Alexander Mueller
After studying computer science at the University of Applied Sciences in Zwickau, I joined GoTo (formally Citrix Online/LogMeIn), working on video processing topics in a distributed system, followed by media server development where I got my fingers into the WebRTC world.
Currently I'm doing more architectural things around our RTC infrastructure for GoTo Meeting, Webinar and Training products.
Session
If you have users complaining about audio issues in their Teams/Meet/Zoom calls and need to find out the reason.
- If you need to configure your network perimeter to optimally support WebRTC traffic under specific regulations
-> this class is for you.
Goal
After this class you will be able to take a capture of a WebRTC Web Conferencing call,
analyze the salient call components for potential problems and implement the required
measures in your network perimeters to overcome such problems.
After a general introduction of WebRTC we will show the use of AI tools for a quick start, and then dive into the packets.
There will be prepared captures, and a live capture sessions - but you can also bring our
own captures or capture with your specific Web conferencing tool (as long as it uses WebRTC)