STUN/TURN

VoIP and WebRTC: Real-Time Communication VoIP and WebRTC make live communication possible over the internet. VoIP is the broad idea of delivering voice over IP networks. WebRTC is a set of browser APIs that let you share audio, video, and data directly between users without plugins. Together they power video calls, voice chats, and real-time collaboration in many services. How they work in simple terms: signaling sets up the call and negotiates how to send media. Once a path is agreed, media streams travel as RTP packets. WebRTC includes built-in encryption and handles much of the media path inside the browser, while traditional VoIP systems often rely on external servers and protocols like SIP or MGCP. This division helps teams pick the right tool for the job. ...

Real-Time Communications: WebRTC in Practice Real-time communications in the browser have become common. WebRTC lets browsers and apps send audio, video, and data directly. The core pieces are getUserMedia for cameras and mics, RTCPeerConnection for media paths, and RTCDataChannel for chat or file transfer. Signaling, the exchange of offers and ICE candidates, happens outside WebRTC, so you need a simple signaling service to exchange offers, answers, and ICE candidates. In practice, you decide how the media travels. A direct peer path works for small, one-to-one calls. For larger groups or when you want to save bandwidth, a media server such as an SFU helps distribute streams. Security and privacy also matter: use encrypted connections and explain permissions clearly to users. ...