2026-11-02 –, Room 2 Language: English
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)
Agenda
Which products use WebRTC and how are they different
Call setup in a nutshell
(explained with a capture - we use one very typical capture throughout the course)
- What happens before WebRTC kicks in (Signaling, SDP)
- Connectivity detection ICE/STUN/TURN
- What can I optimize on client side (restrict candidates)
Analyzing the capture
- What AI can tell us about the capture
- Go through the setup packet by packet and check the media flows
Media Performance
- UDP vs TCP
- Packet loss & delay in a worldwide network
Doing a live capture
- Decrypt signaling
- Check a few selected captures
The nasty stuff (and how to deal with it)
- Packet filters
- Proxies & Zscaler
- VPN & SDWAN
- Reconnect times
- TURN servers
What's the difference in how the various products use WebRTC
Teams
Zoom
Webex
GotoMeeting
Google Meet
Helping customers and engineers to understand problems with communication protocols in their networks.
Debugging communication protocols and analyzing log files for a living and having fun with it.
Finding problems in software since the era of punched tapes.
Currently Freelance