WebRTC Explained: The Technology Behind Browser-Based Video Calls

WebRTC (Web Real-Time Communication) is the open standard that lets browsers exchange audio, video, and data directly, without a plugin or downloaded application. It is the technology that made 'click a link and you're in the meeting' possible, replacing an earlier generation of video conferencing that universally required installed software or browser plugins like Flash.

What WebRTC actually provides

• Direct peer-to-peer media capability: browsers can negotiate and exchange audio/video streams with each other using standardized protocols, without needing a proprietary client application.
• Built into every modern browser: Chrome, Firefox, Safari, and Edge all implement WebRTC natively — no download, no plugin, no separate installation step for the end user.
• Real-time transport optimized for low latency: unlike a typical web page load, WebRTC uses transport protocols (SRTP for media) designed for the millisecond-level responsiveness a live conversation requires.
• Adaptive quality handling: WebRTC includes mechanisms for adjusting video quality dynamically based on real-time network conditions — the technical foundation underneath adaptive bitrate.

Why this matters for meeting platforms

Before WebRTC's maturity, browser-based video calling required plugins (often Flash, later deprecated for security reasons) or forced users into downloaded native applications. WebRTC is why a platform like MeetOye can offer genuinely browser-native meetings — a guest clicking a link joins immediately in Chrome or Safari with no install prompt — while platforms built on older architectures still push users toward app downloads even when a web version technically exists.

The architecture question: peer-to-peer versus SFU

WebRTC supports direct peer-to-peer connections, which work well for 1:1 calls but scale poorly for groups — each additional participant multiplies the connections everyone's device must maintain directly. Production meeting platforms for group calls instead route media through a Selective Forwarding Unit (SFU): a server that receives each participant's stream once and forwards only what each other participant needs to see, rather than every device connecting to every other device directly. This is why group video calls remain smooth even with a dozen participants — the heavy lifting happens server-side, not by asking each laptop to handle eleven simultaneous peer connections.

What WebRTC does not handle by itself

WebRTC provides the media transport layer, not the full meeting experience. Signaling (how participants find and connect to each other), recording, AI transcription, and the actual user interface are all built on top of WebRTC by the platform, not provided by the standard itself. This is why two WebRTC-based platforms can have identical underlying transport technology and still differ enormously in reliability, features, and quality — the standard is the foundation, not the finished building.