Kamailio workshop: Tuesday, February 16th
Time: 10:00 am – 11:00 am CT
Kamailio – SIP Routing 101
During the previous ClueCon, the Kamailio workshop focused on getting started with it, providing directions for finding the relevant resources to understand the design, purpose and how to use it. Now it is time for the next step: the basic concepts of building SIP routing systems with Kamailio. Join the workshop to learn how to configure Kamailio to perform SIP routing, from simple scenarios like stateless or stateful forwarding to load balancing and least cost routing, combined with common operations like target address and SIP headers management.
SignalWire workshop: Tuesday, February 16th
Time: 1:00 pm – 2:00 pm CT
SignalWire Cloud 101
The goal of the workshop is to briefly introduce the SignalWire Cloud offering, including inbound and outbound calls, messaging and number provisioning.
We will start with an introduction to LAML and the REST API, registering to SignalWire and provisioning a phone number, and making our first call. We will wrap up the workshop by building a simple IVR.
OpenSIPS workshop: Tuesday, February 16th
Time: 11:30 am -1 2:30 pm CT
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 workshop: Wednesday, February 17th
Time: 1:00 pm – 2:00 pm CT
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 workshop: Wednesday, February 17th
Time: 2:30 pm – 3:30 pm CT
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.