- Copy/Share log now includes HH:MM:SS timestamps
- callInProgress stays true until call resolves (setup or hangup),
preventing multiple taps from firing multiple place_call offers
- Block place_call when there's a pending incoming call
- leaveVoice clears all call state (callInProgress, pendingCallId)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Filter self from lobby list (double-check in renderLobbyUsers)
- Disable "Direct Call" button when tapping own user
- Debounce call button (callInProgress flag prevents double-tap)
- Block calling own fingerprint
- Stats line shows codec names + fps + audio level
The direct call to the other phone failing is likely because
both phones share the same reflexive addr:port on the same NAT,
making determine_role return None (equal addrs). This is an
existing edge case in reflect.rs — not a UI bug.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Buttons: use text labels (Mic/Spk/End) instead of emoji HTML
entities that rendered as raw text on Android WebView
- Stats: match Rust CallStatus fields (tx_codec, rx_codec,
encode_fps, recv_fps, audio_level, spk_muted)
- Nicknames: register_signal sends derive_alias() as the alias
so other users see "Brave Falcon" instead of "a525:e9b2:..."
- Lobby header shows alias from get_app_info instead of raw fp
- pollStatus uses correct field names from Rust struct
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Discord-style bottom drawer for voice instead of navigating away:
- "Join Voice" hides the FAB, slides up a persistent bottom bar
- Drawer shows: room name, timer, P2P/Relay badge, level meter
- Controls: mic, speaker, end call — all in the drawer
- Direct call info (identicon, name, P2P badge) shown inline
- Lobby stays visible above the drawer at all times
- Stats line shows codec/packet/FEC info
- Leave voice = drawer slides away, FAB returns
Removed: full-screen call-screen, back button, old participant
list, old mic/speaker/hangup buttons. All voice interaction
happens in the 15% bottom drawer while the lobby stays live.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Settings now shows relay list with:
- Visual list of all configured relays
- Active relay highlighted in green with "ACTIVE" badge
- Tap a relay to switch (deregisters + reconnects automatically)
- X button to remove a relay (keeps at least 1)
- Add relay with name + address inputs
- Reconnect flow: deregister → clear lobby → auto-connect to new relay
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
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>
The lobby now populates from PresenceList signal events:
- Relay broadcasts user list on register/deregister
- JS receives "presence_list" signal-event
- Updates lobbyUsers map (excluding self)
- Renders user rows with identicon, name, fingerprint
Users appear in the lobby as soon as they register their
signal channel — no need to join voice first.
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>
Complete JS rewrite for IRC-style lobby flow:
- Auto-connect signal channel on app launch (no connect button)
- Lobby shows online users with identicon, name, voice status
- "Join Voice" FAB toggles room voice on/off
- Tap user → context menu → Direct Call
- Incoming call banner slides up from bottom
- Back button returns from call to lobby
- Settings panel preserved with all debug toggles
~500 lines (down from 1786) — focused on the lobby experience.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
New IRC-style lobby layout:
- Auto-connect on launch, drop into user list
- User rows with identicon, name, fingerprint, voice status
- Speaking indicator (green highlight + pulsing)
- Join Voice FAB (green, toggles to Leave/red)
- Incoming call banner (slides up from bottom)
- User context menu (tap user → Call / Message)
- Settings panel preserved from original
The old connect-screen HTML is removed. The call-screen is kept
intact. JS adaptation next.
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>
New setting: "Birthday attack (opens extra ports for hard NAT)"
- Default: OFF — no extra latency on call setup
- When ON: waits up to 3s for peer's birthday ports if peer has
non-cone NAT, adds them to the dial race
Gated end-to-end: Settings → localStorage → JS invoke →
Rust connect param → birthday wait + target injection.
LAN/cone calls unaffected regardless of setting.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Complete Dialer-side birthday attack integration:
- SignalState stores peer_birthday_ports from HardNatBirthdayStart
- connect command: if peer's HardNatProbe shows non-cone NAT, waits
up to 3s for birthday ports to arrive (Acceptor needs time to open
32 sockets + STUN-probe each)
- When birthday ports arrive, generate_dialer_targets() builds hit
list (known ports + random fill) and adds them to PeerCandidates
- All birthday targets go into the dual-path race as extra candidates
- LAN/cone calls skip the wait entirely (gated on allocation type)
Full waterfall now:
1. Standard candidates (reflexive + mapped) → immediate
2. Port prediction (sequential delta) → immediate
3. Birthday targets (if non-cone peer) → +3s wait
4. All of above raced in parallel via JoinSet
5. Relay runs concurrently with 500ms head-start
599 tests pass, 0 regressions.
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>
New toggle in Settings → "Direct-only mode (no relay fallback)":
- Default: OFF (normal behavior, relay fallback on P2P failure)
- When ON: connect returns error if P2P fails, with full
candidate_diags in the debug log showing why each candidate
failed. Call never falls back to relay.
Useful for testing NAT traversal — you see the exact failure
reason instead of the call silently working through relay.
Wired end-to-end:
- Settings.directOnly persisted in localStorage
- Passed as directOnly param to Rust connect command
- connect:path_negotiated shows direct_only flag
- connect:direct_only_failed emits on failure with diags
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>
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>
