diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 252ec9dfb2..42db281261 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -277,6 +277,7 @@ if (is_ios || is_mac) {
           "../rtc_base",
           "../rtc_base:checks",
           "../rtc_base/system:fallthrough",
+          "../system_wrappers:field_trial",
           "../system_wrappers:metrics",
         ]
 
diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm
index efbf463efb..5df7b8f4ba 100644
--- a/sdk/objc/native/src/audio/audio_device_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_ios.mm
@@ -26,6 +26,7 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
 #import "base/RTCLogging.h"
@@ -517,6 +518,19 @@ void AudioDeviceIOS::HandleInterruptionEnd() {
           "Updating audio unit state.",
          is_interrupted_);
   is_interrupted_ = false;
+  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-iOS-Holding")) {
+    // Work around an issue where audio does not restart properly after an interruption
+    // by restarting the audio unit when the interruption ends.
+    if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+      audio_unit_->Stop();
+      PrepareForNewStart();
+    }
+    if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+      audio_unit_->Uninitialize();
+    }
+    // Allocate new buffers given the potentially new stream format.
+    SetupAudioBuffersForActiveAudioSession();
+  }
   UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
 }