diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index af66cd3634..c6bac621c3 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -491,23 +491,21 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
 
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
-  // |audio_network_adaptor_| is supposed to be configured to output all
-  // following parameters.
-  RTC_DCHECK(config.bitrate_bps);
-  RTC_DCHECK(config.frame_length_ms);
-  RTC_DCHECK(config.uplink_packet_loss_fraction);
-  RTC_DCHECK(config.enable_fec);
-  RTC_DCHECK(config.enable_dtx);
-  RTC_DCHECK(config.num_channels);
+  RTC_DCHECK(!config.frame_length_ms || *config.frame_length_ms == 20 ||
+             *config.frame_length_ms == 60);
 
-  RTC_DCHECK(*config.frame_length_ms == 20 || *config.frame_length_ms == 60);
-
-  SetTargetBitrate(*config.bitrate_bps);
-  SetFrameLength(*config.frame_length_ms);
-  SetFec(*config.enable_fec);
-  SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
-  SetDtx(*config.enable_dtx);
-  SetNumChannelsToEncode(*config.num_channels);
+  if (config.bitrate_bps)
+    SetTargetBitrate(*config.bitrate_bps);
+  if (config.frame_length_ms)
+    SetFrameLength(*config.frame_length_ms);
+  if (config.enable_fec)
+    SetFec(*config.enable_fec);
+  if (config.uplink_packet_loss_fraction)
+    SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
+  if (config.enable_dtx)
+    SetDtx(*config.enable_dtx);
+  if (config.num_channels)
+    SetNumChannelsToEncode(*config.num_channels);
 }
 
 std::unique_ptr<AudioNetworkAdaptor>
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index e8bfaf6830..f024af4cef 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -443,4 +443,25 @@ TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
   config.bitrate_bps = rtc::Optional<int>(14001);
   EXPECT_EQ(rtc::Optional<int>(6), config.GetNewComplexity());
 }
+
+TEST(AudioEncoderOpusTest, EmptyConfigDontAffectEncoderSettings) {
+  auto states = CreateCodec(2);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  AudioNetworkAdaptor::EncoderRuntimeConfig empty_config;
+
+  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config))
+      .WillOnce(Return(empty_config));
+
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead))
+      .Times(2);
+  states.encoder->OnReceivedOverhead(kOverhead);
+  states.encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states.encoder.get(), config);
+}
+
 }  // namespace webrtc