Browser-Based

VoIP and WebRTC: Real-Time Communication for the Web VoIP and WebRTC offer real-time voice and video over the internet. VoIP, or Voice over Internet Protocol, supports phone-like calls over data networks. WebRTC, short for Web Real-Time Communications, lets browsers talk directly with audio and video without plugins. Together, they power modern calls, video meetings, and live support in web and mobile apps. The goal is simple: fast, secure communication that works across devices and networks. ...

WebRTC Case Studies: Real-Time Communication Apps WebRTC makes real-time audio and video possible directly in the browser, with no plugins. Real projects mix signaling, NAT traversal, and optional media servers to handle more users and tougher networks. The following case studies show practical choices and trade-offs that teams face daily. Case Study: Family Video Chat A small team built a family chat app for two to four people. For one-to-one calls, they used a direct peer-to-peer path to keep latency low and avoid extra servers. When a second or third person joined, they switched to a selective forwarding unit (SFU) to forward media streams and reduce bandwidth at each participant. Signaling ran over WebSocket, with public STUN servers during testing and TURN servers for restricted networks in production. Privacy and encryption (DTLS-SRTP) were maintained along the media path, and the UI emphasized simplicity so users focused on talking, not on setup. ...