WebRTC vs WebSockets: What Are the Differences?
Blog post from Stream
Real-time communication technologies like WebRTC and WebSockets are fundamental to modern applications, enabling seamless interactions across various platforms. WebRTC is an open-source protocol designed for peer-to-peer communication, facilitating voice, video, and data sharing without intermediary servers, making it ideal for video streaming and interactive applications. Its main components include RTCPeerConnection, MediaStream, and Data Channel, and it employs JavaScript APIs to integrate real-time communication into web browsers. WebSockets, on the other hand, provide a persistent client-server connection for instant bidirectional data exchange, minimizing latency and bandwidth use, making it suitable for real-time chat, online gaming, and collaborative environments. While WebRTC excels in applications requiring secure, high-quality media streaming, WebSockets is better suited for scenarios requiring low-latency, full-duplex communication, such as live notifications and updates. Though they serve different purposes, WebRTC and WebSockets can complement each other in applications needing both peer-to-peer transmission and real-time message exchange, with WebSockets often handling signaling and data synchronization. Both technologies are continuously evolving, integrating with emerging trends like AI and new protocols like WebTransport, enhancing their capabilities and expanding their applications in fields such as telehealth, online education, and corporate communications.