diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7618dded72..7d70592f76 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -311,24 +311,6 @@ void AudioSendStream::ConfigureStream(
     rtp_rtcp_module_->SetMid(new_config.rtp.mid);
   }
 
-  // RID RTP header extension
-  if ((first_time || new_ids.rid != old_ids.rid ||
-       new_ids.repaired_rid != old_ids.repaired_rid ||
-       new_config.rtp.rid != old_config.rtp.rid)) {
-    if (new_ids.rid != 0 || new_ids.repaired_rid != 0) {
-      if (new_config.rtp.rid.empty()) {
-        rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(RtpStreamId::Uri());
-      } else if (new_ids.repaired_rid != 0) {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
-                                                     new_ids.repaired_rid);
-      } else {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
-                                                     new_ids.rid);
-      }
-    }
-    rtp_rtcp_module_->SetRid(new_config.rtp.rid);
-  }
-
   if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
     absl::string_view uri = AbsoluteCaptureTimeExtension::Uri();
     rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 6ac5f4e8ed..3c5b1423b2 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -246,7 +246,6 @@ struct ConfigHelper {
           .Times(1);
     }
     EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
-    EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1);
   }
 
   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index ad72aae28e..5d4455c2de 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -662,12 +662,10 @@ WebRtcVoiceEngine::GetRtpHeaderExtensions() const {
   RTC_DCHECK(signal_thread_checker_.IsCurrent());
   std::vector<webrtc::RtpHeaderExtensionCapability> result;
   int id = 1;
-  for (const auto& uri :
-       {webrtc::RtpExtension::kAudioLevelUri,
-        webrtc::RtpExtension::kAbsSendTimeUri,
-        webrtc::RtpExtension::kTransportSequenceNumberUri,
-        webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kRidUri,
-        webrtc::RtpExtension::kRepairedRidUri}) {
+  for (const auto& uri : {webrtc::RtpExtension::kAudioLevelUri,
+                          webrtc::RtpExtension::kAbsSendTimeUri,
+                          webrtc::RtpExtension::kTransportSequenceNumberUri,
+                          webrtc::RtpExtension::kMidUri}) {
     result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
   }
   return result;