2022 ClueCon Training

Check back soon for info on lunch & learns!

FreeSWITCH Training: Friday, October 21st
Time: 9:00 am – 1:00 pm CT
Price: $299.99

FreeSWITCH 101

FreeSWITCH Training is aimed at individuals with limited experience in telecommunications.
Experience in SIP/WebRTC is preferred, but not required.
● Attendees are assumed to be familiar with a base knowledge of Linux operating systems
at command line level, administering commands, editing files, troubleshooting logs.
● Attendees should have softphones (Linphone, Voip by AntiSip, Zoiper, Bria, or X-Lite) installed on Laptop and Smartphone.

More Info

Past Workshops at ClueCon

Understanding Kamailio SIP Routing Language
Kamailio has a specific configuration file combining global parameters, which are set in an ini-like fashion, with a scripting language used inside routing blocks, a concept similar to callback functions executed on various events. This tutorial aims to explain the structure of kamailio.cfg to the newcomers and discuss the advanced capabilities for the more experienced users.

Advanced Configuration and Application Development
We will start with a brief review of a FreeSWITCH minimal configuration, delving into best practices and tips for managing your instances. The most common application patterns to interact with a web endpoint will be applied to build a complete PBX/IVR proof of concept. The main focus will be on mod_xml_curl and related modules, and CDR extraction. WebRTC support will be introduced through the Verto protocol, and we will build a web based video conferencing application. We will also examine a few examples of how to integrate advanced SignalWire features into your FreeSWITCH applications with minimal effort.

Custom call statistics and/with Quality-based-Routing for OpenSIPS
The OpenSIPS 3.2 version is able to build custom statistics based on various SIP specifics. Global or per carrier/trunk/user, such statistics may be exported via a Prometheus connector (for further analytics or operational needs) or may be fed into OpenSIPS’s Quality-based-Routing engine in order to ensure the best call quality in relation to carriers. Everything automatically done with realtime adjustments via a feedback loop. Everything to guarantee your user with top quality!

Janus as a WebRTC enabler
It’s very easy to have Janus interact with applications that only understand plain RTP, and thus allow non-WebRTC applications to interact with WebRTC users. In this context, Janus indeed acts as a WebRTC enabler. This workshop will present a few practical examples, like how to interact with legacy endpoints, how to turn a plain RTP stream to a WebRTC broadcast, how to consume and use WebRTC media for different use cases (e.g., media processing), and how to take advantage of this feature to create a scalable application as well.

HOMER and HEP in real-life
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.

Hands on workshop to build functional, zoom-like, video meeting application
In this workshop we’ll build a fully functional zoom-like video conferencing application with the SignalWire Video API.
Together, we’ll build both a web-based frontend and a server-side, command and control, application “server”. The workshop will provide reusable components and step by step directions to build the application. Developers with all skill levels and with experience in any tech stack are welcome. At the end of this workshop, you will walk out with a fully functional video application that you can use with your team. We’ll even tell you how to host the application for free.

Thank you 2022 ClueCon sponsors!