WebRTC
WebRTC (Web Real-Time Communication) is an open standard, built into modern browsers, for sending audio, video and data between devices with delay usually under one second. It powers video calls in the browser without plugins. WebRTC uses UDP-based transport, built-in encryption and techniques to connect devices across home routers and firewalls.
Why it matters
When viewers talk back, through questions, bids or bets, even a few seconds of delay makes a product feel broken. WebRTC delivers delay under one second in any modern browser without plugins. The tradeoff is cost and complexity: scaling WebRTC to large audiences needs specialized media servers rather than standard CDNs.
How it works
- Two endpoints exchange connection details through a signaling server, using a method the app chooses.
- Each side discovers network routes through home routers and firewalls using ICE, STUN and TURN servers.
- Audio and video are encrypted and sent as RTP packets over UDP for minimal delay.
- Each receiver adjusts quality to its network, and a media server can forward streams to many viewers.
WebRTC vs HLS: which is better for live streaming?
Use WebRTC for real-time interaction and HLS for large audiences watching passively. WebRTC delivers delay under one second but needs media servers that cost more to scale. HLS, and LL-HLS at about 2 to 5 seconds, uses standard CDNs and handles millions of viewers at lower cost. Many products combine both: WebRTC for hosts and guests, HLS for the audience. For the audience side, FastPix delivers live streams as LL-HLS at $0.001 per viewer-minute, so the cost of a large passive crowd is easy to predict.
Is WebRTC secure?
Yes, WebRTC encrypts all media by default. Audio and video use SRTP, and data channels use DTLS, so streams cannot be read in transit. Browsers also require user permission before a site can use the camera or microphone. Security still depends on the app itself, such as how signaling and user access are handled.
Can WebRTC scale to thousands of viewers?
Yes, but not peer to peer. Large WebRTC audiences need media servers, usually selective forwarding units (SFUs) that receive one stream and forward copies to many viewers. Cost and complexity rise with audience size, which is why very large broadcasts usually move most viewers to HLS and keep WebRTC for interactive participants.
Where it's used
Live shopping apps, betting apps and video call products use WebRTC when conversation matters and even a few seconds of delay feels broken, such as hosts answering viewer questions live.
By the numbers
- The W3C and IETF announced on January 26, 2021 that WebRTC was officially a standard. (W3C press release: WebRTC is now an official standard)
