feat(webrtc): end-to-end encrypted voice & video calls with adaptive codecs
Add 1:1 voice and video calling over the existing SAS-verified peer connection. Audio and video tracks ride the same RTCPeerConnection as the chat, bundled onto one DTLS-SRTP transport, so media inherits the session's end-to-end encryption. SDP offer/answer is renegotiated in-band over the verified data channel — no signalling server, so the media's DTLS fingerprints are authenticated end-to-end. Calls are gated on a connected, SAS-verified session. Codecs & adaptation: - Opus tuned for lossy links (in-band FEC, DTX, RED redundancy); audio is bandwidth-prioritised and never throttled. - VP9/AV1 single-encoding SVC with H.264/VP8 fallback; video degrades by spatial/temporal layer. - Runtime NetworkAdaptationController trims video bitrate on loss/RTT and recovers as the link clears — no renegotiation. Live connection-quality indicator (Excellent/Good/Fair/Weak) in the call UI. In-call controls: mute, camera on/off (voice→video upgrade in-band), camera flip, minimize-to-widget, hang up, and accept/decline for incoming calls. Production logging disabled (DEBUG_MODE=false); temporary call diagnostic logger removed. Codec rationale in docs/webrtc-config.md.
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@@ -12,6 +12,7 @@ import '../components/ui/Roadmap.jsx';
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import '../components/ui/CommunityCTA.jsx';
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import '../components/ui/FileTransfer.jsx';
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import '../components/ui/IceServerSettings.jsx';
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import '../components/ui/CallUI.jsx';
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// Expose to global for legacy usage inside app code
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window.EnhancedSecureCryptoUtils = EnhancedSecureCryptoUtils;
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Vendored
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@@ -37,6 +37,7 @@
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import(`../components/ui/Roadmap.jsx?v=${timestamp}`),
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import(`../components/ui/FileTransfer.jsx?v=${timestamp}`),
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import(`../components/ui/IceServerSettings.jsx?v=${timestamp}`),
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import(`../components/ui/CallUI.jsx?v=${timestamp}`),
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]);
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// Components are automatically registered on window by their respective modules
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