WebRTCtä

WebRTCtä is a term sometimes used to refer to WebRTC technology, a set of browser-based APIs that enable real-time audio, video, and data communication without plugins. The core components are designed to run in web browsers and on mobile devices, leveraging peer-to-peer connections when possible.

The WebRTC suite includes getUserMedia for capturing media from devices, RTCPeerConnection for negotiating and maintaining peer-to-peer media streams, and RTCDataChannel for sending arbitrary data between participants. Signaling—exchanging session descriptions and network candidates—is performed by the application rather than defined by WebRTC itself. NAT traversal is handled through ICE with STUN and TURN servers.

Security is integral to WebRTC: media is encrypted by default using DTLS-SRTP, and data channels are encrypted with DTLS as well. The standards are maintained by the W3C and the IETF, with ongoing work to improve privacy, performance, and interoperability.

In practice, WebRTC enables use cases such as video conferencing, voice calls, screen sharing, file transfer, and collaborative applications. It is implemented in major browsers including Chrome, Firefox, Safari, and Edge, and is supported on many mobile platforms. Real-world deployments require compatible signaling and, often, media servers for conferencing, recording, or broadcasting.