SRT is the best way to get video from a bad network to a server. WebRTC is the only way to get it into a browser at sub-second delay. Converting SRT to WebRTC lets you have both, and you can test the whole chain from your terminal in a few minutes.
How the SRT to WebRTC pipeline fits together
Why not just use HLS?
You can, and for big audiences that only watch, you should. But standard HLS adds 8–12 seconds, which throws away everything SRT saved on ingest. For auctions, betting or remote operation, WebRTC is the output that keeps the advantage. More on SRT ingest and delivery options, including caller and listener setup.
Try SRT to WebRTC end to end
The complete SRT streaming guide covers encoder settings and troubleshooting.
To run this tutorial against a real server, start a 14-day free trial of Ant Media Server Enterprise. SRT ingest, WebRTC playback and adaptive bitrate are included, and no credit card is needed.
SRT to WebRTC FAQ
Can a browser play SRT?
No. A server has to convert SRT to WebRTC or HLS first.
What latency should I set in FFmpeg's SRT URL?
3–4 times your round-trip time, written in microseconds. For a 100 ms round trip, try latency=400000.
Why disable B-frames for WebRTC?
Browsers handle WebRTC video without B-frames more reliably, and dropping them also cuts a little encoding delay.
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