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WebRTC
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How to Get Started With WebRTC: Intro to Browser APIs
WebRTC provides browser APIs for real-time voice, video, and data sharing without plugins, with mandatory encryption via DTLS-SRTP and DTLS; the standard became a W3C Recommendation in January 2021. The core APIs are getUserMedia for capturing audio/video from user devices, RTCPeerConnection for peer-to-peer media exchange via offer/answer and ICE candidates, and RTCDataChannel for arbitrary data transfer. WebRTC is supported by all major browsers and used by services like Google Meet, Discord (which uses media servers for group calls to avoid expensive peer-to-peer), and AI voice agents. Twilio offers Video and Voice JavaScript SDKs that handle signaling and media relay, enabling calls to phone numbers and SIP endpoints. The post includes a camera preview example using getUserMedia that requires a secure origin and can be served with Python, then directs readers to Twilio’s quickstart tutorials for building video and voice applications.
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