EXPLAINER
WebRTC live streaming explained
WebRTC is the open web standard that lets a browser send and receive live audio and video without plugins. It is the reason you can go live from a phone browser with no app.
What WebRTC is
WebRTC (Web Real-Time Communication) is a set of browser features standardised by the W3C and the IETF. It is built into current versions of Chrome, Safari, Firefox and Edge. A web page asks for permission to use the camera and microphone, gets a live media stream, and sends it over an encrypted connection to other browsers or to a server. All of this happens in the page itself, with no download.
How a broadcast travels
- 1
Capture
The page requests the camera and microphone. The browser shows a permission prompt, and capture only starts if you allow it.
- 2
Connection setup
The page and the service exchange the details needed to connect, which is called signalling. On LiveSnap.tv this runs over HTTPS and does not carry the video.
- 3
Media relay
The picture and sound go to a media server that forwards them to viewers.
- 4
Playback
Each viewer’s browser receives the stream over WebRTC and plays it. Quality depends on their own network and device.
For the exact components used by LiveSnap.tv, see the WebRTC architecture page.
What an SFU does
An SFU (Selective Forwarding Unit) is a media server that receives each broadcast once and forwards it to every viewer. It does not mix the video into a new file. This is why a phone only has to upload a single copy of the stream, no matter how many people watch, which keeps the load on the broadcaster’s connection low.
WebRTC compared with RTMP and HLS
- RTMP is an older protocol that usually needs an encoder app or software on the sending side and a server that converts the stream for viewers.
- HLS splits video into short files that players download one after another. It scales well and suits recordings and very large audiences, but it normally adds several seconds of delay or more.
- WebRTC sends media continuously and is designed for interactive use, so the delay is typically well under a few seconds on a good connection. The trade-offs are more complex infrastructure and quality that adapts to the network rather than staying fixed.
Security and privacy basics
- WebRTC media is encrypted in transit with DTLS-SRTP.
- Browsers only allow camera access on secure (HTTPS) pages and only after your permission.
- Encryption in transit does not mean a stream is private. Anyone allowed to view a public stream can watch and record it.
Limits to know about
- Quality is not fixed. Resolution and frame rate adapt to bandwidth and packet loss.
- Mobile browsers may pause capture when a page goes to the background, so keep the page open while broadcasting.
- WebRTC delivers live pictures. It does not create a recording by itself, and LiveSnap.tv deliberately does not archive broadcasts.
- Corporate networks and some firewalls can block WebRTC traffic.
Frequently asked questions
Do I need a plugin or an app for WebRTC?
No. WebRTC is built into current browsers, which is why LiveSnap.tv works without installing anything.
Does WebRTC work on iPhone?
Yes, in Safari from iOS 11 onwards. Some in-app browsers restrict camera access, so open the page in Safari or Chrome.
Why is WebRTC delay lower than HLS?
HLS delivers video as a series of files, which adds buffering. WebRTC streams media continuously and uses small buffers, so viewers see the picture sooner.
Is WebRTC encrypted?
Yes. Media travels with DTLS-SRTP encryption. Encryption protects the stream in transit, not from the people who are allowed to watch it.
Try it on LiveSnap.tv
Open the home page, watch the current broadcast and scan the QR code to broadcast from your phone. No app and no account are needed.