Three real bugs, one smoke-test session's worth of progress.
1. RELAY: wrong advertised addr in CallSetup
The direct-call CallSetup computed `relay_addr = addr.ip()` where
`addr = connection.remote_address()` — i.e. the CLIENT'S IP, not the
relay's. So the relay was telling both parties "the call room is at
the answerer's IP:4433", which meant each client dialed either the
other client (no server listening) or themselves. Both endpoint.connect
calls hung forever and the call never happened.
Fix: compute the relay's own advertised IP once at startup. If the
listen addr is 0.0.0.0, probe the primary outbound interface via the
classic UDP-bind-and-connect(8.8.8.8:80) trick to discover the LAN
IP the OS would use to reach external hosts. Thread the resulting
advertised_addr_str into the CallSetup sender for both parties.
2. RELAY: accept loop serialized QUIC handshakes
Previously the main accept loop called `wzp_transport::accept` which
did both `endpoint.accept().await` AND `incoming.await` (the server-
side QUIC handshake). A single slow handshake therefore blocked every
subsequent client from being accepted. Unroll the helper here and
move `incoming.await` into the per-connection spawned task, so every
handshake runs in parallel. Also log "accept queue: new Incoming",
"QUIC handshake complete", and "QUIC handshake failed" so we can tell
immediately whether a client's packets are reaching the relay at all.
3. ANDROID: playout was routed to the silent in-call stream
The Oboe playout stream was configured with Usage::VoiceCommunication,
which routes to the Android in-call earpiece stream. That stream is
silent unless the Activity has called AudioManager.setMode(
IN_COMMUNICATION) and, even then, only the earpiece/BT headset get
audio (not the loud speaker). Result: android→mac calls worked
because mac had a normal media output, but mac→android calls were
silent even though packets flowed through the relay just fine.
Switch to Usage::Media + ContentType::Speech so Oboe routes to the
loud speaker and uses the media volume slider. A later polish step
will wire setMode + setSpeakerphoneOn from MainActivity.kt so we can
go back to VoiceCommunication for AEC and proximity-sensor routing.
Plus: heartbeat tracing every 2s in the send/recv tasks — frames_sent,
last_rms, last_pkt_bytes, short_reads on the send side; decoded_frames,
last_decode_n, last_written, decode_errs on the recv side. Will make the
next "no sound" regression trivial to localize.
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