Keynote

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.

Outsourcing to Kamailio

Discussion of services that can be outsourced from FreeSWITCH to Kamailio. Bio

Mastering DID Routing With Kamailio

DID in/out routing is one of the core services in VoIP and telephony. Kamailio offers many variants to implement it and this presentation tries to highlight the benefits and drawbacks for each of them, in order to enable administrators to select the right one for their specific needs, from simple prefix/DID routing solutions to highly scalable DID mapping with caching and distributed servers. No matter you are running a small PBX system or a large telephony operator, this talk can give the right answers for building the DID routing engine with a secure and flexible Kamailio system. Bio

Bringing WebRTC to broadcasters with NDI

NDI is a technology used extensively in the broadcasting industry, which allows media producers to easily incorporate local network sources in their live production setups. In an era where remote interactions are so pervasive, a way to allow producers to also integrate remote streams just as easily in their setup has become of paramount importance. Weirdly enough, though, this is usually done via hacks, and not much has been attempted to try and bridge WebRTC and NDI instead: this talk will address how we tackled exactly that using Janus, and our collaboration with Nimble Ape to make it easily accessible. Bio

AI and Machine Learning: Sentiment Analysis and Speech Applications

AI, Machine learning and related sciences will change the way we are working in many industries: for sure the communication world will be heavily impacted by such upcoming technologies.

During the session, we are going to see how to use Deep Learning based on Convolutional Neural Network to create a voice recognition engine and Recurrent Neural Network application for sentiment analysis.

Design and implementation challenges will also be explained, along with practical examples: we will see how to evaluate the performance of the AI model and how to use some of the main machine learning technique to improve its accuracy. Bio

Quality-Centric Routing using CGRateS

Practical workshop designed to serve as a step-by-step tutorial towards configuring CGRateS as a routing engine for FreeSWITCH, focused especially on quality oriented routing strategies.
CGRateS is a battle-tested Enterprise Billing Suite with support for various prepaid and postpaid billing modes. Bio

Controlling media of ongoing calls with just a SIP Proxy

Learn how to provide enhanced media capabilities (such as music on hold playbacks, PIN prompts, barge-in services, etc) to calls that are passing through OpenSIPS SIP Proxy. Răzvan Crainea is a VoIP developer with more than 10 years of experience working as an OpenSIPS Core Developer.

Controlling media of ongoing calls with just a SIP Proxy

Founder, board member and core-developer of OpenSIPS SIP Server project. Learn how to provide enhanced media capabilities (such as music on hold playbacks, PIN prompts, barge-in services, etc) to calls that are passing through OpenSIPS SIP Proxy.

Azure Communication Services

Mary Anne Noskowski is a program manager at Microsoft working on the new Azure Communication Services. Enabling developers to achieve more and empowering individuals to do their best work is at the heart of everything she does. Whether it’s providing real-time virtual assistance or enabling curb side pickup, businesses across the spectrum are needing to connect with customers. However, building new communication experiences across consumer or enterprise applications can be complex, often requiring considerable investment and specialized expertise. Azure Communication Services makes it easy to add voice and video calling, chat, and SMS text message capabilities to mobile apps, desktop applications, and websites with just a few lines of code, bringing together the best of communication technology, cloud scale, enterprise-grade security, and development efficiency.

Azure Communication Services

PM on Microsoft’s WebRTC native & cloud SDKs, frequent speaker at real-time communications conferences including IIT-RTC, Kranky Geek, allThingsRTC & ClueCon. He works internally to coordinate Microsoft’s RTC strategy across Azure, Xbox, Teams & AI.  Whether it’s providing real-time virtual assistance or enabling curb side pickup, businesses across the spectrum are needing to connect with customers. However, building new communication experiences across consumer or enterprise applications can be complex, often requiring considerable investment and specialized expertise. Azure Communication Services makes it easy to add voice and video calling, chat, and SMS text message capabilities to mobile apps, desktop applications, and websites with just a few lines of code, bringing together the best of communication technology, cloud scale, enterprise-grade security, and development efficiency.

Conquering WebRTC Statistics for Troubleshooting & Debugging at Scale

The pandemic has driven WebRTC-based services to new heights. The increased scale of traffic has also increased the scale of user and call quality issues. While the WebRTC stack includes some built-in statistics – i.e. RTCPeerConnection.getStats – these API’s where not designed for the systematic analysis and monitoring needed in production systems today.

In this talk Chad and Balazs will review common needs and challenges of WebRTC statistics. They will also introduce ObserveRTC – a new open source project for flexibly collecting and analyzing RTC at scale by leveraging other widely-used open source big data projects like Kafka.

Balazs Kreith is an experienced Software Engineer, focusing on Real-Time Communication Technologies. He is passionate about code quality and seeking for novel solutions.

Conquering WebRTC Statistics for Troubleshooting & Debugging at Scale

The pandemic has driven WebRTC-based services to new heights. The increased scale of traffic has also increased the scale of user and call quality issues. While the WebRTC stack includes some built-in statistics – i.e. RTCPeerConnection.getStats – these API’s where not designed for the systematic analysis and monitoring needed in production systems today.

In this talk Chad and Balazs will review common needs and challenges of WebRTC statistics. They will also introduce ObserveRTC – a new open source project for flexibly collecting and analyzing RTC at scale by leveraging other widely-used open source big data projects like Kafka.

Chad is an analyst and consultant at cwh.consulting, a product management, marketing, and strategy advisory specializing in WebRTC and AI in RTC. Chad is a frequent speaker, blogger/editor at webrtcHacks.com and cogint.ai, and event organizer with Kranky Geek and WebRTC Boston. See more at cwh.consulting.

SignalWire Cloud 101 and FreeSWITCH Advanced Configuration and Application Development

Luca Pradovera will be leading both the SignalWire and FreeSWITCH workshops! To learn more about workshops, click here.

HOMER and HEP in real-life

Lorenzo Mangani is  a Senior Voice Engineer, OSS Developer, Real-Time Communications & Telecom Specialist, Sound Engineer, Full-Stack Solutions Architect, plus Dreamweaver. Expert in Open-Source Telephony (VoIP, SIP, RTP, WebRTC/WSS) Packet Collection and Data Analysis/Visualization. He is the Co-Founder and Lead of HOMER SIPCAPTURE and HEPIC/PCAPTURE Projects.

He will be instructing the HOMER Workshop. HEP is HOMER’s open-source lego set designed to build open and distributed monitoring, troubleshooting and data mining integrations for modern cloud Voice, VoIP, Mobile and RTC architectures. In this workshop, we will introduce some of our flagship use-case integrations for SIP and other protocols and data types anyone can assemble in real-life using the QXIP HEP stack, to inspire the next generation of users and engineers.