The broadcast alone wasn't reaching the first client because its
recv loop hadn't started yet when the second client registered.
Now the relay sends PresenceList directly to the new client (right
after RegisterPresenceAck) AND broadcasts to all others.
This guarantees every client gets the full user list:
- New client: via direct send (queued before recv loop starts)
- Existing clients: via broadcast
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
New signal infrastructure for the lobby-first UI:
- PresenceUser struct: { fingerprint, alias }
- SignalMessage::PresenceList: relay broadcasts full user list
to all signal clients on every register/deregister
- SignalHub::presence_list(): builds the list from connected clients
- SignalHub::broadcast(): sends to ALL signal clients
- Relay calls broadcast on register + unregister
- Desktop emits "presence_list" signal-event to JS frontend
This gives clients real-time visibility of who's online via the
signal channel, without needing to join a voice room first.
603 tests pass, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
New SignalMessage variants for P2P quality coordination:
UpgradeProposal/UpgradeResponse/UpgradeConfirm (#28):
- Consensual quality upgrade flow — proposer sends desired profile,
peer accepts/rejects based on own conditions, confirm commits both
- All carry call_id for relay routing
QualityCapability (#30):
- Peer reports its max sustainable profile — enables asymmetric
encoding where each side uses its own best quality instead of
forcing everyone to the weakest link
Relay forwards all 4 signals to the call peer (same pattern as
MediaPathReport, CandidateUpdate, HardNatProbe).
Desktop signal recv loop handles all 4 with debug logging.
Encoder switching TODOs noted for wiring into CallEngine.
4 new serde roundtrip tests. 603 total, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
When --key <64-char-hex> is provided with --replay, the analyzer
decrypts each packet's ChaCha20-Poly1305 payload using the session
key and logs plaintext frame sizes. Prints first 5 + every 100th
decrypt result, and a summary at the end.
This completes all 5 protocol analyzer tasks (#13-17):
- #13: Observer mode (live passive listener) — was done
- #14: TUI with Ratatui (per-participant panels) — was done
- #15: Capture and replay (.wzp format) — was done
- #16: HTML report (Chart.js loss/jitter graphs) — was done
- #17: Encrypted decode (--key for replay) — done now
Usage:
wzp-analyzer --replay session.wzp --key <64-hex-chars> --html report.html
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Birthday attack for random symmetric NATs:
- birthday.rs: open_acceptor_ports() opens N sockets, STUN-probes
each to learn external ports. generate_dialer_targets() builds
hit list (known ports first, then random fill). spray_dialer()
sprays QUIC connects with rate limiting, first success wins.
- Default: 32 acceptor ports, 128 dialer probes, 20ms interval
Signal coordination:
- HardNatBirthdayStart { acceptor_ports, external_ip } sent by
Acceptor when peer's HardNatProbe shows random/sequential NAT
- Relay forwards it like other call signals
- Desktop recv loop handles and logs it
Hybrid waterfall integration:
- On receiving HardNatProbe with non-cone allocation, Acceptor
auto-opens birthday ports and sends BirthdayStart
- Sockets kept alive 10s for NAT mapping persistence
- Dialer spray integration into race() pending (needs transport
hot-swap for background upgrade)
6 new tests, 599 total, 0 regressions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
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>
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>
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>
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>
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>
- 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>
- 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>
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>
Root cause: Hangup had no call_id field. The relay forwarded hangups to
ALL active calls for a user. When user A hung up call 1 and user B
immediately placed call 2, the relay's processing of A's hangup would
also kill call 2 (race window ~1-2s).
Fix: add optional call_id to Hangup (backwards-compatible via serde
skip_serializing_if). When present, the relay only ends the named call.
Old clients send call_id=None and get the legacy broadcast behavior.
Also: clear pending_path_report in Hangup recv handler and
internal_deregister to prevent stale oneshot channels from blocking
subsequent call setups.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Bluetooth: wire existing AudioRouteManager SCO support through both app
variants. Replace binary speaker toggle with 3-way route cycling
(Earpiece → Speaker → Bluetooth). Tauri side adds JNI bridge functions
(start/stop/query SCO, device availability) and Oboe stream restart.
Network awareness: integrate Android ConnectivityManager to detect
WiFi/cellular transitions and feed them to AdaptiveQualityController
via lock-free AtomicU8 signaling. Enables proactive quality downgrade
and FEC boost on network handoffs.
Build: add --arch flag to build-tauri-android.sh supporting arm64,
armv7, or all (separate per-arch APKs for smaller tester binaries).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
PRD 4: Disable IPv6 direct dial/accept temporarily. IPv6 QUIC
handshakes succeed but connections die immediately on datagram
send ("connection lost"). IPv4 candidates work reliably. IPv6
candidates still gathered but filtered at dial time.
PRD 1: Close losing transport after Phase 6 negotiation. The
non-selected transport now gets an explicit QUIC close frame
instead of silently dropping after 30s idle timeout. Prevents
phantom connections from polluting future accept() calls.
PRD 2: Harden accept loop with max 3 stale retries. Stale
connections are explicitly closed (conn.close) and counted.
After 3 stale connections, the accept loop aborts instead of
spinning until the race timeout.
PRD 3: Resource cleanup — close old IPv6 endpoint before
creating a new one in place_call/answer_call. Add Drop impl
to CallEngine so tasks are signalled to stop on ungraceful
shutdown.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The originating relay (where the caller is) never set peer_relay_fp
because the call was created locally. When the callee's answer
arrived via federation, the cross-relay dispatcher handled it but
didn't mark the call as cross-relay. This meant the caller's
MediaPathReport was delivered via local hub.send_to() to a peer
fingerprint that isn't connected locally — silently dropped.
Fix: in the cross-relay answer dispatcher, call
reg.set_peer_relay_fp(call_id, Some(origin_relay_fp)) so the
originating relay knows to forward MediaPathReport via federation.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Add relay_build field to RegisterPresenceAck so the client logs
which relay version it connected to. Shows in the debug log as
register_signal:ack_received {"relay_build":"f843a93"}.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
MediaPathReport was only delivered via local signal_hub, so calls
between peers on different relays always hit peer_report_timeout
and fell back to relay — even when direct P2P worked perfectly.
Fix: check peer_relay_fp in call_registry (same pattern as
DirectCallAnswer). If the peer is on a remote relay, wrap in
FederatedSignalForward and send via federation link. Also fix
the cross-relay dispatcher to deliver to BOTH caller and callee
(not just caller), since the report can come from either side.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The Acceptor's accept() on the shared signal endpoint can dequeue
a stale QUIC connection from a previous call that the Dialer has
already dropped. This results in "connection lost" errors when
media datagrams are sent — 100% drops on both sides.
Fix: after accepting a connection, check close_reason(). If the
connection is already closed, log a warning and re-accept. Also
verify max_datagram_size() is available before returning.
Additionally: emit transport details (remote addr, max_datagram,
close_reason) in the call_engine_starting debug event so stale
connection issues are visible in the user-facing debug log.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
When direct P2P calls show 100% datagram drops, we need to know
WHY send_media() fails. This commit adds:
- Remote address + stable_id logging on A-role accept and D-role
dial success (dual_path.rs) — tells us which candidate won
- Remote address + max_datagram_size on engine transport init —
verifies datagrams are negotiated
- last_send_err in send heartbeat — captures the actual error
from send_datagram() failures
- QuinnTransport::remote_address() helper
Also fixes UI badge: was looking for wrong event name
("dual_path_race_won" → "path_negotiated").
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
CLI binary was missing the new caller_build_version and
callee_build_version fields, causing E0063 compile errors on
Linux relay/client builds.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Add caller_build_version / callee_build_version (git short hash)
to DirectCallOffer and DirectCallAnswer so peers can identify each
other's build in debug logs. Also log own build at register time.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The CallSetup enum gained peer_direct_addr and peer_local_addrs
in Phase 5.5 but the wzp-android signal recv match arm was never
updated, breaking cargo ndk builds.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds a dedicated IPv6 QUIC endpoint (IPV6_V6ONLY=1 via socket2)
alongside the existing IPv4 signal endpoint for proper dual-stack
P2P connectivity. Previous [::]:0 dual-stack attempt broke IPv4
on Android; this uses separate sockets per address family like
WebRTC/libwebrtc.
- create_ipv6_endpoint(): socket2-based IPv6-only UDP socket,
tries same port as IPv4 signal EP, falls back to ephemeral
- local_host_candidates(v4_port, v6_port): now gathers IPv6
global-unicast (2000::/3) and unique-local (fc00::/7) addrs
- dual_path::race(): A-role accepts on both v4+v6 via select!,
D-role routes each candidate to matching-AF endpoint
- Graceful fallback: if IPv6 unavailable, .ok() → None → pure
IPv4 behavior identical to pre-Phase-7
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Two fixes:
## Revert [::]:0 dual-stack sockets → back to 0.0.0.0:0
Android's IPV6_V6ONLY=1 default on some kernels (confirmed on
Nothing Phone) makes [::]:0 IPv6-only, silently killing ALL
IPv4 traffic. This broke P2P direct calls: IPv4 LAN candidates
(172.16.81.x) couldn't complete QUIC handshakes through the
IPv6-only socket, causing local_direct_ok=false and relay
fallback on every call after the first.
Reverted all bind sites to 0.0.0.0:0 (reliable IPv4). IPv6 host
candidates are disabled in local_host_candidates() until a
proper dual-socket approach (one IPv4 + one IPv6 endpoint,
Phase 7) is implemented.
## Fix A (task #35): Oboe playout callback stall auto-restart
The Nothing Phone's Oboe playout callback fires once (cb#0) and
then stops draining the ring on ~50% of cold-launch calls. Fix
D+C (stop+prime from previous commit) didn't help because
audio_stop is a no-op on cold launch.
New approach: self-healing watchdog in audio_write_playout.
Tracks the playout ring's read_idx across writes. If read_idx
hasn't advanced in 50 consecutive writes (~1 second), the Oboe
playout callback has stopped:
1. Log "playout STALL detected"
2. Call wzp_oboe_stop() to tear down the stuck streams
3. Clear both ring buffers (prevent stale data reads)
4. Call wzp_oboe_start() to rebuild fresh streams
5. Log success/failure
6. Return 0 (caller retries on next frame)
This is the same teardown+rebuild that "rejoin" does — but
triggered automatically from the first stalled call instead of
requiring the user to hang up and redial. The watchdog runs
on every write so it fires within 1s of the stall starting.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Every QUIC endpoint was bound to 0.0.0.0:0 (IPv4-only). This
silently killed ALL IPv6 host candidates: the Dialer couldn't
send packets to [2a0d:...] addresses (wrong address family on
the socket), and the Acceptor couldn't receive incoming IPv6
QUIC handshakes. The IPv6 candidates were gathered and advertised
in DirectCallOffer/Answer but were completely non-functional.
On same-LAN with dual-stack (which both test phones have), this
meant:
- JoinSet fanned out 3+ candidates (2× IPv6 + 1× IPv4)
- IPv6 dials failed silently or timed out
- IPv4 dial worked but competed with failed IPv6 for JoinSet
attention
- Sometimes the JoinSet returned an IPv6 failure before the
IPv4 success, causing unnecessary fallback to relay
Fix: bind to [::]:0 (IPv6 any) instead of 0.0.0.0:0. On
dual-stack systems (Linux/Android default), [::]:0 creates a
socket that handles BOTH:
- IPv6 natively (global unicast, ULA)
- IPv4 via v4-mapped addresses (::ffff:172.16.81.x)
One socket, both protocols. All 7 bind sites updated:
- register_signal (signal endpoint)
- do_register_signal
- ping_relay
- probe_reflect_addr (fresh endpoint fallback)
- dual_path::race (A-role fresh, D-role fresh, relay fresh)
With this fix, same-LAN P2P should prefer the IPv6 path (no
NAT, direct routing, lower latency) and fall through to IPv4
if IPv6 fails — relay is the last resort after ALL candidates
are exhausted.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Before Phase 6, each side's dual-path race ran independently and
committed to whichever transport completed first. When one side
picked Direct and the other picked Relay, they sent media to
different places — TX > 0 RX: 0 on both, completely silent call.
Phase 6 adds a negotiation step: after the local race completes,
each side sends a MediaPathReport { call_id, direct_ok, winner }
to the peer through the relay. Both wait for the other's report
before committing a transport to the CallEngine. The decision
rule is simple: if BOTH report direct_ok = true, use direct; if
EITHER reports false, BOTH use relay.
## Wire protocol
New `SignalMessage::MediaPathReport { call_id, direct_ok,
race_winner }`. The relay forwards it to the call peer via the
same signal_hub routing used for DirectCallOffer/Answer. The
cross-relay dispatcher also forwards it.
## dual_path::race restructured
Returns `RaceResult` instead of `(Arc<QuinnTransport>, WinningPath)`:
- `direct_transport: Option<Arc<QuinnTransport>>`
- `relay_transport: Option<Arc<QuinnTransport>>`
- `local_winner: WinningPath`
Both paths are run as spawned tasks. After the first completes,
a 1s grace period lets the loser also finish. The connect
command gets BOTH transports (when available) and picks the
right one based on the negotiation outcome. The unused transport
is dropped.
## connect command flow (revised)
1. Run race() → RaceResult with both transports
2. Send MediaPathReport to relay with our direct_ok
3. Install oneshot; wait for peer's report (3s timeout)
4. Decision: both direct_ok → use direct; else → use relay
5. Start CallEngine with the agreed transport
If the peer never responds (old build, timeout), falls back to
relay — backward compatible.
## Relay forwarding
MediaPathReport is forwarded like DirectCallOffer/Answer: via
signal_hub.send_to(peer_fp) for same-relay calls, and via
cross-relay dispatcher for federated calls.
## Debug log events
- `connect:dual_path_race_done` — local race result
- `connect:path_report_sent` — our report to the peer
- `connect:peer_report_received` — peer's report
- `connect:peer_report_timeout` — peer didn't respond (3s)
- `connect:path_negotiated` — final agreed path with reasons
Full workspace test: 423 passing (no regressions).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Added warn-level log in handle_datagram when a federation
datagram arrives but no matching local room is found. Prints:
- room_hash (8-byte tag from the datagram)
- active_rooms (all rooms the relay currently has)
- seq + peer label
This diagnoses the cross-relay recv_fr=0 issue: if media IS
arriving from the peer relay but the room hash doesn't match any
active room, the log tells us exactly what hash is expected vs
what rooms exist locally. If no datagram log fires at all, the
issue is upstream (peer relay not forwarding, federation link
down, etc.).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
All rooms with names starting with 'call-' are now treated as
global rooms by the federation pipeline. This enables relay-
mediated media fallback for cross-relay direct calls: when Alice
on Relay A and Bob on Relay B both join the same call-<id> room,
the federation media forwarding pipeline (GlobalRoomActive
announcements + datagram forwarding + presence replication)
kicks in automatically without any runtime registration step.
Previously, cross-relay direct calls that couldn't go P2P
(symmetric NAT on either side) failed with "no media path"
because the call-<id> room wasn't in the configured global_rooms
set and media datagrams weren't forwarded across the federation
link.
The relay's existing ACL for call-* rooms (only the two
authorized fingerprints from the call registry can join)
prevents random clients from creating or eavesdropping on
call rooms.
## Changes
### `is_global_room` (federation.rs)
Added `room.starts_with("call-")` check before the static
global_rooms set lookup. Returns true immediately for any
call-prefixed room.
### `resolve_global_room` (federation.rs)
Return type changed from `Option<&str>` to `Option<String>`
(owned) because call-* room names aren't stored on `self` —
they come from the caller and resolve to themselves as the
canonical name. The 13 callers continue to work via String/&str
auto-deref; 4 HashMap lookups needed explicit `.as_str()` or
`&` borrows.
Full workspace test: 423 passing (no regressions).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Three fixes from a field-test log where same-LAN calls were
still losing the dual-path race to the relay path, peers were
getting stuck on an empty call screen when the other side
hung up, and 1-way audio was hard to diagnose because the
GUI debug log had no media-level events.
## 1. Direct-path 500ms head start (dual_path.rs)
The race was resolving in ~105ms with Relay winning even when
both phones were on the same MikroTik LAN with valid IPv6 host
candidates. Root cause: the relay dial is a plain outbound QUIC
connect that completes in whatever the client→relay RTT is
(~100ms), while the direct path needs the PEER to also process
its CallSetup, spin up its own race, and complete at least one
LAN dial back to us. That cross-client sequence reliably takes
longer than 100ms, so relay always won.
Fix: delay the relay_fut with `tokio::time::sleep(500ms)` before
starting its connect. Same-LAN direct dials complete in 30-50ms
typically, so the head start gives direct plenty of time to win
cleanly. Users on setups where direct genuinely can't work
(LTE-to-LTE cross-carrier) pay 500ms extra on the relay fallback,
which is invisible for a call setup.
## 2. Hangup propagation via a new hangup_call command (lib.rs + main.ts)
The hangup button was calling `disconnect` which stopped the
local media engine but never sent a SignalMessage::Hangup to
the relay. The peer never got notified and was stuck on the
call screen with silent audio. My earlier fix (commit e75b045)
only handled the RECEIVE side — auto-dismiss call screen on
recv:Hangup — but the SEND side was still missing.
New Tauri command `hangup_call`:
1. Acquire state.signal.lock(), send SignalMessage::Hangup
over the signal transport (best-effort; log + continue if
signal is down)
2. Acquire state.engine.lock(), stop the CallEngine
JS hangupBtn click handler now calls hangup_call with a fallback
to raw disconnect if the command is missing (older builds).
## 3. Media debug events (engine.rs + lib.rs)
Threaded tauri::AppHandle into CallEngine::start so the send/
recv tasks can emit call-debug events when the user has debug
logs enabled. Added on the Android branch (desktop branch
accepts the arg for API symmetry but doesn't emit yet):
- media:first_send — emitted when the first encoded frame is
handed to the transport. Useful for 1-way audio diagnosis:
if this fires on side A but side B never sees media:first_recv,
A's outbound is broken.
- media:first_recv — emitted when the first packet from the
peer arrives. Mirror of first_send.
- media:send_heartbeat — every 2s with frames_sent, last_rms,
last_pkt_bytes, short_reads, drops. A stalled last_rms
(== 0) tells you the mic isn't producing samples; a frozen
frames_sent tells you the encode pipeline hung.
- media:recv_heartbeat — every 2s with recv_fr, decoded_frames,
last_written, written_samples, decode_errs, codec. Mirror
invariants for the inbound direction.
All four are gated by `call_debug_logs_enabled()` via
`emit_call_debug`, so they only show up in the GUI log when the
user has the Call Flow Debug Logs checkbox on. Tracing::info!
still runs unconditionally so logcat (adb) keeps its copy
regardless.
The `emit_call_debug` fn in lib.rs is now `pub(crate)` so
engine.rs can call it via `crate::emit_call_debug`.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Same-LAN P2P was failing because MikroTik masquerade (like most
consumer NATs) doesn't support NAT hairpinning — the advertised
WAN reflex addr is unreachable from a peer on the same LAN as
the advertiser. Phase 5 got us Cone NAT classification and fixed
the measurement artifact, but same-LAN direct dials still had
nowhere to land.
Phase 5.5 adds ICE-style host candidates: each client enumerates
its LAN-local network interface addresses, includes them in the
DirectCallOffer/Answer alongside the reflex addr, and the
dual-path race fans out to ALL peer candidates in parallel.
Same-LAN peers find each other via their RFC1918 IPv4 + ULA /
global-unicast IPv6 addresses without touching the NAT at all.
Dual-stack IPv6 is in scope from the start — on modern ISPs
(including Starlink) the v6 path often works even when v4
hairpinning doesn't, because there's no NAT on the v6 side.
## Changes
### `wzp_client::reflect::local_host_candidates(port)` (new)
Enumerates network interfaces via `if-addrs` and returns
SocketAddrs paired with the caller's port. Filters:
- IPv4: RFC1918 (10/8, 172.16/12, 192.168/16) + CGNAT (100.64/10)
- IPv6: global unicast (2000::/3) + ULA (fc00::/7)
- Skipped: loopback, link-local (169.254, fe80::), public v4
(already covered by reflex-addr), unspecified
Safe from any thread, one `getifaddrs(3)` syscall.
### Wire protocol (wzp-proto/packet.rs)
Three new `#[serde(default, skip_serializing_if = "Vec::is_empty")]`
fields, backward-compat with pre-5.5 clients/relays by
construction:
- `DirectCallOffer.caller_local_addrs: Vec<String>`
- `DirectCallAnswer.callee_local_addrs: Vec<String>`
- `CallSetup.peer_local_addrs: Vec<String>`
### Call registry (wzp-relay/call_registry.rs)
`DirectCall` gains `caller_local_addrs` + `callee_local_addrs`
Vec<String> fields. New `set_caller_local_addrs` /
`set_callee_local_addrs` setters. Follow the same pattern as
the reflex addr fields.
### Relay cross-wiring (wzp-relay/main.rs)
Both the local-call and cross-relay-federation paths now track
the local_addrs through the registry and inject them into the
CallSetup's peer_local_addrs. Cross-wiring is identical to the
existing peer_direct_addr logic — each party's CallSetup
carries the OTHER party's LAN candidates.
### Client side (desktop/src-tauri/lib.rs)
- `place_call`: gathers local host candidates via
`local_host_candidates(signal_endpoint.local_addr().port())`
and includes them in `DirectCallOffer.caller_local_addrs`.
The port match is critical — it's the Phase 5 shared signal
socket, so incoming dials to these addrs land on the same
endpoint that's already listening.
- `answer_call`: same, AcceptTrusted only (privacy mode keeps
LAN addrs hidden too, for consistency with the reflex addr).
- `connect` Tauri command: new `peer_local_addrs: Vec<String>`
arg. Builds a `PeerCandidates` bundle and passes it to the
dual-path race.
- Recv loop's CallSetup handler: destructures + forwards the
new field to JS via the signal-event payload.
### `dual_path::race` (wzp-client/dual_path.rs)
Signature change: takes `PeerCandidates` (reflex + local Vec)
instead of a single SocketAddr. The D-role branch now fans out
N parallel dials via `tokio::task::JoinSet` — one per candidate
— and the first successful dial wins (losers are aborted
immediately via `set.abort_all()`). Only when ALL candidates
have failed do we return Err; individual candidate failures are
just traced at debug level and the race waits for the others.
LAN host candidates are tried BEFORE the reflex addr in
`PeerCandidates::dial_order()` — they're faster when they work,
and the reflex addr is the fallback for the not-on-same-LAN
case.
### JS side (desktop/main.ts)
`connect` invoke now passes `peerLocalAddrs: data.peer_local_addrs ?? []`
alongside the existing `peerDirectAddr`.
### Tests
All existing test callsites updated for the new Vec<String>
fields (defaults to Vec::new() in tests — they don't exercise
the multi-candidate path). `dual_path.rs` integration tests
wrap the single `dead_peer` / `acceptor_listen_addr` in a
`PeerCandidates { reflexive: Some(_), local: Vec::new() }`.
Full workspace test: 423 passing (same as before 5.5).
## Expected behavior on the reporter's setup
Two phones behind MikroTik, both on the same LAN:
place_call:host_candidates {"local_addrs": ["192.168.88.21:XXX", "2001:...:YY:XXX"]}
recv:DirectCallAnswer {"callee_local_addrs": ["192.168.88.22:ZZZ", "2001:...:WW:ZZZ"]}
recv:CallSetup {"peer_direct_addr":"150.228.49.65:NN",
"peer_local_addrs":["192.168.88.22:ZZZ","2001:...:WW:ZZZ"]}
connect:dual_path_race_start {"peer_reflex":"...","peer_local":[...]}
dual_path: direct dial succeeded on candidate 0 ← LAN v4 wins
connect:dual_path_race_won {"path":"Direct"}
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Before Phase 5 WarzonePhone used THREE separate UDP sockets per
client:
1. Signal endpoint (register_signal, client-only)
2. Reflect probe endpoints (one fresh socket per relay probe)
3. Dual-path race endpoint (fresh per call setup)
This broke two things in production on port-preserving NATs
(MikroTik masquerade, most consumer routers):
a. Phase 2 NAT detection was WRONG. Each probe used a fresh
internal port, so MikroTik mapped each one to a different
external port, and the classifier saw "different port per
relay" and labeled it SymmetricPort. The real NAT was
cone-like but measurement via fresh sockets hid that.
b. Phase 3.5 dual-path P2P race was BROKEN. The reflex addr
we advertised in DirectCallOffer was observed by the signal
endpoint's socket. The actual dual-path race listened on a
DIFFERENT fresh socket, on a different internal (and
therefore external) port. Peers dialed the advertised addr
and hit MikroTik's mapping for the signal socket, which
forwarded to the signal endpoint — a client-only endpoint
that doesn't accept incoming connections. Direct path
silently failed, relay always won the race.
Nebula-style fix: one socket for everything. The signal endpoint
is now dual-purpose (client + server_config), and both the
reflect probes and the dual-path race reuse it instead of
creating fresh ones. MikroTik's port-preservation then gives us
a stable external port across all flows → classifier correctly
sees Cone NAT → advertised reflex addr is the actual listening
port → direct dials from peers land on the right socket →
`endpoint.accept()` in the A-role branch of the dual-path race
picks up the incoming connection.
## Changes
### `register_signal` (desktop/src-tauri/src/lib.rs)
- Endpoint now created with `Some(server_config())` instead of
`None`. The socket can now accept incoming QUIC connections as
well as dial outbound.
- Every code path that previously read `sig.endpoint` for the
relay-dial reuse benefits automatically — same socket is now
ALSO listening for peer dials.
### `probe_reflect_addr` (wzp-client/src/reflect.rs)
- New `existing_endpoint: Option<Endpoint>` arg. `Some` reuses
the caller's socket (production: pass the signal endpoint).
`None` creates a fresh one (tests + pre-registration).
- Removed the `drop(endpoint)` at the end — was correct for
fresh endpoints (explicit early socket close) but incorrect
for shared ones. End-of-scope drop does the right thing in
both cases via Arc semantics.
### `detect_nat_type` (wzp-client/src/reflect.rs)
- New `shared_endpoint: Option<Endpoint>` arg, forwarded to
every probe in the JoinSet fan-out. One shared socket means
the classifier sees the true NAT type.
### `detect_nat_type` Tauri command (desktop/src-tauri/src/lib.rs)
- Reads `state.signal.endpoint` and passes it as the shared
endpoint. Falls back to None when not registered. NAT detection
now produces accurate classifications against MikroTik / most
consumer NATs.
### `dual_path::race` (wzp-client/src/dual_path.rs)
- New `shared_endpoint: Option<Endpoint>` arg.
- A-role: when `Some`, reuses it for `accept()`. This is the
critical change — the reflex addr advertised to peers is now
the address listening for incoming direct dials.
- D-role: when `Some`, reuses it for the outbound direct dial.
MikroTik keeps the same external port for the dial as for
the signal flow → direct dial through a cone-mapped NAT.
- Relay path: also reuses the shared endpoint so MikroTik has
a single consistent mapping across the whole call (saves one
extra external port and makes firewall traces cleaner).
- When `None`, falls back to fresh per-role endpoints as before.
### `connect` Tauri command (desktop/src-tauri/src/lib.rs)
- Reads `state.signal.endpoint` once when acquiring own reflex
addr and passes it through to `dual_path::race`.
### Tests
- `wzp-client/tests/dual_path.rs` and
`wzp-relay/tests/multi_reflect.rs` updated to pass `None` for
the new endpoint arg — tests use fresh sockets and that's
fine because the loopback harness doesn't care about
port-preserving NAT behavior.
Full workspace test: 423 passing (no regressions).
## Expected behavior after this commit on real hardware
Behind MikroTik + Starlink-bypass (the reporter's setup):
- Phase 2 NAT detect → **Cone NAT** (was SymmetricPort — false
positive from the measurement artifact)
- Phase 3.5 direct-P2P dial → succeeds for both cone-cone and
cone-CGNAT cases where the remote side was previously blocked
by our own socket mismatch
- LTE ↔ LTE cross-carrier → still likely relay fallback; that's
genuinely strict symmetric and needs Phase 5.5 port prediction.
## Phase 5.5 (next, separate PRD)
Multi-candidate port prediction + ICE-style candidate aggregation
for truly strict symmetric NATs. Not needed for the 95% case —
Phase 5 alone fixes most consumer-router setups.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Real-world report: a user with one LAN relay + one internet relay
got "Multiple IPs — treating as symmetric" because the LAN relay
saw the client's LAN IP (172.16.81.172) while the internet relay
saw the WAN IP (150.228.49.65). Two observations of "different
public IPs" from the classifier's perspective, but semantically
they describe two different network paths and shouldn't be
compared.
The LAN relay's reflection is always true, just not useful for
public NAT classification: there's no NAT between the client and
the LAN relay, so that path's reflex addr is always the LAN
interface IP regardless of what the public-facing NAT beyond it
looks like.
Fix: new `is_private_or_loopback` helper filters the probe set
before classification. Drops:
- 127.0.0.0/8 loopback
- 10/8, 172.16/12, 192.168/16 RFC1918 private
- 169.254/16 link-local
- 100.64/10 CGNAT shared-transition (same reasoning: a relay
that sees the client with a CGNAT addr is on the same carrier
network and can't describe public NAT state)
- IPv6 loopback, unspecified, fe80::/10 link-local
Failed probes still filtered out of classification (they were
already) but now dimmed in the UI list instead of highlighted
amber. Same rationale: a momentarily-offline probe target isn't
a warning-worthy state, it's just a fact about the probe run.
UI palette rebalance: only Cone gets green, everything else
neutral text-dim. Wording changed from warning-tone
"⚠ must use relay" to informational "ℹ P2P falls back to relay,
calls still work" — symmetric NAT isn't broken state, it just
means media takes the relay path.
Tests added (4 new in wzp_client::reflect):
- classify_drops_private_ip_probes — LAN + public → Unknown
- classify_drops_loopback_probes — loopback + 2 public → Cone
- classify_drops_cgnat_probes — CGNAT + 2 public same-IP-
diff-port → SymmetricPort
- classify_two_lan_probes_is_unknown_not_cone — all LAN → Unknown
Existing multi_reflect integration test updated: two loopback
relays now correctly classify as Unknown (because loopback reflex
addrs are filtered) with the plumbing-works invariant preserved.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
