Data-Channel

Real Time Communications with WebRTC Real Time Communications with WebRTC explains how browsers can send audio, video, and data directly, with help from signaling when needed. It focuses on practical ideas you can use to build simple, reliable experiences for users around the world. How WebRTC Works Signaling to exchange offers and ICE candidates. Negotiation creates a direct or relay path for media and data. NAT traversal uses ICE with STUN and, if required, TURN servers. Media streams travel through RTCPeerConnection; RTCDataChannel can carry text or files. Core Components RTCPeerConnection handles the network path, while getUserMedia captures local media. RTCDataChannel enables non-media data. A signaling server shares session descriptions and candidates but does not move media. STUN and TURN help when peers are behind firewalls. ...

VoIP and WebRTC: Real-time Communication on the Web Real-time communication in the browser has two main names: VoIP and WebRTC. VoIP stands for Voice over IP and has a long history in phone and desktop apps. WebRTC is a newer set of browser technologies that makes audio, video, and data sharing possible directly in the web page. Together, they let people talk, see, and work together without installing extra software. ...

WebRTC in the Real World: Building Flexible Communication WebRTC makes real-time audio, video, and data flow directly between browsers or apps. It avoids plugins and helps surface low-latency communication. In real life, you still rely on signaling and servers to locate peers, but the heavy work happens at the media layer once a connection is established. Architectures fall into three patterns. Mesh means every participant talks to everyone—easy to start but quickly heavy on bandwidth. SFU (Selective Forwarding Unit) forwards streams to all participants and scales well for groups. MCU (Multi-Party Control) mixes streams on the server; it can simplify clients but adds latency and cost. For most teams, SFU is the practical balance. ...