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.

sw.js

@@ -11,7 +11,7 @@ let DYNAMIC_CACHE = 'securebit-pwa-dynamic-v4.7.56';
 // Build stamp — rewritten by scripts/post-build.js on every release so this file's
 // bytes change each deploy. That is what makes the browser detect a new Service Worker,
 // reinstall it, drop stale caches and (via controllerchange) prompt the page to update.
-const SW_BUILD_VERSION = '1784757947117';
+const SW_BUILD_VERSION = '1784825757783';
 
 // Load version from meta.json on install
 async function getAppVersion() {