WebRTC provides a standardized profile for real-time communication that enables interoperability between browsers without plugins. It defines client-side APIs for audio and video calling as well as other real-time communication capabilities. The WebRTC architecture includes the API, codecs, transport mechanisms like STUN and TURN, and network I/O that allow real-time apps to run directly in browsers. Signaling is required to establish connections between users, and the standardization of WebRTC aims to improve interoperability compared to proprietary solutions. However, interoperability is not always in the best interests of businesses. Ultimately, the API is more important than the underlying protocols it uses.