For two years our telehealth product tested video calls the same way: before a release, two people on the QA team joined a call from different rooms and said "I can see you". Regressions in codec negotiation, reconnection and screen sharing reached customers anyway, because the manual check covered one network, two browsers and about ninety seconds of call time. We now run more than forty automated call scenarios on every merge. This post covers the parts that made it work: fake media...
Automating WebRTC tests with fake media devices, getStats() and network shaping
Our video-call regressions were found by customers because every test needed two humans with webcams. We automated it: fake camera and microphone feeds in Chromium and Firefox, two-party calls in one Playwright test, assertions on getStats() instead of screenshots, and tc netem for packet loss and latency.
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I am a Senior Software Engineer with over a decade of hands-on experience building, scaling, and maintaining complex systems across the full stack—spanning frontend, backend, and DevOps. Over the last 10 years, I have navigated diverse architectures and untangled some of the most frustrating, elusive bugs in production. I believe that every production incident is a massive learning opportunity. Having discovered hibug.me, I am excited to contribute to this knowledge base, breaking down root causes, and sharing verified fixes to help fellow developers debug faster and build more resilient systems.
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