Talk to most engineers about real-time apps and you’ll hear about WebSockets, Server-Sent Events, maybe gRPC streaming. WebRTC barely makes the conversation. Yet every real-time audio or video product we’ve shipped — CallVista, telemedicine portals, remote-monitoring apps — has WebRTC sitting underneath it. Here’s why.
WebRTC solves the boring problem
The hard part of real-time isn’t moving data fast. It’s moving audio fast, with low latency, with codec negotiation, with NAT traversal, with peer-to-peer fallbacks, and across every browser ever made. WebRTC is the only library that does all of that out of the box.
It’s not actually a library
WebRTC is a set of standards baked into every modern browser. There’s no install step, no SDK to lock you in, no per-minute pricing. You write some JavaScript, the browser handles ICE, SDP, and the media codec, and you get a working audio stream. The whole thing is free.
Where it falls down
WebRTC is great at peer-to-peer. It’s miserable at one-to-many broadcast — you need a media server (Janus, Mediasoup, LiveKit) to fan out to multiple listeners. We use WebRTC for capture and a media server for distribution. That split keeps the latency low at the source.
Why we keep reaching for it
Two reasons. First, every alternative we’ve tried adds a vendor dependency we don’t want. Second, WebRTC has been stable for ten years — long enough that we’re not betting on it changing. Unsexy, mature plumbing is exactly what you want in the part of the app that can’t fail.