Fix tail truncation, decouple call length from buffer, trim silence

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>

drb-edge-node/app/main.py

@@ -1,5 +1,8 @@
 import asyncio
 from contextlib import asynccontextmanager
+from datetime import datetime, timezone
+from typing import Optional
+
 from fastapi import FastAPI
 from app.config import settings
 from app.models import SystemConfig
@@ -20,6 +23,13 @@ from app.routers import api, ui
 # Event handlers wired up at startup
 # ---------------------------------------------------------------------------
 
+def _iso(epoch: Optional[float]) -> Optional[str]:
+    """Epoch → UTC ISO-8601, matching metadata_watcher's timestamp format."""
+    if epoch is None:
+        return None
+    return datetime.fromtimestamp(epoch, timezone.utc).isoformat()
+
+
 async def on_call_start(data: dict):
     radio_bot.start_stream()
     await mqtt_manager.publish_status("recording")
@@ -35,20 +45,44 @@ async def on_call_start(data: dict):
 
 async def on_call_end(data: dict):
     radio_bot.stop_stream()
-    file_path = await call_recorder.stop_recording(end_epoch=data.get("ended_at_epoch"))
-    if file_path:
+    recording = await call_recorder.stop_recording(end_epoch=data.get("ended_at_epoch"))
+
+    if recording is not None and recording.path is not None:
+        # Silence trimming shortens the audio, so the audio's own bounds no
+        # longer equal the call's. `started_at`/`ended_at` keep meaning the CALL
+        # (what OP25 observed on the control channel) — these extra fields carry
+        # the AUDIO's wall-clock bounds so playback, correlation and incident
+        # timelines can still map an audio offset back to real time:
+        #     wall_clock_of(audio_offset_t) == audio_start_epoch + t
+        data["audio_start_at"] = _iso(recording.audio_start_epoch)
+        data["audio_end_at"] = _iso(recording.audio_end_epoch)
+        data["audio_start_epoch"] = recording.audio_start_epoch
+        data["audio_end_epoch"] = recording.audio_end_epoch
+        data["audio_lead_trimmed"] = round(recording.lead_trimmed, 3)
+        data["audio_tail_trimmed"] = round(recording.tail_trimmed, 3)
+        if recording.clamped_seconds:
+            data["audio_clamped_seconds"] = round(recording.clamped_seconds, 3)
+
+    if recording is not None and recording.path is not None:
         node_cfg = load_node_config()
         audio_url = await call_recorder.upload_recording(
-            file_path,
+            recording.path,
             data["call_id"],
             talkgroup_id=data.get("tgid"),
             talkgroup_name=data.get("tgid_name"),
             system_id=node_cfg.assigned_system_id,
+            audio_start_epoch=recording.audio_start_epoch,
+            audio_end_epoch=recording.audio_end_epoch,
         )
         if audio_url:
             data["audio_url"] = audio_url
         else:
             logger.error(f"Audio upload failed for call {data['call_id']}. Verify C2_URL and Node API Key.")
+    elif recording is not None and recording.all_silence:
+        # Explicit policy: an all-silence recording is not uploaded. It has no
+        # transcript value and silence is what makes Whisper invent text.
+        data["audio_skipped"] = "all_silence"
+        logger.warning(f"Call {data['call_id']} was pure silence — no upload. Investigate the audio path.")
     else:
         logger.warning(
             f"No recording file generated for call {data['call_id']} "