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.

meta.json

@@ -1,10 +1,10 @@
 {
-  "version": "1784757947117",
-  "buildVersion": "1784757947117",
-  "appVersion": "5.4.10",
-  "buildTime": "2026-07-22T22:05:47.156Z",
-  "buildId": "1784757947117-c98a01b",
-  "gitHash": "c98a01b",
+  "version": "1784825757783",
+  "buildVersion": "1784825757783",
+  "appVersion": "5.5.0",
+  "buildTime": "2026-07-23T16:55:57.822Z",
+  "buildId": "1784825757783-e544593",
+  "gitHash": "e544593",
   "generated": true,
-  "generatedAt": "2026-07-22T22:05:47.157Z"
+  "generatedAt": "2026-07-23T16:55:57.823Z"
 }