diff --git a/zebra-report/zebra-spaces.html b/zebra-report/zebra-spaces.html
index 95ad5f1..b185773 100644
--- a/zebra-report/zebra-spaces.html
+++ b/zebra-report/zebra-spaces.html
@@ -1640,6 +1640,7 @@ const sfuStreamsByPubHex = new Map();   // pubHex -> MediaStream
 
 function attachSfuTrack(uuid, stream){
   let a = remoteAudio.get(uuid);
+  const fresh = !a;
   if (!a){
     /* lease from the pre-blessed audio pool so Firefox Android's
      * per-element autoplay grant carries over from the entry-button
@@ -1650,12 +1651,26 @@ function attachSfuTrack(uuid, stream){
     a = leaseAudioElement();
     remoteAudio.set(uuid, a);
     applySinkTo(a);
+    /* one-shot breadcrumbs so we can see whether the new path lights
+     * up at all, vs. the silent-but-state=connected pattern. */
+    a.addEventListener('playing',  () => logLine('', 'rtc playing  '+uuid.slice(0,4)+' ct='+a.currentTime.toFixed(2)), { once: true });
+    a.addEventListener('pause',    () => logLine('err','rtc paused   '+uuid.slice(0,4)+' ct='+a.currentTime.toFixed(2)));
+    a.addEventListener('ended',    () => logLine('err','rtc ended    '+uuid.slice(0,4)));
+    a.addEventListener('stalled',  () => logLine('err','rtc stalled  '+uuid.slice(0,4)));
+    a.addEventListener('error',    () => logLine('err','rtc error    '+uuid.slice(0,4)+' code='+(a.error?a.error.code:'?')));
   }
   a.srcObject = stream;
+  const tracks = stream && stream.getAudioTracks ? stream.getAudioTracks() : [];
+  const t0 = tracks[0];
+  logLine('', 'sfu attach '+uuid.slice(0,4)+' fresh='+(fresh?1:0)+
+              ' tracks='+tracks.length+
+              (t0 ? ' tr0={en='+t0.enabled+' mu='+t0.muted+' rs='+t0.readyState+'}' : ''));
   try {
     const p = a.play();
-    if (p && p.catch) p.catch(e => logLine('err','rtc autoplay '+uuid+': '+e.message));
-  } catch(_){}
+    if (p && p.catch) p.catch(e => logLine('err','rtc autoplay '+uuid.slice(0,4)+': '+e.message));
+  } catch(e){
+    logLine('err','rtc play threw '+uuid.slice(0,4)+': '+e.message);
+  }
   stopMeter(uuid); startMeter(uuid, stream);
   logLine('', 'sfu: receiving '+((members.get(uuid)||{}).handle || uuid));
 }
@@ -3355,6 +3370,88 @@ function stopAlivePings(){
   clearInterval(aliveTimer); aliveTimer = null;
 }
 
+/* Telemetry tick — fired every TELEMETRY_INTERVAL_MS while in a room.
+ * Dumps PC states + receiver RTP counters + <audio> element state to
+ * the page log so we can see in real time whether a "silent" listener
+ * is starved (no RTP arriving), decoded-but-not-playing (RTP fine but
+ * <audio> paused/muted), or DJ-mode-stalled (streamAudio not loading).
+ *
+ * One line per tick per device gives a timeline that we can scroll
+ * back through after a regression and pinpoint exactly when the chain
+ * broke. Verbose by design — fox: "way more telemetry NOW".
+ *
+ * Numbers we lean on:
+ *   - sub.aud.pkt        : packetsReceived on sub PC's audio receiver
+ *   - sub.aud.bytes      : bytesReceived on the same
+ *   - sub.aud.jitter     : current jitter in seconds
+ *   - sub.aud.level      : audioLevel (0..1, 0 = silence)
+ *   - rtc[id]/stream[id] : <audio> element diagnostics
+ *     - ct = currentTime (advancing = audio playing)
+ *     - rs = readyState (4 = enough data)
+ *     - ns = networkState (1=idle 2=loading 3=stalled)
+ *     - pa = paused, mu = muted, vol = volume, err = MediaError code
+ */
+const TELEMETRY_INTERVAL_MS = 5000;
+let telemetryTimer = null;
+async function dumpTelemetry(){
+  const parts = [];
+  parts.push('role=' + myRole);
+  parts.push('sub=' + (sfuSubPC ? sfuSubPC.connectionState + '/' + sfuSubPC.iceConnectionState : 'none'));
+  parts.push('pub=' + (sfuPubPC ? sfuPubPC.connectionState + '/' + sfuPubPC.iceConnectionState : 'none'));
+  parts.push('mesh=' + peers.size);
+  parts.push('sListen=' + (selfListenerMode ? '1' : '0'));
+  parts.push('streamMode=' + streamMode.size);
+  parts.push('muted=' + (muted ? '1' : '0'));
+  /* receiver stats from sfuSubPC (the main listener path) */
+  if (sfuSubPC && typeof sfuSubPC.getStats === 'function'){
+    try {
+      const stats = await sfuSubPC.getStats(null);
+      stats.forEach(r => {
+        if (r.type === 'inbound-rtp' && r.kind === 'audio'){
+          parts.push('aud.recv pkt=' + (r.packetsReceived|0) + ' lost=' + (r.packetsLost|0) +
+                     ' bytes=' + (r.bytesReceived|0) + ' jitter=' + (r.jitter || 0).toFixed(4) +
+                     ' level=' + (r.audioLevel || 0).toFixed(3));
+        }
+        if (r.type === 'inbound-rtp' && r.kind === 'video'){
+          parts.push('vid.recv pkt=' + (r.packetsReceived|0) + ' lost=' + (r.packetsLost|0) +
+                     ' frames=' + (r.framesDecoded|0));
+        }
+      });
+    } catch(e){ parts.push('stats.err=' + e.message); }
+  }
+  /* every <audio> element in the room — both the WebRTC remoteAudio
+   * pool (rtc[]) and the DJ HTTP-pull streamAudio (stream[]). */
+  let i = 0;
+  for (const [uuid, a] of remoteAudio){
+    parts.push('rtc[' + uuid.slice(0,4) + '] rs=' + a.readyState + ' ns=' + a.networkState +
+               ' pa=' + (a.paused?1:0) + ' mu=' + (a.muted?1:0) + ' ct=' + a.currentTime.toFixed(2) +
+               ' err=' + (a.error?a.error.code:'_'));
+    if (++i > 4) { parts.push('rtc…+' + (remoteAudio.size - i) + ' more'); break; }
+  }
+  i = 0;
+  for (const [uuid, a] of streamAudio){
+    parts.push('stream[' + uuid.slice(0,4) + '] rs=' + a.readyState + ' ns=' + a.networkState +
+               ' pa=' + (a.paused?1:0) + ' mu=' + (a.muted?1:0) + ' ct=' + a.currentTime.toFixed(2) +
+               ' src=' + (a.src ? (a.src.length>30 ? '…'+a.src.slice(-30) : a.src) : 'none') +
+               ' err=' + (a.error?a.error.code:'_'));
+    if (++i > 4) { parts.push('stream…+' + (streamAudio.size - i) + ' more'); break; }
+  }
+  logLine('', '· ' + parts.join(' '));
+}
+function startTelemetryLoop(){
+  if (telemetryTimer) return;
+  telemetryTimer = setInterval(() => {
+    dumpTelemetry().catch(e => logLine('err','telemetry: '+e.message));
+  }, TELEMETRY_INTERVAL_MS);
+  /* fire one immediately so the user sees state without waiting for
+   * the first interval to elapse. */
+  dumpTelemetry().catch(()=>{});
+}
+function stopTelemetryLoop(){
+  if (!telemetryTimer) return;
+  clearInterval(telemetryTimer); telemetryTimer = null;
+}
+
 function openSignal(){
   ws = new WebSocket(SIGNAL_URL + '?room=' + encodeURIComponent(roomID));
   ws.onopen = async () => {
@@ -3364,10 +3461,12 @@ function openSignal(){
     const sig = await signBytes(sigJoin(roomID, nonce, myKeys.pubB64, myHandle));
     send({ type:'join', pubkey: myKeys.pubB64, handle: myHandle, nonce, sig });
     startAlivePings();
+    startTelemetryLoop();
   };
   ws.onclose = () => {
     ws = null;
     stopAlivePings();
+    stopTelemetryLoop();
     if (wantConnected){
       setStatus('rendezvous dropped — reconnecting…');
       if (sigReconnect) clearTimeout(sigReconnect);
@@ -5051,8 +5150,8 @@ logLine('', 'ready — pick a handle, type a rendezvous code, enter the space');
 
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