rauntímasamskipti

Rauntímasamskipti (real-time communication, RTC) refers to a set of technologies and practices that enable the immediate exchange of audio, video and data between endpoints, allowing participants to interact as if in the same moment. Real-time communication emphasizes low latency and interactivity, and it differs from asynchronous, store-and-forward messaging.

The technical core of RTC combines signaling and media. Signaling establishes, modifies and ends sessions and can use protocols such as SIP, XMPP, or WebRTC signaling. The media plane transports the actual content—audio, video and data—typically over RTP/RTCP. WebRTC, a widely adopted framework, supports browser-based RTC with encryption (DTLS-SRTP) and NAT traversal (ICE, STUN, TURN).

Architectures vary and can be peer-to-peer, client-server, or media-server-based. In browser-centric RTC, endpoints run in web browsers or applications, while signaling servers coordinate sessions. For larger conferences, media servers such as selective forwarding units (SFU) or multipoint control units (MCU) provide scalability and mixing capabilities.

Applications of RTC include voice and video calls, video conferencing, live collaboration, telemedicine, real-time customer support, online education and interactive gaming. Core benefits are immediacy and interactivity, but requirements include reliable low-latency networks, sufficient bandwidth and robust security.

Key considerations include quality of service (latency, jitter, packet loss, bandwidth), interoperability across platforms, and privacy and security measures. The field has evolved from earlier standards like H.323 and SIP to contemporary WebRTC-driven implementations that enable broad, real-time communication across devices and networks.