diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 57e62f50fd..11a1c5815e 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -144,6 +144,7 @@ AudioReceiveStream::AudioReceiveStream(
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+  Stop();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 94bfabd74b..d30eb110ce 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -119,6 +119,7 @@ struct ConfigHelper {
               .After(expect_set);
           return channel_proxy_;
         }));
+    EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 358c142a1b..71803cbee3 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1424,6 +1424,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
       LOG(LS_INFO) << "Stopping playout for channel #" << channel();
       stream_->Stop();
     }
+    playout_ = playout;
   }
 
  private:
@@ -1444,6 +1445,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioReceiveStream(config_);
     RTC_CHECK(stream_);
+    SetPlayout(playout_);
   }
 
   rtc::ThreadChecker worker_thread_checker_;
@@ -1452,6 +1454,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
   // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
   // configuration changes.
   webrtc::AudioReceiveStream* stream_ = nullptr;
+  bool playout_ = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
 };
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index e40bc773b6..8be3b9d249 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -3340,6 +3340,21 @@ TEST_F(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
             call_.GetNetworkState(webrtc::MediaType::VIDEO));
 }
 
+// Test that playout is still started after changing parameters
+TEST_F(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
+  SetupRecvStream();
+  channel_->SetPlayout(true);
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+
+  // Changing RTP header extensions will recreate the AudioReceiveStream.
+  cricket::AudioRecvParameters parameters;
+  parameters.extensions.push_back(
+      webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
+  channel_->SetRecvParameters(parameters);
+
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+}
+
 // Tests that the library initializes and shuts down properly.
 TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   // If the VoiceEngine wants to gather available codecs early, that's fine but