diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index ab0b4f5b32..de9096ee6f 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -174,6 +174,7 @@ rtc_static_library("audio_device") {
       }
       if (is_ios) {
         public_deps = [
+          "../../base:gtest_prod",
           "../../base:rtc_base",
           "../../sdk:rtc_sdk_common_objc",
         ]
@@ -298,7 +299,7 @@ if (rtc_include_tests) {
     if (is_ios) {
       sources += [ "ios/objc/RTCAudioSessionTest.mm" ]
       if (target_cpu != "x64") {
-        sources += [ "ios/audio_device_unittest_ios.cc" ]
+        sources += [ "ios/audio_device_unittest_ios.mm" ]
       }
       deps += [ "//third_party/ocmock" ]
     }
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.h b/webrtc/modules/audio_device/ios/audio_device_ios.h
index 01dd7cd930..5710299bdb 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "WebRTC/RTCMacros.h"
+#include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
@@ -292,6 +293,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
 
   // Set to true if we've activated the audio session.
   bool has_configured_session_;
+
+  // Exposes private members for testing purposes only.
+  FRIEND_TEST_ALL_PREFIXES(AudioDeviceTest, testInterruptedAudioSession);
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index ba4fe2a963..8cb1cd2b99 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -797,6 +797,7 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
   RTCAudioSession* session = [RTCAudioSession sharedInstance];
   // Subscribe to audio session events.
   [session pushDelegate:audio_session_observer_];
+  is_interrupted_ = session.isInterrupted ? true : false;
 
   // Lock the session to make configuration changes.
   [session lockForConfiguration];
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
similarity index 96%
rename from webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
rename to webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
index 1b7ff5780f..82f4426482 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
@@ -32,6 +32,9 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
+#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
+
 using std::cout;
 using std::endl;
 using ::testing::_;
@@ -822,4 +825,36 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   latency_audio_stream->PrintResults();
 }
 
+TEST_F(AudioDeviceTest, testInterruptedAudioSession) {
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  std::unique_ptr<webrtc::AudioDeviceIOS> audio_device;
+  audio_device.reset(new webrtc::AudioDeviceIOS());
+  std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
+  audio_buffer.reset(new webrtc::AudioDeviceBuffer());
+  audio_device->AttachAudioBuffer(audio_buffer.get());
+  audio_device->Init();
+  audio_device->InitPlayout();
+  // Force interruption.
+  [session notifyDidBeginInterruption];
+
+  // Wait for notification to propagate.
+  rtc::MessageQueueManager::ProcessAllMessageQueues();
+  EXPECT_TRUE(audio_device->is_interrupted_);
+
+  // Force it for testing.
+  audio_device->playing_ = false;
+  audio_device->ShutdownPlayOrRecord();
+  // Force it for testing.
+  audio_device->audio_is_initialized_ = false;
+
+  [session notifyDidEndInterruptionWithShouldResumeSession:YES];
+  // Wait for notification to propagate.
+  rtc::MessageQueueManager::ProcessAllMessageQueues();
+  EXPECT_TRUE(audio_device->is_interrupted_);
+
+  audio_device->Init();
+  audio_device->InitPlayout();
+  EXPECT_FALSE(audio_device->is_interrupted_);
+}
+
 }  // namespace webrtc