Anthony has 20 years of experience in software engineering and 15 of them are dedicated to telecommunications. In 2005, Anthony founded FreeSWITCH, the Open-Source Software-Defined Telecom engine used by the Enterprise worldwide to power their platforms.
FreeSWITCH is a cross-platform software stack written in C and C++ that implements a fully-functional telecommunications engine. FreeSWITCH has a large active community and is used in most popular CPaaS platforms as the core telephony stack.
As an avid open-source advocate, Anthony has spent his career working to create a world where open-source software is not only an effective development model or a way to get free code but also a critical component in a successful business model. He believes that by uniting the two concepts, the developers of the world can learn to accelerate ideas and then take them to the next level.
Together with a team of the best minds in the industry, Anthony founded SignalWire in 2018. SignalWire has a primary mission to bring the complex technology behind real-time communication to the mainstream. Going 2 steps beyond where FreeSWITCH leaves off, SignalWire is solving the recurring problems faced by companies today who are trying to digitally transform their legacy hardware solutions. Likewise, new companies are sprouting up every day and it’s critical that these startups can solve their communications needs quickly and economically.
What does the Cloud Mean for the Future of Telecom Technologies
We’re in the midst of a significant migration of traditional IP PBX, contact center and meetings technologies to SaaS delivery models. This transition will have profound implications on the telco technology landscape over the coming years. It will change SIP trunking; the role of the SBC; the role of standards and peering arrangements; the usage of codecs, and so on. In this talk, I’ll talk about the implications and why. Bio
Chad Hart - cwh.consulting
Kill Your IVR with a Voicebot
In this talk Chad will provide a high-level comparison of some of the commercial and open source bot platform options and how they can be leveraged as IVR platforms. He will compare typical text-bot needs and assumptions built into these platforms with phone bot needs. Finally, he will walkthrough technical approaches to implementing a voicebot for IVR replacement. Bio
David Duffett - TeleSpeak Ltd
Why Nerds Aren’t Heard: 7 Ways Geeks Sabotage Their Own Presentations, And What To Do About It!!!
Sadly, all too often presentations made by well-meaning Geeks are a catalogue of errors in terms of effectively conveying the message they wish to get out there… Of course, there are a few things that when done properly pretty much guarantee the success of a presentation! David has boiled these things down to 7 Power Presenting Protocols (especially for Nerds) and in this session he will identify the mistakes that are often made and then offer one of the Protocols to address it. Your presentations may never be the same again!!!! Bio
James Cadd - Microsoft
I’m a Program Manager in the Developer Ecosystem & Platform team at Microsoft where I’ve managed our native WebRTC stack development for the past 5 years. My work is focused on Windows 10 apps & Azure deployments of WebRTC & ORTC.
Jim OBrien - Counterpath
10 Essential network operations tools/systems/things in 30 minutes!
Jim will explore 10 handy network operations tools/systems things in 30 minutes!
Many UC-VoIP-Next-Gen-Comms related companies-and-organizations have moved from a software-as-a-product to software-as-a-service model. Product companies main concerns were maintaining a reliable web presence and download server to ensure the flow of new purchases and support renewals. With this change, companies have shifted to take on a much more services-oriented persona. Customers can CNRTL-ALT-DELETE your service on a monthly basis providing a reason to invest in uptime, availability, resiliency, geo-location and other fancy words.
Jim will provide some lessons learned and review 10 tools that make a difference. Bio
Fred Posner - LOD
Three Ways Kamailio can Help Your FreeSWITCH Deployment
Kamailio, the open source SIP server, is often used in telecom deployments due to its small footprint and ability to handle massive call volume. This talk will look at Kamailio 5.2 and three quick ways that Kamailio can help your FreeSWITCH deployment today. Bio
James Tagg - Cen Inc
Communicating with AI Avatars
Demonstration of hyperrealistic avatars and discussion of the communications challenges to rendering realistic artificial humans. Bio
Daniel Christian Bogos - ITsysCOM GmbH
Least Cost Routing with QoS and Cost sorting using CGRateS
Dynamic routing is a must-have for companies of all sizes. The classical way of routing based on suppliers prioritization is no longer enough and the availability of alternative solutions within open-source ecosystem is rather limited.
In this talk Dan will present CGRateS SupplierS module, enhancing your FreeSWITCH based installation with advanced vendor sorting capabilities and adaptive filters.
CGRateS is a battle-tested open-source Enterprise Billing Suite with support for
various prepaid and postpaid billing modes. Bio
Alan D Percy - TelcoBridges
Battling Robocallers with STIR/SHAKEN – A Tutorial
Robocalls and falsifying Caller-ID are not just annoying…they have become the tools of scammers.
Which is why the FCC and CRTC have stepped up the pressure and demanded implementation of STIR/SHAKEN by all service providers as a way to put an end to illegal robocalling. During this tutorial, we’ll explain how it works and how service providers and enterprises can quickly implement it in their networks. Bio
Liviu Chircu - OpenSIPS Solutions
Developing SIP Platforms as a Pastime With OpenSIPS 3.0
The major bump version bump of OpenSIPS 3.0 goes well beyond looking cool — it marks a shift in mindset. As developers of the project, we now aim to better serve both script writers and maintainers of the resulting platforms.
Join this talk in order to find out more about the newly available ways of templating your opensips.cfg file (enabling mass deployment to cloud) or how you can benefit from the enhanced troubleshooting support. Lastly, meet the “OpenSIPS Doctor” — a self-diagnosis module available in our new “opensips-cli” command line tool. Bio
Daniel-Constantin Mierla - Kamailio
Versatile SIP Routing With Kamailio Embedded Scripting Languages
Fikri Firat - Voxbone
How Open-Source Projects Catalyzed Cloud Comms
Open-source VoIP projects, like FreeSwitch, catalyzed the emergence of a new industry: CPaaS. In this talk, we’ll take a look into the rise of CPaaS, and what VoIP applications need to succeed in an enterprise market. We’ll:
Dive into the biggest challenges applications face, and how the CaaS layer can work to help eliminate those
Touch on the future of collaboration, the role of the Cloud, compliance-as-a-service, and just how smart integrations can revolutionise applications’ value-add in the enterprise market.
We’ll also share current use-cases that outline some of the challenges and demonstrate how they were conquered. Bio
Evan McGee - SignalWire
SignalWire & Global Microservices
Evan has over a decade of deep experience with the telecommunications industry, including both executive business and technical expertise with traditional wireless MNOs/MVNOs and WebRTC/VoIP OSS. He has published patents covering both hardware and software design and is a frequent presenter at realtime/IoT conferences covering aspects such as scalable SaaS deployments, best business practices, and utilization of machine learning models in a realtime setting.
As the CTO & one of the founders of SignalWire, Evan wants to help bring the power of communication to people who aren’t familiar with the industry through the power of simple-to-use APIs.
Oleg Agafonov - SIP3
VoIP Troubleshooting and Monitoring with SIP3
An exciting journey from troubleshooting to monitoring or how VoIP network monitoring can simplify troubleshooting and shorten customer tickets resolution time.
Find out how SIP3 turned from troubleshooting into monitoring platform, what were the biggest challenges along the way and what is the team behind the project planning ahead.
SIP3 is an open sourced project that helps engineering teams detect voice quality problems and prevent call failures. Take advantage of a fast and efficient platform that scales with your business. The powerful monitoring UI provides key performance metrics that are giving detailed VoIP network insights. Bio
Razvan Crainea - OpenSIPS Solutions
OpenSIPS 3.0 as entry point to private networks
Find out how you can use the new OpenSIPS 3.0 in front of your FreeSWITCH servers behind private networks! Bio
Alessandro Polidori - Nethesis srl
WebRTC Cloud Phone with Asterisk, sipML5 & Janus
The time to use your browser to make phone calls has come!
From this talk you can learn how to implement a SIP Phone WebRTC to be integrated into your Web App to make audio/video phone calls to any devices. We will see great code example, WebRTC technologies and an open source demo available on GitHub derived from a real project on production (NethCTI – www.nethvoice.it).
Two different implementations will be shown using Janus-Gateway and sipML5 libraries. A complete system will be given to the attendees to start playing with WebRTC calls in an easy manner using Asterisk & FreePBX. Bio
Diego Gosmar - XCALLY
OmniBot: Omni Channel Artificial and Human intelligence for Contact Center
John is getting in touch with a customer care in order to have some tourist information, while he is spending the holidays with his wife… And a Self Service Asterisk or Freeswitch IVR integrated with a Natural language machine engine is going to provide him some initial useful answers to his questions, also by using some AI (Artificial Intelligence) predictions… Until something unexpected comes up! …And here the things start to get really interesting, because we are trying to see what we can do when the Asterisk or Freeswitch contact center Robot can’t find reasonable answers for the customer. Bio
Lessons learned from working with VoIP every day for the last 10 years
A presentation which goes over the learning curve from being a regular (non VoIP!) developer to a VoIP developer and all the pitfalls and lessons learned along the way. Bio
James Body - Telet Research
James has been intimately involved with real time communications, in particular with Open Source VoIP and Real Time Communications for many years. He was responsible for building the initial mobile network core for the Global Mobile Network Operator, Truphone. After leaving Truphone two years ago, he set up a new startup, Telet Research, with the bold aim of providing all of the difficult to get components required to build small private mobile networks. He will invariably be found carrying large numbers of phones, SIM cards and pieces of Radio Access Network equipment about his person
James is well known for his ‘Dangerous Demos’ that have become a key fun component within the Cluecon, Kamailio World, Astricon and TAD Summit events.
Andy Abramson - Comunicano
Andy Abramson is currently CEO of Comunicano, Inc., a global value creation strategy agency, working with startups and companies in transition, with 47 exits to his credit in the last 18 years. He is also co-founder of Brand Communication Design, Inc., a full-service marketing communications agency and Velocity RPA, Inc., the enterprise focused Robotic Process Automation solutions company that services Dell and other clients from Austin, TX. In addition Andy serves on the advisory boards of numerous early stage and privately held companies. In the telecom industry Andy is best known for his top-rated VoIPWatch blog.
Brian West - FreeSWITCH/SignalWire
One of the founders of the FreeSWITCH open source project, and an advanced communications pioneer. A key player in helping SignalWire to build a cutting edge platform that will disrupt the telecommunications status quo. In his role as director of support, Brian enjoys interacting with and helping customers to solve some of the unique challenges they encounter
Michael Jerris - FreeSWITCH/SignalWire
One of the founders of the FreeSWITCH open source project, and an advanced communications pioneer. Michael is deeply embedded in the developer community and committed to their success using SignalWire.
Chris Rienzo - SignalWire
Chris is Director of Stack Engineering at SignalWire, the company started by the FreeSWITCH creators, aiming to revolutionize the API-based RTC communication services. A graduate of State University of New York at Buffalo, Chris is experienced in designing and implementing cloud communication systems and as a contributor to the FreeSWITCH project, he is the primary author of mod_unimrcp and mod_http_cache. Previously, Chris has been serving on principal or senior engineering positions at major VoIP companies, among them Citrix, Grasshopper and Genesys.
Dr Moshe Yudkowsky - Disaggregate Corporation
Zombie Apocalypse 2031: The Recovery
The year is 2031, and you’ve survived the Zombie Apocalypse. Like all other survivors, you can simply move in and commandeer a great deal of abandoned business infrastructure, including the expert systems that provided business support. Now you need some way to earn your daily bread. Software engineering just won’t cut it because the economy won’t support it — you need to develop a business that provides physical goods or services.
In this talk we’ll discuss how to use that expertise you’ve inherited, those expert systems, to develop new, unexpected, innovative, and possibly revolutionary business ideas that just might keep you in beer and skittles as the world recovers. Bio
Alberto Gonzalez Trastoy - WebRTC.Ventures
Lessons learned building an AI powered live streaming camera
Real-time communications have been evolving and going mainstream with the improvement and appearance of new open source technologies that make the development more affordable.
In this talk I will go through the design and development process of a live streaming application running on a Raspberry Pi and powered with image detection. I will be talking about some open source media servers and frameworks to achieve that, the pros and cons of some of this potential solutions, what I learned building it and what are some of the potential use cases of AI on WebRTC applications. Bio
Lorenzo Miniero - Meetecho
All roads lead to WebRTC: an intro on Janus
WebRTC stormed the telecommunications world and opened the doors to many different application scenarios that were quite hard, if not impossible, to implement before. This is one of the main reasons why we started working on Janus many years ago: to have something powerful and yet flexible enough to cover considerably different scenarios via WebRTC, from SIP gatewaying to conferencing, broadcasting, IoT, telehealth, gaming and so on. This presentation aims to give an introduction to what Janus is, its main features and how it can be used in different ways, and possibly complement Freeswitch in some interesting use cases. Bio
Giovanni Maruzzelli - OpenTelecom.IT
FusionPBX and FreeSWITCH As ToolKit For High Traffic Scalable SIP Services
FusionPBX is the web interface for FreeSWITCH configuration and management. It strives to be faithful to FreeSWITCH power and flexibility, adding features on top of it, and hiding nothing. So, you can use FreeSWITCH, a SIP Proxy and FusionPBX to build any kind of high traffic SIP B2BUA services, complete of users and roles management, dialplan, php and lua scripting, db transactions audit, CDR visualization, call recording, etc. We’ll also see how to build a new, custom FusionPBX module to deal with the specifics of a new, custom call bridging service. Bio
Nivaldo Montenegro Junior - Citrus
A harmonious 3D experience, for customers 4.0, at 5G speed: A visual context of an intelligent CX Journey
The customer experience is the buzz of the moment and everyone is looking for solutions to understand their customers’ journey. Join this talk in order to meet an innovative way to follow the customer journey using a visual context. A solution that not only provides a unique view of the customer journey, but also connects all customer service and back office to a single communication platform. That monitor in real-time the customer journey to create behavior partners and execute actions. All of these ingredients provide informations that create decision-making insights to increase customer satisfaction and boost sales. Bio
Clint Berry - Weave
The Quest for a Cloud Native Communication Platform
The rise of easy-to-use containers, Kubernetes and other cloud native technology has caused a dev-ops revolution. Software startups to fortune 500 companies are leveraging this new way of working to make it easier than ever for product engineers to focus on the business, and not the infrastructure behind the business. I will talk about strategies we have used to move our communications platform (freeswitch, kamailio, & supporting micro-services) into kubernetes and what benefits that provides and will provide us in the future. Bio
Julien Chavanton - Flowroute
Flowroute WebRTC to VoIP platform
This presentation will provide some insights on how the platform was built using RTP-Engine and Kamailio with modules like WebSocket and Topos.
We will share our JSSIP client implementation publicly and demo it to demonstrate how the existing Flowroute inbound API and SIP interconnections can now be used to receive calls from web browsers to Freeswitch and/or any SIP endpoint. Bio
SC Lee - Poly
Creating Amazing Desktop Phone Experiences with Poly APIs
Poly’s OBi Edition IP Phone is the only family of SIP endpoints that are fully programmable and manageable from the cloud. In this presentation, Poly will demonstrate and walkthrough all the available REST APIs and integration points of their solution. Poly will demonstrate how easily it is to create an amazing and engaging IP Phone experience. APIs for creating on demand announcements, Customer’s profiles popup and etc. Be ready to walk away with knowledge on how to build the next IP Phone App! Bio