An open-source technology that enables real-time voice, video, and data communication directly between browsers or apps without requiring plugins.
Here’s a more detailed explanation:
What it is:
WebRTC is a set of protocols and APIs that allow browsers and mobile applications to establish peer-to-peer communication for voice calls, video conferencing, and data sharing. It eliminates the need for third-party software or dedicated VoIP clients, making real-time communication more accessible and efficient.
How it works:
Two browsers or applications initiate a connection using JavaScript APIs. WebRTC handles signaling (typically through a server using SIP, WebSockets, or other protocols) to exchange connection info.Once signaling is complete, a peer-to-peer connection is established. Voice, video, or data streams are encrypted and transmitted in real time between the endpoints. Built-in codecs and network traversal tools (e.g., STUN/TURN) ensure media flows through firewalls and NAT.
Benefits:
- No plugins required: Works natively in most modern browsers.
- Cross-platform: Supported on web, Android, and iOS apps.
- Real-time performance: Low latency for voice and video.
- Secure by design: Uses end-to-end encryption.
- Easy to integrate: Ideal for embedding calling features into websites or SaaS platforms.
Key components:
- MediaStream API: Captures audio and video from user devices.
- RTCPeerConnection: Establishes direct media channels between peers.
- RTCDataChannel: Supports real-time text or binary data exchange.
- STUN/TURN servers: Help route traffic across NAT/firewalls.
- Signaling protocol (custom): Initiates the connection and shares session details.
Why it’s beneficial:
WebRTC is perfect for businesses looking to offer seamless in-browser communication — like click-to-call, video support, or embedded customer service — without requiring users to install software. It reduces friction, supports remote interactions, and powers modern VoIP, telehealth, education, and collaboration tools.