Low-Latency Radio: Where WebRTC Fits Next to HLS
Traditional internet radio streams (Icecast/Shoutcast, and HLS on the web) buffer several seconds of audio for stability. That is fine for music rotation, where nobody notices a ten-second delay. It is a problem the moment a host wants to take live calls, run an interactive show, or broadcast a live set where timing matters.
That is where WebRTC comes in. Built for real-time voice and video in the browser, it can push glass-to-glass latency under a second — the same technology behind the U-Rock scenester Go-Live feature. The trade-off is that WebRTC is heavier to scale to thousands of simultaneous listeners than a plain HLS or Icecast stream.
The practical answer for most operators is both. Use WebRTC for the live, interactive origin — the host, the call-in, the on-scene broadcast — then transcode that into HLS or an Icecast stream for the mass audience. Real-time where it counts, cheap reach everywhere else.