Why server logs are useless for WebRTC #webrtc #getStats #observability

In this technical talk, the speaker argues that server‑side logs cannot reliably debug WebRTC calls because the media streams are encrypted with SRTP and often bypass the server entirely, traveling peer‑to‑peer or through TURN relays that merely forward unreadable packets. Traditional deep‑packet‑inspection techniques from the SIP era are therefore ineffective. The speaker outlines three observation points: the server (signaling and metadata), media servers (streams and routes that pass through them), and the client (the only entity that sees the actual media, network conditions, and CPU usage). The client’s getStats API provides the true telemetry needed to understand call performance. Relying solely on server logs results in observing a shadow of the call rather than the call itself, so developers should collect and analyze client‑side statistics for accurate debugging.










