RTCT

The Real-Time Communication (RTC) component, often abbreviated as RTCT, refers to a set of technologies and protocols that enable direct, real-time communication between two or more parties over a network. This communication typically involves the exchange of data such as audio, video, and text, with minimal latency.

RTCT functionalities are crucial for applications like video conferencing, instant messaging, voice calls over IP (VoIP), and online gaming. Key technologies that fall under the RTCT umbrella include Session Initiation Protocol (SIP) for establishing, modifying, and terminating real-time sessions, and Real-time Transport Protocol (RTP) for delivering audio and video data.

WebRTC, a prominent example of RTCT, is an open-source project that provides web browsers and mobile applications with real-time communication capabilities. It allows peer-to-peer connections for audio, video, and data sharing directly within a web browser, without the need for plugins or separate downloads.

The underlying infrastructure for RTCT often involves signaling servers to manage session setup and traversal servers (like STUN and TURN) to help devices establish connections even when behind firewalls or Network Address Translators (NATs). The objective of RTCT is to facilitate seamless, interactive communication experiences for users across various platforms and devices.