diff --git a/web/zebra-spaces.html b/web/zebra-spaces.html
index f9cad31..3a6389a 100644
--- a/web/zebra-spaces.html
+++ b/web/zebra-spaces.html
@@ -4050,23 +4050,47 @@ const streamAudio = new Map();       // uuid -> <audio> pulling stream
 function streamUrlFor(pubHex){
   return SFU_BASE + '/stream?room=' + encodeURIComponent(roomID) + '&pub=' + pubHex;
 }
-function startStream(uuid, pubHex){
+async function startStream(uuid, pubHex){
   /* DON'T mute the WebRTC audio yet — if the fresh <audio>'s autoplay
    * is blocked (common on mobile after the entry gesture's grace has
    * passed) we'd be left with zero audio. Mute only after 'playing'
    * fires so the WebRTC path stays as the live fallback. */
+  const wantUrl = streamUrlFor(pubHex);
   let a = streamAudio.get(uuid);
+  /* Idempotent: a second toggle-on for the same pubHex while we're
+   * already loading/playing is a no-op. The previous behaviour reset
+   * .src which aborted the in-flight load — Firefox surfaces this as
+   * 'fetching process aborted at user's request' on the play()
+   * promise, which we then mistakenly logged as autoplay-blocked. */
+  if (a && a.src === wantUrl && !a.error) {
+    return;
+  }
   if (!a){
     a = document.createElement('audio');
     a.autoplay = true; a.controls = false;
+    /* preload=auto: encourage the browser to start buffering immediately
+     *   instead of waiting for play() — helps slow mobile links fill
+     *   the initial buffer before the user-gesture window expires.
+     * playsInline: tell Safari/iOS not to escalate audio playback into
+     *   a fullscreen player. No effect on Firefox/Chrome but cheap. */
+    a.preload = 'auto'; a.playsInline = true;
     document.body.appendChild(a);
     streamAudio.set(uuid, a);
-    applySinkTo(a);
+    /* setSinkId is async and can re-init the media pipeline. If we
+     * call it AFTER setting src + play(), the re-init aborts the load
+     * with the abort error fox saw. Await it FIRST, before any src
+     * assignment. Safe to await on browsers that don't support it —
+     * applySinkTo early-returns. */
+    await applySinkTo(a);
   }
   a.muted = false;
-  a.src = streamUrlFor(pubHex);
+  a.src = wantUrl;
   /* mute WebRTC the moment the HTTP stream actually plays a frame,
-   * not before. Removed on stop / error so WebRTC takes over again. */
+   * not before. Removed on error / autoplay reject so WebRTC takes
+   * over again. NOTE: 'stalled' is NOT treated as a failure — it
+   * fires constantly during normal startup on mobile cellular and
+   * each fire was causing the audio path to flip-flop. Real failure
+   * shows up as 'error' or as a rejected play() promise. */
   const muteWebRtcOnPlay = () => {
     const w = remoteAudio.get(uuid);
     if (w) try { w.muted = true; } catch(_){}
@@ -4078,7 +4102,6 @@ function startStream(uuid, pubHex){
   };
   a.addEventListener('playing', muteWebRtcOnPlay, { once: true });
   a.addEventListener('error', () => unmuteWebRtcOnFail('error'), { once: true });
-  a.addEventListener('stalled', () => unmuteWebRtcOnFail('stalled'));
   try {
     const p = a.play();
     if (p && p.then){
@@ -4756,9 +4779,9 @@ logLine('', 'ready — pick a handle, type a rendezvous code, enter the space');
 </script>
 
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