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{
"SPEAKERS":[
{
"NAME":"Bogdan-Andrei Iancu",
"SPEECHTITLE":"Key Notes - An introduction in OpenSIPS 2.4",
"SPEECHABSTRACT":"OpenISPS 2.4 is focused on clustering and distribution, but not limited to that. A walk through what this new OpenSIPS version has to over, like SIPREC, distributed user location, full anycast support, distributed presence and many more.",
"TEXTIMG":"http://i.imgur.com/c0q8TjB.png",
"TEXTTITLE":"Bogdan-Andrei Iancu",
"COMPANY":"OpenSIPS Project",
"DESC":"Bogdan is the OpenSIPS project founder with an experience of 15+ year in the SIP world. Practicing the symbioses between managing the Open Source project and building commercial products around OpenSIPS, gives the best results in producing a viable SIP Server software for the real-life needs.",
"TWITTER":"https://twitter.com/@bogdan_iancu",
"POSITION":"Founder"
},
{
"NAME":"David Duffett",
"SPEECHTITLE":"Asterisk Update",
"SPEECHABSTRACT":"Attend this session to hear the latest news from the Asterisk project, now at version 15.x",
"TEXTIMG":"https://secure.gravatar.com/avatar/c104436f064ea8d6eeb3038bf0127c1a?s=500",
"TEXTTITLE":"",
"COMPANY":"Digium Inc.",
"DESC":"Asterisk is the most popular open source IP telephony engine in the world today. With the addition of adaptors (that come in a range of shapes and sizes) Asterisk can be connected to most legacy PSTN interfaces like PRI, BRI, and FXO connections in addition to IP telephony protocols SIP and IAX2, which are natively supported.",
"TWITTER":"https://twitter.com/@test",
"POSITION":"Director, Worldwide Asterisk Community"
},
{
"NAME":"Jonas Borjesson",
"SPEECHTITLE":"In the Trenches of a Globally Spanning SIP Network",
"SPEECHABSTRACT":"In this talk I will share my experience of running a globally spanning SIP network deployed on Amazon Web Services. We will be looking into the cause of some incidents we have seen over the years and what we learned from them. In particular, we will look into loops, registration issues, when to fail-over, when not to fail-over, network partitions and more. We will also look into some best-practices when all your SIP elements are deployed on AWS.",
"TEXTIMG":"https://media-exp2.licdn.com/media/p/8/000/1d6/1e7/13d7e3d.jpg",
"TEXTTITLE":"Jonas Borjesson",
"COMPANY":"Twilio",
"DESC":"For over a decade, Jonas Borjesson has built everything from complete SIP-based VoIP and presence solutions to RTP relays and audio recorders. He has taken part in building SIP stacks from scratch and was part of the JSR359 Expert Group, defining next gen Java based SIP container. In his spare time, he builds SIP stacks.",
"TWITTER":"https://twitter.com/@borjessonjonas",
"POSITION":"Software Architect"
},
{
"NAME":"Alejandro Rios Peña",
"SPEECHTITLE":"Curated \"Cloud Design Patterns\" for Opensips-based platforms",
"SPEECHABSTRACT":"Cloud-based contact center solutions have many challenges, but also common roads to success. With 15+ years of experience, Livevox manages a volume of 9 billion calls a year. In this session we'll share a curated selection of Cloud Design-Patterns (CDP) that you can apply on your own designs.",
"TEXTIMG":"https://media-exp2.licdn.com/media/AAEAAQAAAAAAAAjEAAAAJDIxOTUwYzllLWUxYTktNDllYi04NzE3LTQxM2U5MjA4ZTcxZg.jpg",
"TEXTTITLE":"Alejandro Rios Peña",
"COMPANY":"Livevox Inc.",
"DESC":"Innovation Facilitator, with both development and product design management backgrounds, whose purpose is facilitating personal development of passionate people by taking to reality innovative high-impact ideas. \nHe likes volcano summits, half marathons, and oil painting. His portfolio of projects go from Cloud Call Center and PBX unified communications based on Open Source and voice over IP, through several Twitter mashups, Open and Social Innovation, Design-thinking for startups, Agile practices combined with PMI, to an alternative photo magazine and his family organic coffee growing farm certified by RainForest Alliance.\nPortfolio: http://alerios.org/\nMedellin (COL) / San Francisco (US)",
"TWITTER":"https://twitter.com/@alerios",
"POSITION":"Senior Manager of Telecom Software Engineering"
},
{
"NAME":"Eric Tamme",
"SPEECHTITLE":"Supporting mobile clients in a distributed SIP infrastructure with OpenSIPS",
"SPEECHABSTRACT":"Mobile clients face challenging network conditions. This talk focuses on RFC5626, SIP Outbound, implementation and asynchronous operations in OpenSIPS to support WebRTC, and native, mobile clients in a distributed SIP signalling infrastructure.",
"TEXTIMG":"https://avatars0.githubusercontent.com/u/21685?s=460&v=4",
"TEXTTITLE":"Eric Tamme",
"COMPANY":"OnSIP",
"DESC":"Principal engineer at OnSIP, federated SIP evangelist, and phone phreak.",
"TWITTER":"",
"POSITION":"Principal Engineer"
},
{
"NAME":"Rogelio Perez",
"SPEECHTITLE":"Telephony Applications HA using Container Routing",
"SPEECHABSTRACT":"There are multiple ways of implementing High Availability for telephony applications, however we think our unique approach might be of interest for those already using containerized FOSS. We'll demonstrate how to leverage container routing features to create applications with built-in HA.",
"TEXTIMG":"https://media.licdn.com/media/AAEAAQAAAAAAAAw4AAAAJDU5ZGYxMmYxLWYyZTQtNDY4ZS1hMGZkLWFhNDY2ZjY5OTc3ZQ.jpg",
"TEXTTITLE":"Rogelio Perez",
"COMPANY":"Telnyx LLC",
"DESC":"Rogelio works as Voice Engineers at Telnyx, where they are in charge of building and maintaining the Core Telephony Engine. Rogelio has over 15 years of experience with SIP and VoIP technologies. He graduated from Universidad Blas Pascal in Argentina with a degree in Engineering, Telecommunications.",
"TWITTER":"https://twitter.com/@rogelioperez000",
"POSITION":"Voice Engineer"
},
{
"NAME":"Ramon Torres",
"SPEECHTITLE":"Telephony Applications HA using Container Routing",
"SPEECHABSTRACT":"There are multiple ways of implementing High Availability for telephony applications, however we think our unique approach might be of interest for those already using containerized FOSS. We'll demonstrate how to leverage container routing features to create applications with built-in HA.",
"TEXTIMG":"assets/images/speakers/Ramon_Torres.jpg",
"TEXTTITLE":"Ramon Torres",
"COMPANY":"Telnyx LLC",
"DESC":"Ramon works as Voice Engineers at Telnyx, where they are in charge of building and maintaining the Core Telephony Engine. Ramon Torres is a VoIP engineer with 4 years of experience. He is very passionate about open source systems and exploring new technologies. He graduated from Universidad Politecnica de Madrid in Telecommunications Engineering and has a masters in Information Technology from IIT Chicago.",
"TWITTER":"",
"POSITION":"Voice Engineer"
},
{
"NAME":"Giovanni Maruzzelli",
"SPEECHTITLE":"FreeSWITCH clustering with OpenSIPS (done well)",
"SPEECHABSTRACT":"How to scale so that you can handle your growing SIP traffic by adding more FreeSWITCH machines, while keeping the functionality of a single machine? Let's review the best practices and the latest techniques for load balancing, registrations, presence, call transfer, queues, conferences, etc",
"TEXTIMG":"https://secure.gravatar.com/avatar/9a8d3098ee2abd638e99eae52de1d2b1?s=500",
"TEXTTITLE":"Giovanni Maruzzelli",
"COMPANY":"OpenTelecom.IT",
"DESC":"Giovanni Maruzzelli (OpenTelecom.IT) is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. He's a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC, Kamailio and OpenSIPS. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies - www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank - IFC. Since 2005 he's based in Italy, and serves ICT and Telco companies worldwide.",
"TWITTER":"https://twitter.com/@gmaruzz",
"POSITION":""
},
{
"NAME":"Flavio Goncalves",
"SPEECHTITLE":"SIP in the wild!",
"SPEECHABSTRACT":"Wholesale ITSPs have a challenge to support any SIP client to any SIP termination. Very often, failures in SIP signaling generate problems on accounting, and cause hanged and/or disconnected calls. In this presentation you will learn how to mitigate some of these problems.",
"TEXTIMG":"http://i.imgur.com/5PS7U7r.png",
"TEXTTITLE":"Flavio Goncalves",
"COMPANY":"SIPPulse",
"DESC":"",
"TWITTER":"https://twitter.com/@SipPulse",
"POSITION":"CTO"
},
{
"NAME":"Lorenzo Miniero",
"SPEECHTITLE":"Troubleshooting and monitoring Janus: a HEPIC journey!",
"SPEECHABSTRACT":"This talk will present the efforts devoted by Meetecho and QXIP in developing a shared solution to monitor, debug and troubleshoot issues in Janus sessions via HEPIC. Use cases with real data from existing deployments will be presented as well.",
"TEXTIMG":"http://i.imgur.com/4rfCiZB.png",
"TEXTTITLE":"Lorenzo Miniero",
"COMPANY":"Meetecho",
"DESC":"Lorenzo Miniero is the chairman and co-founder of Meetecho, a company providing consulting services on everything related to real-time multimedia, while also regularly providing streaming and remote participation services for well known events around the world (e.g., IETF and ACM). Lorenzo received his degree and Ph.D. at the Computer Science Department of the University of Napoli Federico II, where he started working on multimedia conferencing and met the colleagues with whom he co-founded Meetecho as an academic spin-off. He is an active contributor to the Internet Engineering Task Force (IETF) standardization activities, especially in the framework of real-time multimedia applications. He is most known as the author of the Janus WebRTC Server, an open source WebRTC server-side implementation.",
"TWITTER":"https://twitter.com/@elminiero",
"POSITION":"Chairman"
},
{
"NAME":"Vladut-Stefan Paiu",
"SPEECHTITLE":"Unusual and Unorthodox OpenSIPS use-cases",
"SPEECHABSTRACT":"While OpenSIPS is the 'go to' general purpose SIP server, it can fit perfectly in unusual use cases within a VoIP platform, not necessarily processing SIP traffic.\nAlso, OpenSIPS is a perfect fit for some unorthodox (close to blackhat) SIP scenarios. How to implement & how to mitigate against them.",
"TEXTIMG":"https://media-exp2.licdn.com/media/p/3/000/0ca/051/38c0356.jpg",
"TEXTTITLE":"Vladut-Stefan Paiu",
"COMPANY":"",
"DESC":"",
"TWITTER":"",
"POSITION":""
},
{
"NAME":"Lorenzo Mangani",
"SPEECHTITLE":"HOMER & OpenSIPS",
"SPEECHABSTRACT":"HOMER & OpenSIPS: A Match made in Heaven!",
"TEXTIMG":"http://i.imgur.com/XJS0Ihj.png",
"TEXTTITLE":"Lorenzo Mangani",
"COMPANY":"QXIP",
"DESC":"Lorenzo is the Founder and CEO of QXIP BV, the Amsterdam based R&D developing HOMER, HEPIC, HEP and many other Monitoring and Packet Capture projects.",
"TWITTER":"https://twitter.com/@qxip",
"POSITION":"CEO"
},
{
"NAME":"Dan Christian Bogos",
"SPEECHTITLE":"LCR with real-time number portability using OpenSIPS and CGRateS",
"SPEECHABSTRACT":"Modern Telecommunications require operators dealing with large databases of numbers ported from original source in order to comply with industry regulations. Moreover choosing to ignore this part can bring serious financial losses due to incorrect termination cost calculations.",
"TEXTIMG":"http://i.imgur.com/2D2AjWr.png",
"TEXTTITLE":"Dan Christian Bogos",
"COMPANY":"ITsysCOM GmbH",
"DESC":"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. 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.",
"TWITTER":"https://twitter.com/@danbogos",
"POSITION":"Founder"
},
{
"NAME":"Michael Mavroudis",
"SPEECHTITLE":"Got Training?",
"SPEECHABSTRACT":"Become a VoIP master of the universe thought the acquisition of knowledge via FreeSWITCH certified training. Leverage the opportunity to learn the tips, tricks, and undocumented gems to rocket towards the FreeSWITCH event horizon peer inside and follow the reply_route back with a certification. Milk is optional. Belgium beer is required!",
"TEXTIMG":"https://media-exp2.licdn.com/media/p/2/000/01c/3c2/2946a2d.jpg",
"TEXTTITLE":"Michael Mavroudis",
"COMPANY":"FreeSWITCH",
"DESC":"",
"TWITTER":"",
"POSITION":""
},
{
"NAME":"Razvan Crainea",
"SPEECHTITLE":"Full Anycast support at the edge of your platform using OpenSIPS 2.4",
"SPEECHABSTRACT":"This talk will present the new mechanisms added in OpenSIPS 2.4 that allow you to build a fully flavored anycast setup using OpenSIPS at the edge of your network. In this presentation I will show you the problems that appear when using SIP over anycast networks and how they are solved using OpenSIPS and the new replication mechanism for the transaction layer.",
"TEXTIMG":"http://i.imgur.com/YbMlg4m.png",
"TEXTTITLE":"Razvan Crainea",
"COMPANY":"OpenSIPS Project",
"DESC":"Răzvan has joined the OpenSIPS project in the summer of 2010, as a junior C developer. Soon after familiarizing with the project and the VoIP world, he started working on key features such as CDR accounting, media proxy-ing, NAT handling, WebRTC, etc. After a couple of years he became one of the Core Developers of OpenSIPS, involved in the design and implementation of new features, as well as actively maintaining the existing ones. He is also in charge with preserving OpenSIPS compatibility with both newer and older Operating Systems. Throughout the years he was engaged in various OpenSIPS-based platforms development, gathering vast experience in VoIP development.",
"TWITTER":"https://twitter.com/@razvancrainea",
"POSITION":"Core Developer"
},
{
"NAME":"Norm Brandinger",
"SPEECHTITLE":"Using the mid_registrar module along with registration redirection",
"SPEECHABSTRACT":"A discussion of the features and functions of mid_registrar module will take place. Live examples using traffic generated by SIPP will be demonstrated. The discussion will continue by showing how redirection can be used to further reduce load on traditional registrars",
"TEXTIMG":"assets/images/speakers/Norm.png",
"TEXTTITLE":"Norm Brandinger",
"COMPANY":"Vonage",
"DESC":"A strategic technology leader I leverage my expertise in software and hardware to deliver value including revenue generation and customer acquisition. I have built systems and architected complex communication solutions to drive business growth. My work includes creating business plans that apply emerging technologies to launch VoIP provider platforms.",
"TWITTER":"",
"POSITION":"Call Processing Architect"
},
{
"NAME":"Tymothy Meyerhoff",
"SPEECHTITLE":"Supporting Blue/Green Deployments and Autoscaling in AWS with clustered OpenSIPS",
"SPEECHABSTRACT":"Let's discuss the versatility of OpenSIPS clustering, specifically how it can enable seamless blue/green deployments and help support an auto scaling infrastructure in AWS",
"TEXTIMG":"assets/images/speakers/tmeyerhoff.png",
"TEXTTITLE":"Tymothy Meyerhoff",
"COMPANY":"Vonage",
"DESC":"I've worked at Vonage for 9 years supporting residential, carrier, and business services and now Vonage Business Cloud. I have worked extensively with OpenSIPS for the past few years and happily continue to develop, deploy and support a variety of solutions around it.",
"TWITTER":"",
"POSITION":"DevOps Software Engineer"
},
{
"NAME":"Maxim Sobolev",
"SPEECHTITLE":"SIP in the age of IoT",
"SPEECHABSTRACT":"WebRTC is good but is it good enough to utilize all wonders of the RFC3261 and RFC3550 combined?! We are not quite sure about that, there might be bigger emerging markets to go after.",
"TEXTIMG":"http://sobomax.sippysoft.com/2efb44_9d016a626c1b0fded8397399c5401f6b.png",
"TEXTTITLE":"Maxim Sobolev",
"COMPANY":"Sippy Software, Inc.",
"DESC":"Long term OpenSIPS contributor, open source aficionado. Author and maintainer of few open source projects.",
"TWITTER":"https://twitter.com/@sobomax",
"POSITION":""
},
{
"NAME":"Alex Goulis",
"SPEECHTITLE":"Creating a large, scalable, and redundant voicemail cluster using OpenSIPS and FreeSWITCH",
"SPEECHABSTRACT":"This discussion/how-to will focus on building a large scalable voicemail system using opensips as the front end and free switch as the clustered voicemail backend. Our discussion will include methods and modules used with opensips to automatically maintain the availability of servers in the cluster. We'll also cover balancing of traffic across all the servers.",
"TEXTIMG":"http://i.imgur.com/FvHRqNc.png",
"TEXTTITLE":"Alex Goulis",
"COMPANY":"RateTel Inc",
"DESC":"",
"TWITTER":"https://twitter.com/@ratetel",
"POSITION":""
},
{
"NAME":"Liviu Chircu",
"SPEECHTITLE":"Distributed User Location Models with OpenSIPS Cluster",
"SPEECHABSTRACT":"Distributing the SIP user location across multiple data centers is a key problem that any large-scale VoIP provider must solve. Moreover, they must also take into account concepts such as traversing the NAT in front of contacts, keeping these NAT pinholes alive or being able to easily scale the service up/down while also ensuring a maximal uptime for it. Most state-of-the-art solutions revolve around the usage of a geo-distributed database coupled with some OpenSIPS scripting in order to route calls to the datacenter on which the targeted user is binded. Using this as a starting point, each solution is then customized according to the needs of the provider. In this presentation, the discussion will focus around the designs that went into OpenSIPS 2.4 with the purpose of exposing generic and intuitive script primitives in order to build robust SIP user location services that solve one or more of the above core problems.",
"TEXTIMG":"http://i.imgur.com/LmSk7gI.jpg",
"TEXTTITLE":"Liviu Chircu",
"COMPANY":"OpenSIPS Project",
"DESC":"Full time VoIP engineer / software developer. Open-source enthusiast. Passionate about proper software design and appropriate selection of tools for the given task while reusing existing code as much as possible.\nLiviu Chircu has been involved with OpenSIPS and the VoIP world for almost 6 years now. He is a software developer and VoIP consultant for OpenSIPS Solutions. He has extensive experience with OpenSIPS inner-workings, as well as deploying and troubleshooting various SIP setups involving OpenSIPS.\nHis speaker experience includes talks at the OpenSIPS Summits, as well as FOSDEM, AstriCon and ClueCon.\nLiviu also contributes with tech articles on the OpenSIPS blog.",
"TWITTER":"https://twitter.com/@liviuchircu",
"POSITION":"Core Developer"
},
{
"NAME":"Pete Kelly",
"SPEECHTITLE":"Lessons learned from working with opensips every day for the past 10 years",
"SPEECHABSTRACT":"",
"TEXTIMG":"https://user-images.githubusercontent.com/1423657/31589533-a7cf4f5e-b203-11e7-8a52-bf2f2cfdf72a.png",
"TEXTTITLE":"Pete Kelly",
"COMPANY":"SourceVox",
"DESC":"",
"TWITTER":"https://twitter.com/@p3k4y",
"POSITION":""
},
{
"NAME":"Guillaume Montassier",
"SPEECHTITLE":"OpenSIPS Platform of a telecom carrier",
"SPEECHABSTRACT":"The presentation proposes to introduce to the audience the new OpenIP telecom platform, which is base on Opensips solution.",
"TEXTIMG":"",
"TEXTTITLE":"Guillaume Montassier",
"COMPANY":"OpenIP",
"DESC":"My name is Guillaume MONTASSIER. I'm an R&D engineer at OpenIP and in charge of the development of the Telecom platform.",
"TWITTER":"",
"POSITION":"R&D engineer"
},
{
"NAME":"Vasilii Rogin",
"SPEECHTITLE":"Building determined and fully controlled PBX with nodejs and asterisk",
"SPEECHABSTRACT":"I want to describe our method of building QTel PBX system. We are not using asterisk's dialplan at all (almost), every incoming call is passed into FastAGI, and every outgoing leg is created via AMI Originate.\nSimple calls, queues, IVR - everything is controlled by our nodejs application.",
"TEXTIMG":"https://papercallio-production.s3.amazonaws.com/uploads/user/avatar/21956/I0QMEeC8yWM_1_.jpg",
"TEXTTITLE":"Vasilii Rogin",
"COMPANY":"Qlave AB, Sweden",
"DESC":"I am working with Qlave AB, we are making cloud pbx system QTel. I did backend part and some of frontend.",
"TWITTER":"",
"POSITION":""
},
{
"NAME":"Iñaki Baz Castillo",
"SPEECHTITLE":"Building multi-party video conference applications with mediasoup",
"SPEECHABSTRACT":"\"Our hyper cool video conference app requires Chrome browser\"\nTired of reading that? Are you building your own multi-party WebRTC app but can't even figure out how to deal with the different API and feature set of each browser?\nWe feel your pain. That's why we started the mediasoup project.",
"TEXTIMG":"https://mediasoup.org/images/ibc.jpg",
"TEXTTITLE":"Iñaki Baz Castillo",
"COMPANY":"",
"DESC":"I'm passionate about new technologies, Open Source, WebRTC, modern Web development, Node.js, C++ and, above all, Real-Time Communications.\nDuring the last years of my life I've been deeply involved in Voice over IP technologies, collaborating in many well known Open Source software projects and designing and implementing others of my own. I also have authored and co-authored some protocol specifications in the IETF, including the RFC 7118 “The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)” where my transition to WebRTC began.\nSince the first days of WebRTC my professional efforts go around it, and this is where mediasoup takes place. Made with love, this is by far my largest Open Source project, combining several of my preferred technical fields plus so many sleepless nights coding and reading specifications.",
"TWITTER":"https://twitter.com/ibc_tw",
"POSITION":""
},
{
"NAME":"José Luis Millán",
"SPEECHTITLE":"Building multi-party video conference applications with mediasoup",
"SPEECHABSTRACT":"\"Our hyper cool video conference app requires Chrome browser\"\nTired of reading that? Are you building your own multi-party WebRTC app but can't even figure out how to deal with the different API and feature set of each browser?\nWe feel your pain. That's why we started the mediasoup project.",
"TEXTIMG":"https://mediasoup.org/images/jmillan.jpg",
"TEXTTITLE":"José Luis Millán",
"COMPANY":"",
"DESC":"I'm a Real Time Communication passionate and developer. I have many years of experience in SIP and VoIP, and for the last years I've been focused on WebRTC development in client and server side.\nI'm a core author of JsSIP “The JavaScript SIP library” and co-author of the RFC 7118 “The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)” and have taken part in the specification of Object RTC (ORTC) API for WebRTC",
"TWITTER":"https://twitter.com/jomivi",
"POSITION":""
},
{
"NAME":"Vasilios Tzanoudakis",
"SPEECHTITLE":"How to build LRN Service using OpenSIPS",
"SPEECHABSTRACT":"I will explain to the community how here in VOICELAND we used OpenSIPS to create an LRN (Location Routing Number) Service. We used OpenSIPS 2.2.5 along with cachedb_redis and cachedb_memcached modules and we managed to get response times lower than 500 Microseconds.",
"TEXTIMG":"assets/images/speakers/Vasilios.Tzanoudakis.jpg",
"TEXTTITLE":"Vasilios Tzanoudakis",
"COMPANY":"VOICELAND P.C.",
"DESC":"",
"TWITTER":"",
"POSITION":"Technology Director & Co-Founder"
},
{
"NAME":"Oleg Khovayko",
"SPEECHTITLE":"Blockchain for VOIP: ENUMER - Distributed ENUM implemetation",
"SPEECHABSTRACT":"Introduction into a decentralized implementation of the ENUM VoIP protocol based on the Emercoin blockchain. We discuss how the ENUMER system is organized and how it is different from other ENUM implementations.",
"TEXTIMG":"assets/images/speakers/oleg_khovayko.jpg",
"TEXTTITLE":"Oleg Khovayko",
"COMPANY":"Emercoin Group",
"DESC":"Born in 1967, MS degree in CS from MEPhI. Has a strong background in the realm of finance and specializes in the technical aspects of blockchains and cryptography.",
"TWITTER":"",
"POSITION":"CTO"
},
{
"NAME":"Shlomi Gutman",
"SPEECHTITLE":"Distribute queues with multi channels/agents in multi DCs, for future contact center solutions",
"SPEECHABSTRACT":"Voicenter cloud contact center offers one of the most innovative telephony system in the telecom and call center industry, while maintaining full flexibility for the call centers' needs, along with smart telephony tools customized to the different type of end users, with full API suite supporting integration to many business applications and CRMs. How we are building a contact center cloud which allows us to manage and optimize call centers - using Opensips, RabbitMQ, NodeJS, REDIS, some logic, and a lot of love :-). We will look at how we use real-time events to manage your system, callers and agent state, while keeping your systems stateless.",
"TEXTIMG":"https://media-exp2.licdn.com/media/AAEAAQAAAAAAAAQgAAAAJGIwZjA4NTQwLTY3M2QtNDUxZC04OTNjLWI5NDY0MDM0M2E1OA.jpg",
"TEXTTITLE":"Shlomi Gutman",
"COMPANY":"Voicenter Ltd",
"DESC":"",
"TWITTER":"https://twitter.com/@VoicenterCloud",
"POSITION":"Owner"
},
{
"NAME":"Nitzan Gutman",
"SPEECHTITLE":"Distribute queues with multi channels/agents in multi DCs, for future contact center solutions",
"SPEECHABSTRACT":"Voicenter cloud contact center offers one of the most innovative telephony system in the telecom and call center industry, while maintaining full flexibility for the call centers' needs, along with smart telephony tools customized to the different type of end users, with full API suite supporting integration to many business applications and CRMs. How we are building a contact center cloud which allows us to manage and optimize call centers - using Opensips, RabbitMQ, NodeJS, REDIS, some logic, and a lot of love :-). We will look at how we use real-time events to manage your system, callers and agent state, while keeping your systems stateless.",
"TEXTIMG":"https://media.licdn.com/media/p/2/000/192/191/3a19785.jpg",
"TEXTTITLE":"Nitzan Gutman",
"COMPANY":"Voicenter Ltd",
"DESC":"",
"TWITTER":"https://twitter.com/@VoicenterCloud",
"POSITION":"CEO"
},
{
"NAME":"Perry Ismangil",
"SPEECHTITLE":"PJSIP: A decade of portability",
"SPEECHABSTRACT":"Tracing the history of PJSIP through the years with lessons learned",
"TEXTIMG":"https://media-exp2.licdn.com/media/AAEAAQAAAAAAAAN0AAAAJDBhM2NjY2Y4LWNhNzktNDUxNC1iMTRiLTNjMjNmMzI2OTdjMw.jpg",
"TEXTTITLE":"Perry Ismangil",
"COMPANY":"Teluu",
"DESC":"",
"TWITTER":"https://twitter.com/@ismangil",
"POSITION":"Co-Founder"
},
{
"NAME":"Konstantin Mikhaylov",
"SPEECHTITLE":"High-load VoIP monitoring and troubleshooting instrument SIP3.IO Tapir. ",
"SPEECHABSTRACT":"Stop blindly digging through tons of raw pcap files while working on VoIP support tickets. SIP3.IO Tapir is an open-source designed to be scalable and fault-tolerant even under thousands SIP messages per second. Tapir has clear and simple UI - resolve issues in minutes and save core team time.",
"TEXTIMG":"",
"TEXTTITLE":"Konstantin Mikhaylov",
"COMPANY":"SIP3.IO",
"DESC":"10 years in the past of hands-on operations in telecom with value add services platforms - SS7, MAP, IN/CAMEL, SMS, Voice, Payments, etc. Last 10 years of commerce and entrepreneurship in areas of messaging, VoIP/Web calling, monitoring and management solutions. SIP3.IO Co-founder and CEO. ",
"TWITTER":"https://twitter.com/@sip3_io",
"POSITION":"Co-Founder"
},
{
"NAME":"Rik Broers",
"SPEECHTITLE":"TBD",
"SPEECHABSTRACT":"TBD",
"TEXTIMG":"https://media-exp2.licdn.com/media/p/4/005/027/297/075006f.jpg",
"TEXTTITLE":"Rik Broers",
"COMPANY":"Motto Communications",
"DESC":"",
"TWITTER":"",
"POSITION":"Voice Engineer"
}
],
"schedule": [
{ "id": "first-day",
"active": "in active",
"items": [
{ "time": "8:30-9:30",
"description": "Registration & Breakfast",
"subtitle": "Venue Front-Desk",
"icon": "assets/images/lunch.png" },
{ "time": "9:30-13:00",
"description": "First Conference Day",
"subtitle": "Speakers TBA",
"icon": "assets/images/speaker.png" },
{ "time": "13:00-14:00",
"description": "Opensource Lunch",
"subtitle": "Sponsored by OpenSIPS",
"icon": "assets/images/lunch.png" },
{ "time": "14:00-17:00",
"description": "Conference Day",
"subtitle": "Speakers TBA",
"icon": "assets/images/speaker.png" },
{ "time": "17:00-18:00",
"description": "Round Tables",
"subtitle": "Featuring Speakers & OpenSIPS Team",
"icon": "assets/images/round-table.png" }
]
},
{ "id": "second-day",
"active": "",
"items": [
{ "time": "9:30-13:00",
"description": "Second Conference Day",
"subtitle": "Speakers TBA",
"icon": "assets/images/speaker.png" },
{ "time": "13:00-14:00",
"description": "Opensource Lunch",
"subtitle": "Sponsored by OpenSIPS",
"icon": "assets/images/lunch.png" },
{ "time": "14:00-17:00",
"description": "Conference Day",
"subtitle": "Speakers TBA",
"icon": "assets/images/speaker.png" },
{ "time": "17:00-18:00",
"description": "Round Tables",
"subtitle": "Featuring Speakers & OpenSIPS Team",
"icon": "assets/images/round-table.png" }
]
},
{ "id": "third-day",
"active": "",
"items": [
{ "time": "10:00-11:00",
"description": "Interactive Demos",
"subtitle": "DTMF based routing with FreeSWITCH and OpenSIPS, Liviu Chircu",
"icon": "http://i.imgur.com/LmSk7gI.jpg" },
{ "time": "11:00-12:00",
"description": "Interactive Demos",
"subtitle": "Realtime call recording using SIPREC and OREKA, Razvan Crainea",
"icon": "http://i.imgur.com/YbMlg4m.png" },
{ "time": "12:00-13:00",
"description": "Interactive Demos",
"subtitle": "CGRates for LCR, Dan Christian Bogos",
"icon": "http://i.imgur.com/2D2AjWr.png" },
{ "time": "13:00-14:00",
"description": "Opensource Lunch",
"subtitle": "Sponsored by OpenSIPS",
"icon": "assets/images/lunch.png" },
{ "time": "14:00-15:00",
"description": "Interactive Demos",
"subtitle": "Some clustering magic in real life scenarios, Vlad Patrascu",
"icon": "assets/images/speaker.png" },
{ "time": "15:00-16:00",
"description": "Interactive Demos",
"subtitle": "Running distributed presence, Bogdan Iancu",
"icon": "http://i.imgur.com/c0q8TjB.png" },
{ "time": "16:00-17:00",
"description": "Interactive Demos",
"subtitle": "Workshops and Demos",
"icon": "assets/images/speaker.png" }
]
},
{ "id": "fourth-day",
"active": "",
"items": [
{ "time": "9:00-17:00",
"description": "OpenSIPS Training",
"text": "The OpenSIPs Training will cover the new clustering capabilities of OpenSIPS 2.4 . The concept of this training course is to spend the day building from the ground up a fully operational OpenSIPS cluster for typical class 4 VoIP services.",
"icon": "assets/images/training.png" },
{ "time": "13:00-14:00",
"description": "Opensource Lunch",
"subtitle": "Sponsored by OpenSIPS",
"icon": "assets/images/lunch.png" },
{ "time": "09:00-17:00",
"description": "FreeSWITCH Training",
"text": "FreeSWITCH Training will cover the installation and configuration of FreeSWITCH. We will walk through making calls, administer various configurations, enable and utilize various modules. We'll also cover some additional functions of FreeSWITCH such as video call recording, video conferencing, Call Detail Recording, troubleshooting, logging, and interacting with Event Socket Library.",
"icon": "assets/images/training.png" }
]
}
],
"sponsors": []
}