MediaStreams

VoIP and WebRTC: Real Time Communication in the Browser VoIP and WebRTC bring real-time voice and video into the browser. VoIP is the broader idea of sending voice over IP networks, while WebRTC is a modern technology stack that lets browsers capture audio and video and connect with others. Together they power calls, meetings, and live collaboration directly in a webpage, without extra software. Key pieces of WebRTC include getUserMedia to capture media, RTCPeerConnection to send it, and a signaling path to exchange session data. WebRTC also runs under strong security by default: connections use DTLS-SRTP and are established over HTTPS. This makes browser calls safer and easier than older methods. ...

VoIP and WebRTC: Real-Time Voice and Video on the Web VoIP and WebRTC bring real-time communication directly to the browser. VoIP moves voice over the internet, while WebRTC adds live audio, video, and data channels without plugins. This makes web apps feel closer to native experiences, from customer support chats to remote education. WebRTC is built into modern browsers and relies on a few core ideas that developers can use in everyday work. ...

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. ...