WebRTC signaling with WebSocket and Node.js
Blog post from LogRocket
WebRTC (Web Real-Time Communication) is a protocol that facilitates secure, bidirectional, real-time peer-to-peer communication for web applications, enabling the transfer of video, audio, and other media types through simple web APIs. It encompasses four main stages: signaling, connecting, securing, and communicating, relying on various underlying protocols like ICE, DTLS, and SRTP to ensure secure and efficient connections. WebRTC's widespread applications include video chatting, screen sharing, and live broadcasting, supported by JavaScript APIs such as Media Streams, RTCPeerConnection, and RTCDataChannel. Complementary to WebRTC, the WebSocket protocol offers real-time communication between clients and servers using a single TCP connection, enhancing media transfer capabilities across browsers and native platforms. A sample implementation using Node.js demonstrates how to set up a basic WebSocket server, illustrating how WebSocket signaling can handle inter-browser WebRTC calls effectively. Despite the evolving browser support for WebRTC, libraries like Adapter.js help mitigate compatibility issues, and tools such as LogRocket can provide insights into network performance and user experiences for Node-based applications.