Android: voice calls, peer to peer by default

The one third-party library in this app, and unavoidable: the alternative to
`io.github.webrtc-sdk:android` is hand-rolling ICE, DTLS-SRTP and an audio
pipeline. Worth naming rather than absorbing quietly, because everything else
here is deliberately auditable and this is a large native blob that is not.

🔴 **Peer to peer by default.** ICE tries a direct path first — same network,
then each side's address as seen through NAT — and only falls back to relaying
through TURN when both ends sit behind NAT that refuses one. `iceTransportsType`
is ALL rather than RELAY: forcing every call through the server is the opposite
of the default chosen here.

**Signalling rides the ratchet, not the relay.** The offer and answer are
ordinary content envelopes, so the SDP — codecs, and every candidate address —
is end-to-end encrypted like any other message. A relay that could read
signalling would learn both parties' addresses even on a call it never carries.

⚠️ **A direct call shows the other person your IP address.** That is what makes
it direct, and it is not something to find out afterwards. Explained before the
FIRST call, and it waits for an answer — once, not every time, because a prompt
people see repeatedly is one they stop reading. The acknowledgement lives in
ordinary preferences rather than the vault: it is not a secret, and storing it
encrypted would re-show the warning on every sign-out, which is how a one-time
prompt becomes one people click past.

⚠️ **No TURN server is deployed** — `FN_TURN_SECRET` and `FN_TURN_HOST` are
unset, so `/e2e/turn` answers 503. Handled rather than hidden: STUN alone still
connects through most NATs, and the banner says plainly that this server has no
relay so the call connects only if a direct path exists. "Calling did not
connect", with no reason, is the kind of failure people blame on the app for
ever.

Trickle ICE is off. Candidates are gathered before the offer is sent, so
signalling is one encrypted envelope each way rather than a stream of them —
a second or two slower to connect, and far simpler over a channel that is
polled rather than pushed.

⚠️ The microphone is held by the audio source, not the peer connection.
Closing one without disposing the other leaves the mic live with no call
attached, which is the worst failure this particular feature has available.

⚠️ **WebRTC's native libraries dominate the APK**: ~41 MB across four ABIs, of
which x86 and x86_64 (~24 MB) exist only for emulators. Release is restricted
to arm64-v8a and armeabi-v7a — 102 MB debug becomes a 22 MB release — while
debug keeps all four so it still runs on an emulator. Restricted rather than
split per-ABI on purpose: this app is handed to testers as one file, and
per-ABI outputs turn "send them the APK" into "send them the right APK".

A second offer while busy is declined rather than dropped, so the caller hears
something. Group calls are deliberately absent: different key handling, and an
enabled-looking button that cannot work is worse than none.

Tested: 108 engine, 67 client, none failing; debug and release APKs both build.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>

app/src/main/AndroidManifest.xml

@@ -12,6 +12,9 @@
     <!-- The app is usable without a microphone; it simply cannot record. -->
     <uses-feature android:name="android.hardware.microphone" android:required="false" />
 
+    <!-- Voice calls: WebRTC switches the audio mode and routes to earpiece. -->
+    <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+
     <application
         android:allowBackup="false"
         android:label="@string/app_name"