webrtc UDPTLSRTPSAVPF

WebRTC (Web Real-Time Communication) is a free, open-source project that enables real-time communication between browsers, mobile applications, and other devices through simple APIs. It uses a combination of protocols and technologies such as UDP (User Datagram Protocol), TLS (Transport Layer Security), RTP (Real-time Transport Protocol), and SAVPF (Secure Audio / Video Profile with Feedback).

UDP is a transport protocol that provides low-latency, connectionless communication. It is used by WebRTC to transmit audio and video data between peers in real-time.

TLS is a cryptographic protocol that provides secure communication over the internet. It is used by WebRTC to encrypt and authenticate data transmissions between peers.

RTP is a protocol that provides real-time transmission of audio and video data over IP networks. It is used by WebRTC to package and transmit audio and video data between peers.

SAVPF is a profile that provides secure transmission of audio and video data with feedback. It is used by WebRTC to ensure that data transmissions are secure and reliable.

In summary, WebRTC uses UDP for low-latency, real-time communication, TLS for secure communication, RTP for real-time transmission of audio and video data, and SAVPF for secure transmission with feedback. These protocols and technologies work together to enable seamless, secure, and real-time communication between devices over the internet