WebRTC
WebRTC lets browsers and apps exchange video and audio in real time with sub-second latency, without routing through a conventional streaming pipeline.
Why use it in live commerce?
A conversation only works if the host can answer the moment a viewer asks. Past roughly ten seconds of delay, questions and answers cross, and formats where order matters — auctions, first-come coupons — stop working. WebRTC keeps that gap under a second.
Is WebRTC always the answer?
No. At very large concurrent audiences, cost and infrastructure complexity climb and stable quality gets harder. The usual split is WebRTC where real time is decisive, low-latency HLS where large-scale viewing is the goal.
Latency and use by delivery method
Method | Typical latency | Where it's used |
|---|---|---|
RTMP | 1–5s | Camera or encoder up to the server |
HLS | 10–30s | Large audiences, stable quality |
Low-latency HLS | 2–5s | General live where timing matters |
WebRTC | Under 1s | Q&A, auctions, first-come drops |
Frequently Asked Questions
Q. Does it work in a browser without installs?
Major browsers support it natively, so no plugin is needed. In corporate networks, check firewall and port policy.
Q. How is it different from RTMP?
RTMP is mostly used for the camera-to-server leg; WebRTC handles delivery down to viewers at low latency too. They're different legs of the path rather than substitutes.
Q. How does cost scale with audience?
Server resources rise close to linearly with viewers, so at large scale the unit cost exceeds low-latency HLS. That's why many teams reserve it for the broadcasts where real time decides revenue and use another path the rest of the time.
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