The protocol used for real-time communication in web browsers is WebRTC, short for Web Real-Time Communication.
WebRTC is a set of browser APIs and supporting standards that enables web pages to handle live audio, video, and data communication without requiring a separate plugin or native application. It is commonly used for browser-based video calls, voice calls, screen sharing, and peer-to-peer file transfers.
A WebRTC connection may communicate directly between peers, but establishing that path across home routers and firewalls requires networking techniques such as ICE, STUN, and TURN. WebRTC also uses secure media transport and standardized codecs so different browsers can interoperate.
SIP is a signaling protocol often used to initiate voice and video sessions, while HTTP is primarily a request-and-response web protocol. WebSocket supports persistent, two-way browser-server communication, but it is not itself the browser’s complete real-time audio-video framework. WebRTC can use signaling systems—including WebSocket—to arrange a call, but WebRTC provides the communication capability.