 – 09:25 – SEMS – Adaptive Session Border Controller

Speaker: Vladimir Broz (Director of Engineering at Frafos)

Description: Frafos, the company behind SIP Express Media Server (SEMS), created the ABC SBC product on top of the open source media server. The SBC primarily aims to enable standard-based SIP communications with advanced business logic, allowing programmers and integrators to build telephony applications rapidly, avoiding to load their apps with abundant protocol details.

09:30 – 10:00 – Packet Capture for Security Risk Prevention in Telephony

Speaker: Alexandr Dubovikov (Founder of Homer Project)

Description: how to capture and aggregate signaling traffic from several SIP server nodes and generate reports for technical (troubleshooting) and business (marketing) needs. As the business of small and big telephony operators grows, so does the risk of fraud and sophisticated security attacks against their infrastructure – CDRs are no longer a reliable source for timely fraud detection and revenue protection. Leveraging centralized packet capture and parsing (based on Kamailio and award-winning HOMER and its powerful API) simple and complex scripts can be programmed to detect and react to fraud events and other kinds of attacks in real-time. Actions can be as simple as an email, or as complex as re-configuring routing, banning and spoofing back crafted responses to the attackers by analyzing and profiling the telephony traffic.

10:05 – 10:35 – Jitsi – the softphone for rich communications

Speaker: Emil Ivov (Founder of Jitsi Project)

Description: During the past ten years Jitsi has traveled a long way from a proof of concept call application to arguably the most complete and feature rich FOSS communicator on the planet. Emil will walk us through the various flagship features of Jitsi such as high quality audio/video calling, conferencing, call encryption, file transfer, desktop sharing, instant messaging and many others. The talk will approach also Jitsi’s sister projects: libjitsi, a component that allows building communications client or server-side applications for SIP, XMPP and WebRTC; and the newborn Jitsi Videobridge which makes it possible for anyone to deploy a hangout-like video conferencing solution.

10:35 – 11:00 – Coffee Break

 

11:00 – 11:30 – VoIP Security Tools

Speaker: Anton Roman Portabales (CTO Quobis)

Description: What are the most common attacks and how to test if your network is protected with existing tools. Fraud is still biggest business threat in VoIP, learn how to stay ahead of attackers and prevent being a victim that can ruin yourself.

11:35 – 12:05 – Managing Large Telephony Systems

Speaker: Henning Westerholt (Head of IT Operations Internet Access & Communications at 1&1 Internet AG)

Description: remarks from years of operating a telephony platform serving millions of subscribers and routing billions of voice minutes per month. How open source can leverage large companies to be competitive and keep the pace on the market with innovative services

12:10 – 12:40 – An Analysis of WebRTC Impact on Future Communications

Speaker: Dean Bubley (Founder of Disruptive Analysis)

Description: Status & forecasts for adoption of WebRTC in fixed & mobile – Key early use-cases and market drivers for WebRTC – How WebRTC fits with (or against) SIP – Implications for VoIP developers, telcos & enterprises

12:40 – 13:40 – Lunch Break

 

13:40 – 14:10 – Case study – Using Kamailio as IN SIP application server at Bezeq

Speaker: Uri Shacked (Senior Architect and Bezeq)

Description: Bezeq is the largest Israeli telephony operator. The IN system at Bezeq carries all the number translation services, including features like business IVR, call screening, busy rerouting, no answer rerouting, music on hold and other features.  Using Kamailio, Bezeq managed to deploy a telco grade NTS and IVR system serving 2 million calls per day. The system is resilient enough to also support 911 and other lifesaving 1800 services.  In this presentation we will demonstrate how Bezeq leveraged Kamailio’s advantages such as the ability to manipulate headers and call flows, store large amount of data in memory, carry load with high performance, stability and fast implementation in order to meet the project’s tight schedule and goals.

14:15 – 14:45 – VoIP with Free Software in Enterprises

Speaker: Markus Lindenberg (consultant at Gonicus)

Description: The fast changing requirements and the integration of telecommunication and information technology infrastructures open up new opportunities for Open-Source-Software to set foot in the telephony infrastructures of large enterprises. This talk will address the experiences in a project to replace a classic PBX by a new VoIP infrastructure within an insurance company. Aside from technologies used (like Kamailio or Asterisk) the presentation will also explain motivation for such a project as well as talk about implementation and operation of an Open-Source-VoIP-solution for more than 3.500 users.

14:50 – 15:20 – WebRTC Business Opportunities

Speaker: Björn Schwarze (CEO of  Junghanns Communications)

Description: WebRTC allows browsers to act as softphones, by activating a piece of Javascript we can contact any other WebRTC user in the world. The talk will look at WebRTC technology from business perspective, identifying what are the new opportunities for rich communication services.

15:20 – 15:50 Coffee Break

 

15:50 – 16:20 – Practical deployments in the North American PSTN Sphere

Speaker: Alex Balashov (principal of Evariste Systems)

Description: Key considerations of PSTN-oriented deployments in the North American telephony environment, viewed from the standpoint of regulation, standard information services (number portability queries, calling name/directory queries, law enforcement intercept, etc.). Enriched by Evariste’s experience as a nearly 100% Kamailio-oriented shop, the general trends in the demands (of Kamailio) of North American ITSP and CLEC customers will be discussed.

16:25 – 16:55 – CGRateS – Carrier grade real-time charging

Speaker: Dan Christian Bogos (founder of ITsysCOM)

Description: Internet Telephony has evolved over the years from simple office PBX to clustered voice services. CGRateS project was born out of today’s market demand for performance and scalability when it comes to Real-Time rating. In this talk Dan will briefly introduce CGRateS’s particularities together with the current state of the project and features on the road-map.

17:00 – 17:30 Load Balancing Load Balancers

Speaker: Daniel-Constantin Mierla (Asipto, co-founder of Kamailio)

Description: is your subscriber base growing fast? You keep adding new servers behind load balancer. What happens when you reach hardware limits of load balancers? This is a presentation of how to scale the load balancers farm.

17:30 – 18:00 – Panel – Future of Kamailio Project – Open Discussions

Participants: Kamailio Developers

18:00 -18:10 – Closing Session