Fixes from real-world 5G↔Starlink testing:
NAT tickle fix:
- tokio::net::UdpSocket::bind() doesn't set SO_REUSEADDR, so binding
to the same port as quinn silently failed. Now uses socket2::Socket
with explicit SO_REUSEADDR + SO_REUSEPORT (via libc on unix).
- Tickle now logs success/failure for debugging.
Diagnostic fixes:
- connect:dual_path_race_start shows both dial_order_raw and
dial_order_smart so we can see what filtering removed
- Grace-period timeout (relay wins first, direct still running)
now fills "timeout:grace" diags for unrecorded candidates
- Previously candidate_diags was empty when relay won the race
Dependencies:
- Added socket2 = "0.5" to wzp-client
593 tests pass, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Major P2P improvements for cross-network calls:
Smart candidate filtering (smart_dial_order):
- Strip LAN candidates when peer's public IP differs from ours
(172.16.x.x is unreachable from a different network)
- Strip all IPv6 candidates (Phase 7 disabled, wastes dial slots)
- Only keep mapped + reflexive for cross-network calls
- LAN candidates preserved when both peers share the same public IP
Acceptor NAT tickle:
- A-role sends a 1-byte UDP packet to each peer candidate BEFORE
accepting. This opens the NAT pinhole for return traffic from
the Dialer's IP — critical for address-restricted NATs that only
allow inbound from IPs they've seen outbound traffic to.
- Uses SO_REUSEADDR on the same port as the quinn endpoint.
Direct timeout increased from 2s to 4s:
- Cross-network QUIC handshakes through CGNAT can take 2-3s
- 2s was too aggressive for 5G/LTE networks
Diagnostic fix:
- Record "timeout:4s" for candidates still in-flight when the
timeout fires (previously these had no diagnostic entry)
5 new tests for smart_dial_order edge cases.
593 tests pass, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The callDebugBuffer.length=0 in showCallScreen() ran AFTER the
connect command returned, wiping all connect: events (path_negotiated,
race_start, race_done, candidate_diags). Only media: events survived
because they arrived after the clear.
Removed all automatic buffer clearing. The reverse().find() already
handles stale data by picking the most recent event. The manual
"Clear log" button (line 624) is the only way to clear now.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Clearing callDebugBuffer in showConnectScreen() wiped all debug
events the moment a call ended, so the user saw empty logs. Moved
the clear to showCallScreen() instead — the buffer is reset at the
START of a new call, not the end. This way:
- After hanging up, all events from the call are still visible
- Starting a new call clears stale data from the previous one
- The reverse().find() for P2P badge still gets fresh data
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
findLast() requires Chrome 97+ / Android WebView 97+. Older Android
devices crash with TypeError in pollStatus(), killing all status
updates including the debug log. Use [...arr].reverse().find() which
works everywhere.
Also pass peerMappedAddr in the direct-call connect invoke.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Added CandidateDiag struct to RaceResult with per-candidate:
- address attempted
- result (ok / skipped:ipv6 / error:reason)
- elapsed time in ms
Surfaced in call-debug events:
- connect:dual_path_race_start now includes dial_order + peer_mapped
- connect:dual_path_race_done now includes candidate_diags array
Upgraded dual_path tracing from debug to info for IPv6 skips and
dial failures so they appear in logcat/console.
Helps diagnose why P2P fails on specific networks (5G CGNAT,
address-restricted NATs, etc).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The callDebugBuffer persisted across calls, so .find() returned the
path_negotiated event from Call 1 (P2P Direct) when rendering the
badge during Call 2 (Relay). Two fixes:
1. Clear callDebugBuffer in showConnectScreen() between calls
2. Use .findLast() instead of .find() so the most recent event wins
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
End-to-end integration of sequential port prediction:
- place_call: spawns background detect_port_allocation() + sends
HardNatProbe signal after offer (doesn't delay call setup)
- answer_call: same for AcceptTrusted answers (privacy mode skips)
- Signal recv loop: stashes HardNatProbe in SignalState.peer_hard_nat_probe
- connect: reads peer's probe, if Sequential{delta} runs predict_ports()
and adds predicted addrs to PeerCandidates.local for the dual-path race
- parse_sequential_delta() helper for "sequential(delta=N)" strings
The full flow: both peers independently detect their NAT's port
allocation, exchange HardNatProbe via relay, and the connect command
uses the peer's sequence to predict which ports to dial — all before
the dual-path race starts.
588 tests pass, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- PROGRESS.md: hard NAT Phase A, relay cross-wiring, 588 tests
- ARCHITECTURE.md: hard NAT port prediction diagram + pattern table
- PRD-p2p-direct.md: Phase 8.6 split into a/b/c/d with status
- PRD-hard-nat.md: Phase A done, B signal ready, effort table updated
- PRD-netcheck.md: port_allocation field + probe documented
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Phase A of hard NAT traversal (PRD-hard-nat.md):
- PortAllocation enum: PortPreserving / Sequential{delta} / Random / Unknown
- detect_port_allocation(): sequential STUN probes from single socket,
analyzes port sequence for allocation pattern
- classify_port_allocation(): pure function with jitter tolerance,
wraparound handling, 60% threshold for noisy sequences
- predict_ports(): generates target port range from last_port + delta
- HardNatProbe signal message: carries port_sequence, allocation
pattern, external_ip for peer coordination
- Relay forwards HardNatProbe to call peer
- Netcheck gains port_allocation field + format_report display
588 tests pass (17 new), 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
4-phase design:
A. Port allocation pattern detection (sequential vs random)
B. Sequential port prediction (~80% success, <2s)
C. Birthday attack for random NATs (98% success, ~10s)
D. Hybrid waterfall with background relay-to-direct upgrade
Taskmaster tasks #84-87 added.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
build-android-docker.sh builds the old Kotlin app in android/app/
(18M APK), not the live Tauri app (209M). Renamed to
build-android-docker-LEGACY.sh so it's never picked by accident.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Edition 2024 on local macOS auto-resolves the Emitter trait, but the
Docker builder's Rust/Tauri version requires the explicit import for
AppHandle::emit() to resolve. Keeps the warning locally to avoid
breaking the CI build.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
After 30s stable at a tier, the AdaptiveQualityController actively
probes the next tier up by switching the encoder and observing for 5s.
If loss/RTT stay within the target tier's thresholds, the upgrade
commits. If >1 bad report, the probe aborts with a 60s cooldown.
Probing is disabled on cellular (studio tiers aren't classified there)
and skipped when already at Studio64k (highest tier).
This complements the passive upgrade path (10 consecutive good reports)
by actively discovering that a path can sustain higher quality, rather
than waiting for the classification to drift upward.
New: ProbeState struct, check_probe() method, 4 constants, 5 tests.
377 tests passing.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
#15 - Replay mode: --replay <file.wzp> reads captured sessions offline,
feeds packets through the same stats engine, prints summary.
CaptureReader mirrors CaptureWriter's binary format.
#16 - HTML report: --html <report.html> generates self-contained HTML
with Chart.js line charts (loss% and jitter over time per-stream),
participant summary table, dark theme. Works with live sessions
(after exit) or replay mode.
#17 - Encrypted decode: --key <hex> flag accepted and stored. Full audio
decode deferred — SFU E2E encryption requires session key + nonce
context from both endpoints. Header-only analysis (loss, jitter,
codec, packet count) works without decryption.
Usage:
wzp-analyzer --replay session.wzp --html report.html
wzp-analyzer relay:4433 --room test --capture out.wzp --html report.html
372 tests passing, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
P2P calls now adapt codec quality based on observed network conditions,
matching what relay calls already had.
Three-layer implementation:
- QualityReport::from_path_stats(): construct reports from local quinn
stats (loss%, RTT, jitter) without needing relay-generated reports
- CallEncoder.pending_quality_report: one-shot attachment to next
source packet (consumed on encode, not repeated)
- Engine send tasks: generate quality report every 50 frames (~1s)
from quinn_path_stats() and attach via set_pending_quality_report()
- Engine recv tasks: self-observe from own QUIC path stats every 50
packets, feed to AdaptiveQualityController for P2P adaptation
(works even if peer isn't sending quality reports yet)
Both relay and P2P calls now have adaptive quality. On relay calls,
both peer-sent reports AND local observations feed the controller.
Hysteresis (3 consecutive bad reports to downgrade) prevents thrashing.
372 tests passing (+4 new: from_path_stats encoding, clamping, zero
values, encoder quality report attachment).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Full analysis of relay lock contention with precise inventory of every
lock acquisition in the hot path. Evaluates 4 design options:
A) Per-room Arc<Mutex<Room>> (recommended — 100x improvement for multi-room)
B) DashMap (good but less explicit)
C) Channel-based fan-out (over-engineered for current scale)
D) Snapshot-on-change via arc-swap (best perf, more complex)
Phase 1: per-room locks, Phase 2: federation lock fix, Phase 3: quality
tracking out of critical path. Estimated 1.5-2.5 days total.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Partial reads from the capture ring consumed samples that were then
discarded when the send loop retried from buf[0]. For 20ms codecs this
was invisible (single Oboe burst fills 960 samples in one read), but
40ms codecs (Opus6k, 1920 samples) needed 2 bursts — the first partial
read consumed 960 real samples and threw them away.
Result: Opus6k produced ~11 frames/s instead of 25 (~44% of expected).
Fix: expose wzp_native_audio_capture_available() and check it before
reading, matching the desktop capture_ring.available() pattern. Partial
reads no longer occur because we only read when enough samples exist.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
frame_samples was immutable — when adaptive quality switched from 20ms
(Opus24k, 960 samples) to 40ms (Opus6k, 1920 samples), the send loop
kept reading 960 samples and feeding half-sized frames to the encoder.
This caused Opus6k to produce ~11 frames/s instead of 25, making audio
choppy.
Fix:
- frame_samples is now mut and updated on profile switch
- buf sized for max frame (1920) with frame_samples-bounded slices
- RMS, mute, encode, and capture reads all use &buf[..frame_samples]
- Applied to both Android and desktop send tasks
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Keystores are gitignored so git reset --hard deletes them. The build
script now copies them from a persistent $BASE_DIR/data/keystore/ cache
into the source tree before building. This ensures both primary and alt
servers always have signing keys available.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Extend Tier enum from 3 to 6 levels: Studio64k/48k/32k + Good +
Degraded + Catastrophic with asymmetric hysteresis (down:3, up:5,
studio:10)
- Handle QualityDirective signals in both desktop and Android engines
— relay-coordinated codec switching now works end-to-end
- Add periodic TAP STATS to debug tap: packets in/out, fan-out avg,
seq gaps, codecs seen (every 5s)
- Mark task #2 done (ParticipantInfo in federation signals already
implemented)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
build.sh was producing unsigned APKs because it reimplemented the Docker
build inline without the signing step from build-tauri-android.sh. Now
uses the same pipeline: find keystore (release preferred, debug fallback),
zipalign -f 4, apksigner sign with keystore credentials.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
find was picking up a cached 384MB debug APK over the fresh 25MB release
APK because the old file was listed first. Now:
1. Delete all APKs before the build starts (clean slate)
2. On upload, prefer *release*.apk over any other match
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Wire AdaptiveQualityController into desktop engine send/recv tasks
(mirrors Android pattern: AtomicU8 pending_profile, auto-mode check)
- Wire same into Android engine send task (was only in recv before)
- QualityDirective SignalMessage variant for relay-initiated codec switch
- ParticipantQuality tracking in relay RoomManager (per-participant
AdaptiveQualityController, weakest-link tier computation)
- Relay broadcasts QualityDirective to all participants when room-wide
tier degrades (coordinated codec switching)
- Oboe stream state polling: poll getState() for up to 2s after
requestStart() to ensure both streams reach Started before proceeding
(fixes intermittent silent calls on cold start, Nothing Phone A059)
Tasks: #7, #25, #26, #31, #35
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Without clearCommunicationDevice(), the BT headset stays locked in SCO
mode after the call. Media playback (video, music) can't route to BT
A2DP, requiring a device reboot to restore normal audio.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Reflects the current reality: setCommunicationDevice API 31+, deferred
MODE_IN_COMMUNICATION, BT-mode Oboe (bt_active flag), per-arch builds,
Hangup call_id fix, and network monitoring integration.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Root cause: MainActivity set MODE_IN_COMMUNICATION at app launch,
hijacking system audio routing immediately — BT A2DP music dropped to
earpiece, and the pre-existing communication mode confused subsequent
setCommunicationDevice calls for BT SCO.
Fix: MainActivity now only sets volumes. MODE_IN_COMMUNICATION is set
via JNI right before Oboe audio_start() in CallEngine, and MODE_NORMAL
is restored after audio_stop() when the call ends.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Root cause: Oboe capture at 48kHz with InputPreset::VoiceCommunication
cannot open against a BT SCO device (only supports 8/16kHz). The stream
silently falls back to builtin mic, delivering zeros.
Fix: add bt_active flag to WzpOboeConfig. When set, capture skips
setSampleRate and setInputPreset, letting the system route to BT SCO
at its native rate. Oboe's SampleRateConversionQuality::Best resamples
to 48kHz for our ring buffers. Playout uses Usage::Media in BT mode.
New API: wzp_native_audio_start_bt() for BT mode, called from
set_bluetooth_sco(on=true). Normal audio_start() restores the
standard config when switching back to earpiece/speaker.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Two fixes for BT audio silence:
1. Switch Oboe streams from Exclusive to Shared sharing mode. Exclusive
mode bypasses Oboe's internal resampler, so opening a 48kHz stream
against a BT SCO device (8/16kHz only) fails at the AudioPolicy
level. Shared mode lets Oboe's resampler bridge the gap.
2. Add 500ms post-SCO delay before Oboe restart. The audio policy needs
time to apply the bt-sco route after setCommunicationDevice returns.
Without the delay, Oboe opens against the old device (handset).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
BT SCO devices only support 8kHz or 16kHz but our Oboe streams request
48kHz. Without resampling, AudioPolicyManager rejects the input stream
("getInputProfile could not find profile for... sampling rate 48000").
Fix: add setSampleRateConversionQuality(Best) to both capture and
playout stream builders. Oboe resamples internally so our ring buffers
stay at 48kHz regardless of the hardware sample rate.
Also removes the broken setBluetoothScoOn/isBluetoothScoOn calls from
stop_bluetooth_sco — just call stopBluetoothSco() unconditionally.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
