This GitHub repository details "Alfajer," a private 1-to-1 video/audio calling Progressive Web App (PWA) built using WebRTC for media transmission and a custom Bun+ElysiaJS WebSocket signaling server. The architecture emphasizes privacy with no persistent storage, logs, or accounts, and features end-to-end encryption notes via RTCRtpScriptTransform, though it's currently disabled pending a cross-browser capability handshake. Recent commits detail a "5-tier UX overhaul" and automation for TURN secret rotation.
The immediate signal is that Alfajer is moving from a basic private calling demo into a more complete calling stack, with explicit UX polish, chat, screen share, and automated TURN credential rotation. That matters in the streaming delivery and real-time comms ecosystem because the repository pairs privacy claims with concrete operational details: in-memory rooms, no logs, and negotiated E2EE on top of WebRTC’s DTLS-SRTP. The clearest next marker is whether the RTCRtpScriptTransform path stays disabled only for compatibility, or appears enabled across the latest Chrome and Safari browsers described in the README.
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