It takes an input video and converts it into H.264/Opus RTP streams that you can blast at your video call systems (WebRTC, SFUs, etc.). It also injects network chaos like packet loss, jitter, and bitrate throttling to see how things break
It scales from 1 to n participants, depending on the compute and memory of the host system Best part? It’s packaged with Nix, so it builds the same everywhere (Linux, macOS, ARM, x86). No dependency hell
It supports both UDP (with a relay chain for Kubernetes) and WebRTC (with containerized TURN servers). Chaos spikes can be distributed evenly, randomly, or front/back-loaded for different test scenarios. To change this, just edit the values in a single config file
Cool. I did this exact same thing at Cruise to prove and defend my approach to monitoring e2e safety/latency in the remote assistance tool.
In particular, i established that sudden changes in connection speed create latency in the video channels that do not correlate with the data channels
very cool is there something like this but for SIP?