diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 0bddb42807..07def99559 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -73,6 +73,12 @@ AudioMixer::Source::AudioFrameInfo AudioIngress::GetAudioFrameWithInfo(
   constexpr double kAudioSampleDurationSeconds = 0.01;
   output_audio_level_.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
 
+  // If caller invoked StopPlay(), then mute the frame.
+  if (!playing_) {
+    AudioFrameOperations::Mute(audio_frame);
+    muted = true;
+  }
+
   // Set first rtp timestamp with first audio frame with valid timestamp.
   if (first_rtp_timestamp_ < 0 && audio_frame->timestamp_ != 0) {
     first_rtp_timestamp_ = audio_frame->timestamp_;
@@ -127,10 +133,6 @@ void AudioIngress::SetReceiveCodecs(
 }
 
 void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
-  if (!IsPlaying()) {
-    return;
-  }
-
   RtpPacketReceived rtp_packet_received;
   rtp_packet_received.Parse(rtp_packet.data(), rtp_packet.size());
 
diff --git a/audio/voip/test/audio_ingress_unittest.cc b/audio/voip/test/audio_ingress_unittest.cc
index 3a2a66a325..01b4d67dad 100644
--- a/audio/voip/test/audio_ingress_unittest.cc
+++ b/audio/voip/test/audio_ingress_unittest.cc
@@ -181,5 +181,48 @@ TEST_F(AudioIngressTest, PreferredSampleRate) {
   EXPECT_EQ(ingress_->PreferredSampleRate(), kPcmuFormat.clockrate_hz);
 }
 
+// This test highlights the case where caller invokes StopPlay() which then
+// AudioIngress should play silence frame afterwards.
+TEST_F(AudioIngressTest, GetMutedAudioFrameAfterRtpReceivedAndStopPlay) {
+  // StopPlay before we start sending RTP packet with sine wave.
+  ingress_->StopPlay();
+
+  // Send 6 RTP packets to generate more than 100 ms audio sample to get
+  // valid speech level.
+  constexpr int kNumRtp = 6;
+  int rtp_count = 0;
+  rtc::Event event;
+  auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) {
+    ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length));
+    if (++rtp_count == kNumRtp) {
+      event.Set();
+    }
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
+  for (int i = 0; i < kNumRtp * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    fake_clock_.AdvanceTimeMilliseconds(10);
+  }
+  event.Wait(/*give_up_after_ms=*/1000);
+
+  for (int i = 0; i < kNumRtp * 2; ++i) {
+    AudioFrame audio_frame;
+    EXPECT_EQ(
+        ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
+        AudioMixer::Source::AudioFrameInfo::kMuted);
+    const int16_t* audio_data = audio_frame.data();
+    size_t length =
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_;
+    for (size_t j = 0; j < length; ++j) {
+      EXPECT_EQ(audio_data[j], 0);
+    }
+  }
+
+  // Now we should still see valid speech output level as StopPlay won't affect
+  // the measurement.
+  EXPECT_EQ(ingress_->GetSpeechOutputLevelFullRange(), kAudioLevel);
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index a93df73a22..2a29033a2b 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -161,8 +161,7 @@ void VoipCore::ReleaseChannel(ChannelId channel) {
   // Destroy channel outside of the lock.
   rtc::scoped_refptr<AudioChannel> audio_channel;
 
-  // Check if process thread is no longer needed.
-  bool stop_process_thread = false;
+  bool no_channels_after_release = false;
 
   {
     MutexLock lock(&lock_);
@@ -173,18 +172,24 @@ void VoipCore::ReleaseChannel(ChannelId channel) {
       channels_.erase(iter);
     }
 
-    // Check if this is the last channel we have.
-    stop_process_thread = channels_.empty();
+    no_channels_after_release = channels_.empty();
   }
 
   if (!audio_channel) {
     RTC_LOG(LS_WARNING) << "Channel " << channel << " not found";
   }
 
-  if (stop_process_thread) {
+  if (no_channels_after_release) {
     // Release audio channel first to have it DeRegisterModule first.
     audio_channel = nullptr;
     process_thread_->Stop();
+
+    // Make sure to stop playout on ADM if it is playing.
+    if (audio_device_module_->Playing()) {
+      if (audio_device_module_->StopPlayout() != 0) {
+        RTC_LOG(LS_WARNING) << "StopPlayout failed";
+      }
+    }
   }
 }
 
@@ -280,7 +285,15 @@ bool VoipCore::StopSend(ChannelId channel) {
 
 bool VoipCore::StartPlayout(ChannelId channel) {
   auto audio_channel = GetChannel(channel);
-  if (!audio_channel || !audio_channel->StartPlay()) {
+  if (!audio_channel) {
+    return false;
+  }
+
+  if (audio_channel->IsPlaying()) {
+    return true;
+  }
+
+  if (!audio_channel->StartPlay()) {
     return false;
   }
 
@@ -305,24 +318,6 @@ bool VoipCore::StopPlayout(ChannelId channel) {
 
   audio_channel->StopPlay();
 
-  bool stop_device = true;
-  {
-    MutexLock lock(&lock_);
-    for (auto kv : channels_) {
-      rtc::scoped_refptr<AudioChannel>& channel = kv.second;
-      if (channel->IsPlaying()) {
-        stop_device = false;
-        break;
-      }
-    }
-  }
-
-  if (stop_device && audio_device_module_->Playing()) {
-    if (audio_device_module_->StopPlayout() != 0) {
-      RTC_LOG(LS_ERROR) << "StopPlayout failed";
-      return false;
-    }
-  }
   return true;
 }