Balázs Kreith of the open-source WebRTC monitoring project, ObserveRTC shows how to calculate WebRTC latency – aka Round Trip Time (RTT) – in p2p scenarios and end-to-end across one or more with SFUs. WebRTC’s getStats provides relatively easy access to RTT values, bu using those values in a real-world environment for accurate results is more difficult. He provides a step-by-step guide using some simple Docke examples that compute end-to-end RTT with a single SFU and in cascaded SFU environments.
Search Results for: SFU
The Ultimate Guide to Jitsi Meet and JaaS
A full review and guide to all of the Jitsi Meet-related projects, services, and development options including self-install, using meet.jit.si, 8×8.vc, Jitsi as a Service (JaaS), the External iFrame API, lib-jitsi-meet, and the Jitsi React libraries among others.
Meet vs. Duo – 2 faces of Google’s WebRTC
A very detailed look at the WebRTC implementations of Google Meet and Google Duo and how they compare using webrtc-internals and some reverse engineering.
Optimizing WebRTC Power Consumption (Markus Handell)
The performance of WebRTC in Chrome as well as other RTC applications needed to be improved a lot during the pandemic when more people with a more diverse set of machines and network connections started to rely on video conferencing. Markus Handell is a team lead at Google who cares a lot about performance of […]
How does WebRTC End-to-End Encryption work? Matrix.org example (Dave Baker)
One of WebRTC’s great features is its mandated strong encryption. Encryption mechanisms are built-in, meaning developers don’t (often) need to deal with the details. However, these easy, built-in encryption mechanisms assume you have: 1) media is communicated peer-to-peer and 2) a secure signaling channel setup. Most group-calling services make use of a media server device, […]