How to Integrate Live Audio and Video Into iOS Apps Using WebRTC

WebRTC is a versatile, open-source protocol developed by Google that facilitates low-latency, secure peer-to-peer communication for web and mobile applications, enabling audio, video, and data exchange without the need for plugins. It supports various platforms, including web browsers and mobile operating systems like iOS and Android, and offers advantages such as cost-effectiveness, security, and cross-platform functionality. Developers can integrate WebRTC into iOS applications using platforms like LiveKit or Swift packages, employing signaling servers to manage connections and enable features like audio and video conferencing, file sharing, and real-time communication. Signaling and negotiation are crucial for establishing connections between devices, with Session Description Protocol (SDP) managing session metadata and STUN servers facilitating communication through NAT gateways. WebRTC's flexibility allows developers to choose their preferred signaling protocols, incorporating servers like Node.js or Swift with WebSocket to manage connectivity and enhance user communication experiences.