Why Scaling WebRTC Applications Is Mostly an Infrastructure Problem

When people talk about WebRTC, the conversation usually revolves around APIs. getUserMedia() , RTCPeerConnection , ICE candidates the browser side gets most of the attention. And to be fair, getting a basic video call working isn’t particularly hard anymore. But scaling WebRTC is rarely a browser problem. It’s an infrastructure problem. Once traffic increases or enterprise users start connecting from unpredictable network environments, the weak points begin to show. TURN usage spikes under restrictive NATs. Signaling servers struggle with session churn. Packet timing issues surface under load even though CPU graphs look normal. The challenge isn’t building the feature. The challenge is building the system around it. For example: Corporate networks that block UDP can quietly push most sessions through TURN relays. Poor signaling design can introduce state synchronization delays across instances. Overloaded media nodes can amplify jitter even when bandwidth appears sufficient. These issu