WebRTC vs HLS: Low-Latency Live Streaming Compared

The Need for Low Latency

In interactive live streaming, online education, and remote meetings, low latency is a core requirement. Traditional HLS live streaming has a delay of 6–30 seconds, while WebRTC can achieve ultra-low latency under 500ms. Each has its strengths, and the right choice depends on your specific use case.

WebRTC: Strengths and Limitations

WebRTC is built on UDP and uses P2P or SFU architectures, achieving latency as low as 200–500ms. It's native to browsers, requires no plugins, and is ideal for real-time interactive scenarios.

However, WebRTC has notable limitations: high server costs at scale (requiring dedicated SFU servers), demanding network quality requirements (performing poorly on weak networks compared to HLS), and while browser compatibility is good, mobile support still needs work.

Low-Latency HLS

Apple introduced Low-Latency HLS (LL-HLS) in 2019, using techniques like partial segments, preload hints, and blocking playlist reloads to reduce HLS latency from the traditional 10–30 seconds down to 2–5 seconds.

LL-HLS's advantage is full compatibility with existing HLS infrastructure — no additional server architecture needed. For scenarios already running HLS, upgrading to LL-HLS is relatively inexpensive.

How to Choose

If latency must be under 1 second (video calls, interactive gaming), choose WebRTC. If 2–5 seconds is acceptable (e-commerce live, online classes), Low-Latency HLS is more cost-effective. For massive audiences where latency isn't critical (sports events), traditional HLS with a CDN works fine.