WebRTC, or Web Real-Time Communication, is an open-source project designed to facilitate real-time, peer-to-peer communication of data, audio, and video between web browsers and apps using JavaScript APIs. Since its inception by Google in 2011, it has been embraced by major companies like Apple, Microsoft, and Mozilla, leading to widespread adoption across sectors such as e-learning, telehealth, and collaboration tools. WebRTC enables high-quality voice and video calls with low latency and includes features like encryption for secure data transfer. It utilizes various protocols such as STUN and TURN for network traversal and employs JavaScript APIs like RTCPeerConnection and RTCDataChannel to manage connections and data exchange. WebRTC supports a range of audio and video codecs, ensuring compatibility and performance in different network conditions. Alternative technologies like SIP, XMPP, and MQTT offer different capabilities for real-time communication, but WebRTC's integration with modern web browsers positions it as a robust solution for applications requiring real-time multimedia communication. As internet connectivity and the demand for real-time apps increase, WebRTC is expected to continue evolving, with significant implications for industries like gaming, healthcare, and customer service.