WebRTCprotokollia

WebRTCprotokollia refers to the set of protocols that enable real-time communication capabilities directly within web browsers and mobile applications. These protocols facilitate the transmission of audio, video, and generic data between peers without the need for intermediary servers or plugins, although servers are often used for signaling and NAT traversal.

The core protocols underpinning WebRTC include Session Description Protocol (SDP) for describing the media session, and the Real-time Transport Protocol (RTP) and its companion Real-time Transport Control Protocol (RTCP) for the actual delivery of media data. Security is handled by Secure Real-time Transport Protocol (SRTP).

For establishing connections and discovering network addresses, WebRTC relies on technologies like Interactive Connectivity Establishment (ICE). ICE uses Session Traversal Utilities for NAT (STUN) to determine a peer's public IP address and port, and Traversal Using Relays around NAT (TURN) servers to relay traffic when direct peer-to-peer connections are not possible due to restrictive network configurations.

Signaling, which involves exchanging session descriptions, network information, and control messages to set up and tear down calls, is not part of the WebRTC specification itself. Instead, developers typically implement their own signaling mechanisms, often using technologies like WebSockets, to facilitate this process between clients. This allows for flexibility in how real-time communication applications are built.