Debugging Tools and Techniques for WebRTC
summary
WebRTC problems are diagnosed with concrete traces: use Chrome WebRTC internals to capture per-second stats and API events, extract the big JSON dump, and analyze bytes/packets, jitter spikes and bandwidth ramps to spot 1% call failures, repeated drops (600k → 40k) or long jitter spikes. Check for missing relay candidates (type=relay) to detect blocked UDP/TCP and for zero audio bytes (a 2–3% sleep/mic bug). Instrument all sessions with an auto collector (GitHub joint project with Tokbox) to get reproducible dumps.
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