AstriCon 2018

As the use of cloud-native computing environments continues to grow, many people find themselves at a crossroad: would it be more cost effective to run you own datacenter, or continue to grow in a hosted environment? What do you give up in that trade, and is what are the options?Levering tools like Kubernetes or Docker Swarm help abstract away the underlying machine providers until they become a commodity themselves. Why not run certain services that are disk intensive at a certain provider that provides excellent I/O, while running traffic that is bandwidth costly in a different location? To reduce operating expensive and increase competitiveness, organizations need to be able to spread their service load to where & when it is needed. If your traffic is spiking in New York, that should be near the location where additional resources are automatically deployed. In this talk we'll discuss actual numbers of running certain services on a few major providers (GCP, DigitalOcean, AWS) and deployment strategies using tools like Kubernetes & Docker Swarm. We'll focus on when the logic dictates to utilize your own hardware and the business trade-offs that come with that type of decision.

Over the last few years machine learning has exploded into the world of computing thanks to remarkably enhanced algorithms, tremendously increased data, and significantly more-powerful computer hardware. Google, Facebook and Amazon all have platforms that allow developers to utilize voice and text data to perform specific actions based on the needs of a developer. The sky’s the limit. However, one has to ask, how can I integrate these services into open source FreePBX utilizing Asterisk? In 2014 Asterisk released ARI, a powerful asynchronous API that allows developers to build their own custom communications applications. This API opened the floodgates for Asterisk to be used with machine learning tools. In this session learn how to utilize these machine learning, text to speech and speech to text with the powerful dialplan of FreePBX and Asterisk's ARI to create FreePBX modules that can learn from your users to do anything from turning lights on an off, adjusting your thermostat, accepting pizza orders or getting the current housing prices in any neighborhood.

I will go through how to set up Asterisk dialplan for automatically accepting agents and calls in an adhoc small callcenter, and creating a PHP backend server communicating with the Asteriskover AMI and a live Javascript driven webpage over a WebSocket. All of the pieces will be simple and to-the-point to be able to easily show how each piece of the architecture fits together.The talk will go through all of the Asterisk, PHP and Javscript code, step by step.

Today the challenge of communication platforms is to make management activities easier and more efficient. Given this scenario, the recently released ISO considers new modules for IssabelPBX such as queue continue, survey, write queue log and dynamic routes created to be used as a basis to capture data and interact with the different market-leading CRMs. These modules make the installation, configuration, and maintenance of the Issabel platform easier.

During this session, the participants will know the functionalities and scope of the new modules released in the recent ISO update which can be accessed from the administration panel.

Find out how FreePBX® grew to be more than just an easy way to write config files for Asterisk®, to becoming the most widely adopted Open Source PBX Platform and EcoSystem in the world. In this talk, we will provide an overview of what the FreePBX® project is and the history of how FreePBX® has evolved into what it is today, including how our millions of users continue to benefit as the project evolves. Come learn about how the project has evolved and what's coming next.

Following Sangoma and Digium getting together, we knew you would have a lot of questions about Asterisk, FreePBX, and the future. We've assembled a great collection of very senior people so that you can Ask Them Anything! This panel is the place to get those answers! MC Duffett will preside...

Learn everything about the latest OpenSIPS release and the types of services which can be built on top of it. Distributed call center queues, SIP registrars and presence servers, each acting as a single entity from an interaction perspective. Achieve platform-wide call rate / concurrent calls limiting for your SIP customers and providers using nothing more than a handful of configuration options.Learn about the optional "High Availability" extensions of the major OpenSIPS modules, and how you can blend them together into carrier-grade services with "5 nines" uptime guarantees. Here we will discuss solutions such as Anycast (for PoP redundancy), Virtual IPs (for instance redundancy) and distributed databases -- building blocks which help us meet our SLAs.

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many applications can be developed using a wide range of free tools.FreeSWITCH was originally designed and implemented by Anthony Minessale II with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis and we support interop with Asterisk over SIP.FreeSWITCH can perform full video transcoding and MCU functionality using its conferencing module. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols. FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz in mono or stereo and can bridge channels of different rates. The G.729 codec is also available under a commercial license. FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms. FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.

With exciting new technologies just around the corner, and Virtual Reality becoming more mainstream with devices such as the Google Daydream and the HTC Vive being readily available; I'm going to show you how Asterisk and more specifically it's SFU fits into these new technologies and how the Web joins them all together.

Asterisk WebRTC technology open huge scenarios of applications for unified communications.In this session we will look at that technology to realize a SIP Phone WebRTC directly integrated into your browser to provide a real-time audio & video communication WebApp. We will consider two different solutions, sipML5 and Janus Gateway, showing pros and cons of both solutions.

FreePBX was introduced in 2004, a full year before Apple launched their mobile App Store and 4 years before the first Android devices were launched. Mobile technology has since become entangled in our lives and we’ve been busy keeping pace. FreePBX recognized this trend early and has spent the last few years designing and re-architecting itself for the “mobile first” world of today. This talk will take you through the journey of transforming FreePBX into a mobile-accessible application to ensure users and administrators can access their systems where and when they needed it. This endeavor has led to our flagship product, Zulu UC, with clients spanning Windows, Mac, Linux, iOS and Android devices delivering a new and mobile-enabled experience to users and administrators alike.

Machine learning is an exploding field - with the arrival of cloud GPU instances and libraries like TensorFlow and PyTorch, models can be trained in ways that were never practical before. For us working in RTC, we can now leverage those tools to both solve issues with realtime communications and create new ones. Who’s interested in crime?In this talk we’ll discuss the future of ML with RTC. We’ve recently seen rapid advances in ASR, TTS, and machine translation. Incorporating these directly into our platforms is now a simple API call away. Next up is optimization of audio and video using models like Google’s RAISR, allowing us to realtime upscale the image stream on the client side with no artifacting. Similarly we can do advanced noise suppression using tools like Mozilla’s RRNoise or, even more exciting, only send a subset of audio and recreate speech on the far side using a model of the speakers voice.Machine learning works best when there is voluminous amounts of well-structured data – something that we generate each call in spades! We can also use machine learning to analyze data and watch metrics during a session to re-route or optimize bitstreams accordingly. After the call, we can take a look at the statistics to improve future performance.But what if you weren’t the ethical, upstanding citizen you are today? These tools can also be used for nefarious purposes. Imagine receiving a call from someone you loved saying there were in the hospital, so you quickly leave your house unlocked. And are we OK putting voice over actors out of work? We’ll talk through some of the societal repercussions of these advances.

An update on the chan_pjsip channel driver's throughput and performance. Progress made over the past year and remaining issues to be resolved.

2003 we stared developing an asterisk infrastructure for an carrier grade setup. 2008 it became much more complex by adding business PBX feature to this environment. Since than we basically stuck with this setup and there were no big movements technology wise since there was not reason to. In Asterisk 12 Digium introduced ARI with a bunch of new possibilities and changed the handling of channels so we had the need to change our Code in order to keep the asterisk up to date. With ARI as a good balance between Call Setup manipulation and in call control we chose ARI as basis of our new way for communication between Asterisk and our business logic. With message driven technologies we had the opportunity to decouple the asterisk from it's script by using a message bus, in our case we've choose kafka to do this job. With stream based programming and state machines using java akka we first made the most complex playback of the tt-monkeys. On this basis we are developing our new call controlling back end for the asterisk (ongoing). I will show the basic concept of an distributed akka, kafka message driven call routing engine. As well I will share our leanings with an HA and fail over concept and how this works all together using kamailio dispatcher routing and Stasis fallback.