WebRTC (Web Real-Time Communication) is an open-source framework that enables real-time audio, video, and data communication directly between web browsers and mobile applications without requiring plugins or additional software. Originally designed for video conferencing, WebRTC is increasingly used for ultra-low-latency live streaming.
How WebRTC works
WebRTC establishes a peer-to-peer connection between two endpoints using a signaling server to exchange connection metadata. Once the connection is established, media flows directly between peers (or through a relay server if needed), achieving sub-second latency.
Signaling: exchange of SDP (Session Description Protocol) offers and answers
ICE (Interactive Connectivity Establishment): finds the best network path between peers
DTLS: encrypts the media transport layer for security
SRTP: delivers encrypted audio and video in real time
Streaming latency comparison
Latency is the delay between capture and display. Different streaming technologies offer different latency trade-offs.
Technology
Typical latency
Best for
WebRTC
< 1 second
Interactive events, auctions, sports betting
LL-HLS / LL-DASH
2-5 seconds
Live events, Q&A sessions, webinars
Standard HLS
15-30 seconds
Broadcasts, concerts, scheduled live events
RTMP
3-5 seconds
Ingest from encoders (not viewer-facing)
WebRTC use cases in video
Beyond video conferencing, WebRTC has found several applications in the video streaming industry where ultra-low latency is critical.
Live auctions: bidders need to see the current item and price in real time
Sports betting: viewers need to see the action before placing bets
Interactive live events: audience participation, polls, and Q&A in real time
Remote production: camera operators and directors collaborate with sub-second delay
Surveillance and monitoring: real-time video feeds from cameras
How Videas uses WebRTC
Videas integrates WebRTC for live streaming scenarios where ultra-low latency is essential. The Stream product supports WebRTC ingest for sub-second latency delivery to viewers, combined with automatic fallback to HLS for larger audiences. This hybrid approach provides the best of both worlds: real-time interactivity for a core audience and scalable delivery for thousands of concurrent viewers.
WebRTC stands for Web Real-Time Communication. It is an open-source framework for real-time audio, video, and data transfer between browsers.
WebRTC offers sub-second latency, making it ideal for interactive events. However, it doesn't scale as easily as HLS for large audiences. Most professional platforms use a hybrid approach: WebRTC for interactivity and HLS for mass distribution.
No. WebRTC is natively supported by all modern browsers (Chrome, Firefox, Safari, Edge) without any plugins or extensions. This is one of its key advantages for real-time video communication.