--- a/dom/media/webrtc/MediaEngineWebRTC.h
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@@ -358,16 +358,28 @@ private:
 /**
  * Representation of data tied to an AllocationHandle rather than to the source.
  */
 struct Allocation {
   Allocation() = delete;
   explicit Allocation(const RefPtr<AllocationHandle>& aHandle);
   ~Allocation();
 
+#ifdef DEBUG
+  /**
+   * We call this every time we append data to the track for this Allocation.
+   * It asserts that we only append once per iteration at most.
+   */
+  void RegisterLastAppendTime(MediaStreamGraphImpl* aGraph);
+
+  // The MSGImpl::IterationEnd() of the last time we appended data.
+  // Graph thread only.
+  GraphTime mLastAppendTime = 0;
+#endif
+
   const RefPtr<AllocationHandle> mHandle;
   RefPtr<SourceMediaStream> mStream;
   TrackID mTrackID = TRACK_NONE;
   PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
   bool mEnabled = false;
 };
 
 /**

--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -50,16 +50,29 @@ LogModule* AudioLogModule() {
 }
 
 Allocation::Allocation(const RefPtr<AllocationHandle>& aHandle)
   : mHandle(aHandle)
 {}
 
 Allocation::~Allocation() = default;
 
+#ifdef DEBUG
+void
+Allocation::RegisterLastAppendTime(
+    MediaStreamGraphImpl* aGraph)
+{
+  aGraph->AssertOnGraphThreadOrNotRunning();
+  MOZ_ASSERT((aGraph->IterationEnd() == 0 && mLastAppendTime == 0) ||
+             aGraph->IterationEnd() > mLastAppendTime,
+             "Iteration time didn't advance since last append");
+  mLastAppendTime = aGraph->IterationEnd();
+}
+#endif
+
 void
 WebRTCAudioDataListener::NotifyOutputData(MediaStreamGraph* aGraph,
                                           AudioDataValue* aBuffer,
                                           size_t aFrames,
                                           TrackRate aRate,
                                           uint32_t aChannels)
 {
   MutexAutoLock lock(mMutex);
@@ -878,16 +891,31 @@ MediaEngineWebRTCMicrophoneSource::Pull(
 {
   LOG_FRAMES(("NotifyPull, desired = %" PRId64, (int64_t) aDesiredTime));
 
   StreamTime delta = aDesiredTime - aStream->GetEndOfAppendedData(aTrackID);
   if (delta <= 0) {
     return;
   }
 
+  {
+    MutexAutoLock lock(mMutex);
+    size_t i = mAllocations.IndexOf(aHandle, 0, AllocationHandleComparator());
+    if (i == mAllocations.NoIndex) {
+      // This handle must have been deallocated. That's fine, and its track
+      // will already be ended. No need to do anything.
+      return;
+    }
+
+#ifdef DEBUG
+    mAllocations[i].RegisterLastAppendTime(
+      static_cast<MediaStreamGraphImpl*>(aStream->Graph()));
+#endif
+  }
+
   // Not enough data has been pushed so we fill it with silence.
   // This could be due to underruns or because we have been stopped.
 
   AudioSegment audio;
   audio.AppendNullData(delta);
   aStream->AppendToTrack(aTrackID, &audio);
 }
 
@@ -1153,16 +1181,20 @@ MediaEngineWebRTCMicrophoneSource::Inser
     }
   }
 
   for (Allocation& allocation : mAllocations) {
     if (!allocation.mStream) {
       continue;
     }
 
+#ifdef DEBUG
+    allocation.RegisterLastAppendTime(allocation.mStream->GraphImpl());
+#endif
+
     TimeStamp insertTime;
     // Make sure we include the stream and the track.
     // The 0:1 is a flag to note when we've done the final insert for a given input block.
     LogTime(AsyncLatencyLogger::AudioTrackInsertion,
             LATENCY_STREAM_ID(allocation.mStream.get(), allocation.mTrackID),
             (&allocation != &mAllocations.LastElement()) ? 0 : 1, insertTime);
 
     // Bug 971528 - Support stereo capture in gUM