node-26#9. decoder_control.py's AIS mode now actually launches AIS-catcher
instead of 400ing: a background thread reads its stdout JSON stream
(one message per line) and keeps a live in-memory snapshot keyed by mmsi,
since AIS-catcher streams rather than writing a periodic file the way
dump1090's --write-json does. Messages are merged onto the existing entry
per mmsi rather than replacing it, since AIS-catcher emits static data
(name) and position reports (lat/lon) as separate message types — a naive
overwrite would blank the name back to null on every position-only update
(caught by a standalone unit check against a fake stdout stream before
this fix, not by pytest — this container has no test suite yet).
edge-node's telemetry_uplink_loop now posts non-empty vessel snapshots to
the new /telemetry/ais the same way it already does for aircraft.
UNVERIFIED against real hardware/binary in this session, same caveat as
the ADS-B commit — AIS-catcher's JSON field names are believed correct
from its docs, not confirmed against a real capture.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
node-26#9. New secondary-sdr-container claims the node's SECOND physical
SDR (RTL-SDR index 1 — op25 always claims index 0; no serial-based binding
yet, same gap op25 itself has). Its control API (start/stop/status/data,
mirroring op25_controller.py) launches dump1090 in adsb mode and exposes
the decoded aircraft.json snapshot; AIS mode 400s until it's wired next.
edge-node: on_config_push starts/stops it when secondary_sdr_mode changes,
lifespan resumes it after a restart if already configured, and a new
telemetry_uplink_loop polls its /secondary/data every 10s and POSTs
non-empty snapshots to C2's new /telemetry/adsb (same bearer-key pattern
call_recorder.py already uses for audio upload).
UNVERIFIED: this container has not been built or run against real hardware
in this session (sandboxed authoring machine, no docker) — dump1090's
--write-json field names are believed correct from its docs but not
confirmed against a real capture. Build + hardware smoke test before this
reaches a real node.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Adds a per-node secondary_sdr_mode config field (none|adsb|ais|op25_2),
applied the same way as hardware_preset/ppm_override so it survives system
reassignment. op25-container gets a GET /devices endpoint that counts
connected SDRs via lsusb; the edge-node checkin now reports sdr_count and
secondary_sdr_mode up to the server, the first node-initiated hardware
report (everything else was C2 pushing config down). Tracked as node-26#9.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
The single encode at save time was %mp3(bitrate=16), and the comment said why:
it matched what Liquidsoap pushes to Icecast. That was the wrong thing to
match. Icecast is the LISTENING path and 16 kbps is a bandwidth budget for a
live stream; this file is the ACCURACY path -- it is what Whisper transcribes,
and CLAUDE.md is explicit that everything downstream is hostage to it. P25 has
already been through a vocoder, so 16 kbps MP3 stacked a second lossy stage on
the one copy that had to stay faithful.
FLAC instead. Lossless, so the bytes Whisper receives are the bytes PulseAudio
captured. ~1.3 MB/min against 120 KB/min, which keeps a 600 s call (the time
cap) around 13 MB -- inside Whisper's 25 MB request cap and well inside
upload_max_bytes. Icecast's own 16 kbps stream is untouched; nothing about
live listening changes.
Capture, buffering, silence detection and the byte-offset trim are all
unchanged: they operate on raw PCM and never saw the encode. The sample rate
stays pinned to pcm.SAMPLE_RATE so the encode remains a straight pass -- the
trim arithmetic depends on that, and Whisper resamples to 16 kHz itself.
encode_mp3 is now encode_recording, the upload sends audio/flac, and the test
that pinned the old contract now pins losslessness instead, including an
assertion that no bitrate constant comes back.
This is the before/after boundary for STT quality. Last night's window is the
16 kbps baseline.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The source password had a `hackme` fallback in five places -- entrypoint.sh,
docker-compose.yml, setup.sh's prompt default, .env.example, edge-node's
config.py -- plus op25-container's os.getenv default and two README rows. Any
node whose operator pressed Enter through setup.sh is running a credential
that is written down in this repo.
That matters more than the usual default-password case because Icecast binds
all interfaces and the SOURCE password is write access: it does not just let a
LAN neighbour listen, it lets them PUSH audio into the stream the frontend and
mobile clients play as live radio. Injecting fake traffic into a public-safety
feed is the failure worth preventing.
Approach: remove every fallback rather than change them to a better default.
- icecast/entrypoint.sh refuses to start if either password is empty, and says
how to generate one. This is the single hard gate; everything else is
defence in depth behind it.
- docker-compose.yml uses ${VAR:?message} so a missing value stops the stack
at compose time with a readable error instead of becoming an empty string.
- setup.sh GENERATES a random password when the operator presses Enter, via
openssl rand -base64 24 with a /dev/urandom fallback. Pressing Enter now
gives you a random password rather than a known one, which is the actual
behaviour change -- a prompt default nobody types over is not a default, it
is the value.
- .env.example ships the keys empty with the generation command in a comment,
and README.md now marks both as required with no default.
Client suite: 185 passed.
Note this does NOT rotate anything already deployed. node-002's .env still has
whatever it was set up with; that is an operational step, tracked in the issue.
Closeslogan/node-26#3
Pure formatting, no behaviour change: strip trailing whitespace from blank
lines, give top-level defs in system_cacher.py their two blank lines, wrap
the long discord_radio.join signature, and split the duplicated
active_config ternary in main.py and routers/api.py across lines.
Verified clean with flake8 --max-line-length=120, matching CI.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Two unauthenticated surfaces closed on the edge node.
Dashboard and API: the local dashboard and every /api/* route were open to
anything on the node's LAN. Adds a login page plus session-cookie auth for
the browser, and cookie-or-Basic for the API so scripted callers stay
possible. Passwords are hashed with stdlib scrypt (no new dependency, this
runs on a Pi) and compared in constant time; the salt and session-signing
secret persist in credentials.json. Startup warns while the default password
is still in place. No non-browser callers of the node API exist today
(C2 talks to nodes over MQTT and nodes call C2 outbound), so nothing breaks.
Adds python-multipart, which FastAPI's Form() needs for the login POST and
which was missing from requirements entirely.
MQTT: nodes authenticated with a shared drb-node password, and the broker
ACL keyed off %c — the client-supplied client id — so any holder of that one
password could claim another node's topic namespace. Nodes now connect as
username=<node_id>, password=<their C2-issued api_key>, which mosquitto's
dynamic-security plugin checks, with the ACL keyed off the authenticated %u.
TLS is gated on MQTT_TLS and uses default CA verification.
The old key_request MQTT path stays in place behind TODO(mqtt-cutover)
markers as the fallback until the cutover is proven; a node with no api_key
on disk logs a clear repeated refusal rather than spinning.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The control channel was wrong in both directions. Grants fire 0.84-1.62s
before anyone speaks, and srcaddr can drop to 0 while someone is still
talking - one recording came back "-1.61s lead, -0.00s tail", the trim
finding nothing to remove because the window had closed on live speech.
Confirmed by ear: the cut lands at a word boundary on an unfinished word.
Audio is ground truth for WHEN. The console remains the only source of
WHO, so it still supplies talkgroup, alias and rid.
START voice onset in the captured audio, with a 0.25s pre-roll that
now covers only chunk quantisation and threshold ramp-up rather
than a variable control-channel offset.
STOP call_silence_timeout seconds of silence heard in the audio.
LABEL resolved AT CLOSE from a bounded rolling history of console
observations overlapping the window, +4s/-2s, because there is
no guaranteed ordering between a grant and its audio.
SPLIT a console talkgroup change still forces a cut, since two calls
with no silence between them would otherwise merge into one.
Capture now emits raw PCM instead of MP3. Silence detection becomes
integer arithmetic per chunk with no decode, trimming becomes a byte
offset slice rather than a second ffmpeg pass, and MP3 encoding happens
exactly once at save - uploads are no longer double-encoded.
Audio with no talkgroup anywhere in its window is discarded rather than
uploaded: an untagged call silently poisons incident correlation, which
is worse than losing the audio. Logged at ERROR and counted on
/api/status.
When capture produces no audio at all the old console state machine
still runs, so a node with a broken audio path keeps reporting radio
activity. That is now the only consumer of call_idle_timeout.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The tgid_change and tgid_change_unlogged paths closed the outgoing
segment with no tail pad, on the reasoning that the new grant's
timestamp is an exact, already-known boundary. That is exact only in
control-channel time. The buffered audio lags control timestamps by
~1.5s (measured 0.84-1.62s across seven field calls), so slicing there
cut roughly the outgoing call's last 1.5s of speech - recordings ending
mid-word with ~0s trailing silence.
Both paths now pad, and the recorder's bounded tail wait blocks until
that audio has actually been captured. TAIL_WAIT_TIMEOUT_SECONDS goes
2.0 -> 4.0 so it can satisfy the 3.0s pad instead of giving up and
warning on every talkgroup switch.
The incoming call's pre-roll is served from the ring buffer, so the
delay costs it nothing. The two slices overlapping in the underlying
audio is correct: the stream genuinely contains one call's tail and
then the next call's start.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The recording window is anchored to OP25 control-channel timestamps, but
the buffered audio lags those by roughly 1.5s. Trim logs across seven
calls measured the offset at 0.84-1.62s, consistently present.
At a 1.0s pad a short call closed its window before the voice arrived:
a 0.97s control-channel call closed at T+1.97 while voice started around
T+1.5, capturing ~0.4s of speech and cutting mid-word. Confirmed by a
0.57s file whose final 0.10s measured -12.2dB against its own -18.2dB
average - clipped speech, not a tail - and by two short calls that
logged no trim at all because no trailing silence remained.
Being generous is free here: trim_silence already strips trailing
silence back to the guard margin before upload, so long calls are
unaffected while short ones gain the window they need. Over-capture
costs nothing; under-capture loses words permanently.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The pulse_socket named volume survives container recreation, so after a
compose recreate the previous container's /run/pulse/pid and native
socket were still present. PulseAudio read the stale pid file, decided a
daemon was already running, and refused to start:
E: [pulseaudio] pid.c: Daemon already running.
The entrypoint still reported "PulseAudio socket ready" because it only
checked that the socket file existed - and a stale one did. Capture then
failed in a restart loop against a dead daemon.
Readiness in both the op25 entrypoint and drb-edge-node now means a
pactl probe actually succeeds. Stale pid/socket are removed only when
that probe fails, so a live daemon's socket is never deleted.
pulseaudio-utils was missing from the edge-node image (only libpulse0
was installed), so no pactl binary existed there at all - added.
Capture exits are now classified: a missing source logs at ERROR and
names the configured PULSE_SOURCE, rather than looking identical to
"daemon not up yet". Retrying forever against a wrong source name is how
the April PulseAudio failure stayed hidden.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Measured six recordings off a live P25 node and found two independent
defects causing lost audio at the end of calls:
- stop_recording() sliced the ring buffer immediately, so if the MP3
muxer had not yet delivered the tail the file was silently short.
Now waits (bounded, 2s) until buffered audio covers the end epoch.
- TGID-change closes used the new grant's epoch as the end with no pad
at all, guaranteeing truncation on every split. Tail pad is now a
setting, default raised 0.5s -> 1.0s.
The ring buffer also capped maximum call length: a call longer than the
buffer had its front silently clamped. The ring now serves the pre-roll
only, with a per-call accumulator for the rest, bounded at 4.8MB.
Clamping is loudly warned rather than silent.
Uploads averaged 63% silence, which inflates STT cost and is a known
Whisper hallucination trigger. Leading/trailing silence is now trimmed
conservatively (-40dB, 0.25s guard, internal pauses untouched).
started_at/ended_at still describe the call; new audio_* fields carry
the trimmed audio bounds so playback can map back to wall clock.
All-silence recordings are skipped and logged instead of uploaded.
Also: log measured control-channel idle on idle-timeout closes so
CALL_IDLE_TIMEOUT can be tuned from data, and quiet the httpx logger
which emitted ~170k lines/day of poll noise.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
- RadioBot.join() accepts system_name parameter
- Sets bot activity to ActivityType.listening → system name on connect
- Clears presence (activity=None) on leave
- main.py passes system_name from MQTT payload to radio_bot.join()
op25-container/system.pa New PulseAudio config: auth-anonymous=1 so the edge-node can connect without group membership
op25-container/Dockerfile Copies system.pa into /etc/pulse/system.pa
drb-edge-node/Dockerfile Added libpulse0 — ffmpeg needs this runtime library to use -f pulse
drb-edge-node/app/internal/call_recorder.py Switched recording from Icecast (burst-buffer overlap) to PulseAudio monitor (zero-delay, exact call boundaries)
docker-compose.yml: Added a pulse_socket named volume mounted at /run/pulse in both op25 and edge-node. Also set PULSE_SERVER=unix:/run/pulse/native in edge-node so libpulse (and ffmpeg's pulse input) finds the right socket.
discord_radio.py: Removed _icecast_url and changed _play_stream() to use -f pulse -i default.monitor. This reads directly from the PulseAudio sink monitor — zero buffer delay. The PULSE_SERVER env var is inherited by the ffmpeg subprocess.
Note: default.monitor captures whatever audio is playing on the default sink. If OP25 uses a named virtual sink, you may need to replace default.monitor with <sink_name>.monitor (run pactl list sinks short inside the op25 container to find the name).
Issue 2 — No audio URL / GCS credentials
storage.py: storage.Client() was using ADC but ADC isn't configured in the container. Now uses storage.Client.from_service_account_json(settings.gcp_credentials_path) when GCP_CREDENTIALS_PATH is set — same credential file Firebase already loads.
You also need to mount the key file into the server container in docker-compose.yml:
c2-core:
volumes:
- ./gcp-key.json:/app/gcp-key.json:ro
And set GCS_BUCKET=your-bucket-name in .env.
Issue 3 — Token orphaning
mqtt_manager.py: Every checkin now includes "discord_connected": radio_bot.is_connected.
mqtt_handler.py: On checkin, if discord_connected is explicitly False, calls release_token(node_id). Only fires on explicit false (missing field = unknown = no action).
node_sweeper.py: When a node is swept to offline, its token is released too. This covers the case where the node stops checking in entirely (crash/power loss).
Logan CusanoandClaude Sonnet 5