diff --git a/web/zebra-spaces.html b/web/zebra-spaces.html
index 9fd2b84..2bebcb9 100644
--- a/web/zebra-spaces.html
+++ b/web/zebra-spaces.html
@@ -3921,10 +3921,13 @@ async function dumpTelemetry(){
     } catch(e){ parts.push('stats.err=' + e.message); }
   }
   /* publisher-side stats — what we are actually emitting on the wire.
-   * Catches publisher stalls (X11 compositor blocking the audio thread
-   * during a window wiggle is a real one). gap = now - lastPacketSent
-   * — a gap > frame-interval is a wire stall. rtt is the most recent
-   * round-trip from remote-inbound-rtp (the SFU's view of our sender). */
+   * One line per kind so screen-audio shows up independently of
+   * screen-video (earlier collapse hid scr.send.aud entirely under
+   * scr.send.vid). Critical for diagnosing whether Firefox
+   * getDisplayMedia couples its audio + video source producers under
+   * X11 contention. gap = now - lastPacketSent — a gap > frame-
+   * interval is a wire stall. rtt is the most recent round-trip from
+   * remote-inbound-rtp (the SFU's view of our sender). */
   const pubPCs = [
     ['mic.send',    sfuPubPC],
     ['cam.send',    sfuCameraPC],
@@ -3936,29 +3939,35 @@ async function dumpTelemetry(){
     try {
       const stats = await pc.getStats(null);
       const nowMs = Date.now();
-      let out = null, remoteIn = null;
+      /* keyed by ssrc so audio + video each get a slot; remote-inbound
+       * pairs back via .ssrc on remote-inbound-rtp → ssrc on outbound. */
+      const outsBySSRC = new Map();
+      const remoteInsBySSRC = new Map();
       stats.forEach(r => {
         if (r.type === 'outbound-rtp' && (r.kind === 'audio' || r.kind === 'video')){
-          if (!out || (r.packetsSent|0) > (out.packetsSent|0)) out = r;
+          outsBySSRC.set(r.ssrc, r);
         }
-        if (r.type === 'remote-inbound-rtp') {
-          if (!remoteIn || (r.packetsLost|0) > (remoteIn.packetsLost|0)) remoteIn = r;
+        if (r.type === 'remote-inbound-rtp' && typeof r.ssrc !== 'undefined') {
+          remoteInsBySSRC.set(r.ssrc, r);
         }
       });
-      if (!out) continue;
-      const gap = out.lastPacketSentTimestamp
-        ? ((nowMs - out.lastPacketSentTimestamp) / 1000).toFixed(1) + 's'
-        : '?';
-      let s = label + ' pkt=' + (out.packetsSent|0) +
-              ' bytes=' + (out.bytesSent|0) +
-              ' gap=' + gap;
-      if (out.kind === 'video') s += ' frames=' + (out.framesEncoded|0);
-      if (remoteIn) {
-        s += ' rtt=' + ((remoteIn.roundTripTime||0).toFixed(3)) +
-             ' rlost=' + (remoteIn.packetsLost|0) +
-             ' rjit=' + ((remoteIn.jitter||0).toFixed(4));
+      for (const out of outsBySSRC.values()){
+        const gap = out.lastPacketSentTimestamp
+          ? ((nowMs - out.lastPacketSentTimestamp) / 1000).toFixed(1) + 's'
+          : '?';
+        const kindLabel = out.kind === 'audio' ? '.aud' : '.vid';
+        let s = label + kindLabel + ' pkt=' + (out.packetsSent|0) +
+                ' bytes=' + (out.bytesSent|0) +
+                ' gap=' + gap;
+        if (out.kind === 'video') s += ' frames=' + (out.framesEncoded|0);
+        const remoteIn = remoteInsBySSRC.get(out.ssrc);
+        if (remoteIn) {
+          s += ' rtt=' + ((remoteIn.roundTripTime||0).toFixed(3)) +
+               ' rlost=' + (remoteIn.packetsLost|0) +
+               ' rjit=' + ((remoteIn.jitter||0).toFixed(4));
+        }
+        parts.push(s);
       }
-      parts.push(s);
     } catch(_){}
   }
   /* every <audio> element in the room — both the WebRTC remoteAudio
@@ -5941,8 +5950,8 @@ logLine('', 'ready — pick a handle, type a rendezvous code, enter the space');
 
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