Earlier quoted context omitted.
WebRTC excels at sub-second latency peer to peer, so you can do near-realtime video, so anywhere that is useful. Say you wanted to do a virtual portal installation connecting views from two different cities with live audio, you could have ffmpeg feed off a professional cinema or DSLR camera device with a clean audio feed and stream that over WebRTC into a webpage-based live viewer. Or say you wanna do a webpage that…
The irony is that you don't _actually_ need WebRTC to get subsecond latency; you can fairly reliably get ~100–200ms (plus network latency) with a completely normal TCP stream.[1] But since browsers have effectively standardized on HLS, whose design is completely antithetical to low-latency (you _can_ do low-latency HLS, but only with heroic efforts), low-latency streaming video has never really been part of their bre…
Apple’s LL-HLS spec shrinks those segments into “partial segments” and uses CMAF to let the player start decoding while a chunk is still arriving. With careful encoder tuning, HTTP/2/3 push, CDN support, and a compatible player you can reach 2–5 s, sometimes a bit lower—but every link in the chain has to cooperate, so implementations are still called “heroic” for a reason.
Safari plays HLS natively and on other browsers you can bolt on hls.js via Media Source Extensions. DASH, RTMP, SRT, etc. all need extra code or plugins, so HLS became the “safe default” for on-demand and broadcast-style streaming even though it isn’t low-latency friendly.