Real Time Communication - WebRTC Tutorial
Autor: Idego Group

This tutorial introduces Web Real-Time Communication (WebRTC), an open-source C++ framework enabling real-time audio and video communication directly between web browsers. Modern browsers have evolved into powerful platforms capable of rendering graphics, accessing devices, and streaming data using complex network protocols, making specialized applications increasingly unnecessary.
WebRTC employs Interactive Connectivity Establishment (ICE) techniques using STUN and TURN protocols to overcome network address translation challenges. These technologies determine optimal data transmission paths between computers hidden behind NAT units.
The tutorial provides practical code examples for building a simple video chat application. First, it demonstrates accessing peripherals using the MediaDevices.getUserMedia() function, which returns a Promise that resolves to a MediaStream object including audio and video tracks. This requires user permission to access microphone, speaker, and camera.
Next, establishing peer connections uses the RTCPeerConnection interface. The process involves creating connection objects, setting remote peer callbacks, adding local tracks, and exchanging Session Description Protocol (SDP) objects through offer and answer mechanisms.
Next steps include implementing signaling servers for inter-browser communication or deploying dedicated media servers like Kurento for multi-endpoint connections and stream mixing in production environments.