diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 8de813ae0e..14dfd90bf8 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -325,10 +325,6 @@ void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
   associated_send_stream_ = send_stream;
 }
 
-void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-}
-
 void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 49969a2779..86301a3bc6 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -91,7 +91,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetMinimumPlayoutDelay(int delay_ms) override;
 
   void AssociateSendStream(AudioSendStream* send_stream);
-  void SignalNetworkState(NetworkState state);
   void DeliverRtcp(const uint8_t* packet, size_t length);
   const webrtc::AudioReceiveStream::Config& config() const;
   const AudioSendStream* GetAssociatedSendStreamForTesting() const;
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 8933f2f567..479216aabf 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -472,10 +472,6 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   return stats;
 }
 
-void AudioSendStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(worker_thread_checker_.IsCurrent());
-}
-
 void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 3649ddf026..e063849f1a 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -79,7 +79,6 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   webrtc::AudioSendStream::Stats GetStats(
       bool has_remote_tracks) const override;
 
-  void SignalNetworkState(NetworkState state);
   void DeliverRtcp(const uint8_t* packet, size_t length);
 
   // Implements BitrateAllocatorObserver.
diff --git a/call/call.cc b/call/call.cc
index 62a4378a6c..e5eef1970b 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -648,7 +648,6 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
       }
     }
   }
-  send_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return send_stream;
 }
@@ -706,7 +705,6 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
       receive_stream->AssociateSendStream(it->second);
     }
   }
-  receive_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return receive_stream;
 }
@@ -1010,17 +1008,8 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   }
 
   UpdateAggregateNetworkState();
-  {
-    ReadLockScoped read_lock(*send_crit_);
-    for (auto& kv : audio_send_ssrcs_) {
-      kv.second->SignalNetworkState(audio_network_state_);
-    }
-  }
   {
     ReadLockScoped read_lock(*receive_crit_);
-    for (AudioReceiveStream* audio_receive_stream : audio_receive_streams_) {
-      audio_receive_stream->SignalNetworkState(audio_network_state_);
-    }
     for (VideoReceiveStream* video_receive_stream : video_receive_streams_) {
       video_receive_stream->SignalNetworkState(video_network_state_);
     }