diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 6717878abb..d4c7910f56 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -146,7 +146,7 @@ AudioReceiveStream::AudioReceiveStream(
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
+  RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << remote_ssrc();
   Stop();
   channel_receive_->SetAssociatedSendChannel(nullptr);
   channel_receive_->ResetReceiverCongestionControlObjects();
@@ -157,7 +157,7 @@ void AudioReceiveStream::RegisterWithTransport(
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   RTC_DCHECK(!rtp_stream_receiver_);
   rtp_stream_receiver_ = receiver_controller->CreateReceiver(
-      config_.rtp.remote_ssrc, channel_receive_.get());
+      remote_ssrc(), channel_receive_.get());
 }
 
 void AudioReceiveStream::UnregisterFromTransport() {
@@ -170,8 +170,8 @@ void AudioReceiveStream::ReconfigureForTesting(
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
 
   // SSRC can't be changed mid-stream.
-  RTC_DCHECK_EQ(config_.rtp.remote_ssrc, config.rtp.remote_ssrc);
-  RTC_DCHECK_EQ(config_.rtp.local_ssrc, config.rtp.local_ssrc);
+  RTC_DCHECK_EQ(remote_ssrc(), config.rtp.remote_ssrc);
+  RTC_DCHECK_EQ(local_ssrc(), config.rtp.local_ssrc);
 
   // Configuration parameters which cannot be changed.
   RTC_DCHECK_EQ(config_.rtcp_send_transport, config.rtcp_send_transport);
@@ -269,7 +269,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
     bool get_and_clear_legacy_stats) const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   webrtc::AudioReceiveStream::Stats stats;
-  stats.remote_ssrc = config_.rtp.remote_ssrc;
+  stats.remote_ssrc = remote_ssrc();
 
   webrtc::CallReceiveStatistics call_stats =
       channel_receive_->GetRTCPStatistics();
@@ -398,7 +398,7 @@ AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
 }
 
 int AudioReceiveStream::Ssrc() const {
-  return config_.rtp.remote_ssrc;
+  return remote_ssrc();
 }
 
 int AudioReceiveStream::PreferredSampleRate() const {
@@ -407,7 +407,7 @@ int AudioReceiveStream::PreferredSampleRate() const {
 
 uint32_t AudioReceiveStream::id() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return config_.rtp.remote_ssrc;
+  return remote_ssrc();
 }
 
 absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
@@ -471,9 +471,9 @@ uint32_t AudioReceiveStream::local_ssrc() const {
   return config_.rtp.local_ssrc;
 }
 
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return config_;
+const std::string& AudioReceiveStream::sync_group() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.sync_group;
 }
 
 const AudioSendStream* AudioReceiveStream::GetAssociatedSendStreamForTesting()
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 3f4675cd72..6fc9f7fe62 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -134,7 +134,10 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
     return config_.rtp.remote_ssrc;
   }
 
-  const webrtc::AudioReceiveStream::Config& config() const;
+  // Returns a reference to the currently set sync group of the stream.
+  // Must be called on the packet delivery thread.
+  const std::string& sync_group() const;
+
   const AudioSendStream* GetAssociatedSendStreamForTesting() const;
 
   // TODO(tommi): Remove this method.
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 1d6183a1d4..a1e8c1f61f 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -131,6 +131,8 @@ struct ConfigHelper {
           EXPECT_THAT(codecs, ::testing::IsEmpty());
         }));
     EXPECT_CALL(*channel_receive_, SetSourceTracker(_));
+    EXPECT_CALL(*channel_receive_, GetLocalSsrc())
+        .WillRepeatedly(Return(kLocalSsrc));
 
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
diff --git a/call/call.cc b/call/call.cc
index 5b306610ea..1a0357fc75 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1009,8 +1009,9 @@ void Call::DestroyAudioReceiveStream(
   audio_receive_stream->UnregisterFromTransport();
 
   uint32_t ssrc = audio_receive_stream->remote_ssrc();
-  const AudioReceiveStream::Config& config = audio_receive_stream->config();
-  receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config.rtp))
+  receive_side_cc_
+      .GetRemoteBitrateEstimator(
+          UseSendSideBwe(audio_receive_stream->rtp_config()))
       ->RemoveStream(ssrc);
 
   audio_receive_streams_.erase(audio_receive_stream);
@@ -1018,7 +1019,7 @@ void Call::DestroyAudioReceiveStream(
   // After calling erase(), call ConfigureSync. This will clear associated
   // video streams or associate them with a different audio stream if one exists
   // for this sync_group.
-  ConfigureSync(audio_receive_stream->config().sync_group);
+  ConfigureSync(audio_receive_stream->sync_group());
 
   UnregisterReceiveStream(ssrc);
 
@@ -1471,7 +1472,7 @@ AudioReceiveStream* Call::FindAudioStreamForSyncGroup(
   RTC_DCHECK_RUN_ON(&receive_11993_checker_);
   if (!sync_group.empty()) {
     for (AudioReceiveStream* stream : audio_receive_streams_) {
-      if (stream->config().sync_group == sync_group)
+      if (stream->sync_group() == sync_group)
         return stream;
     }
   }