diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index e2f8f14836..0d75307b4b 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -124,9 +124,6 @@ AudioReceiveStream::AudioReceiveStream(
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
-    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
-      LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri
-                      << " is no longer supported for audio.";
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index cb5fe0905c..8c03dbaa48 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -79,9 +79,6 @@ AudioSendStream::AudioSendStream(
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
-    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
-      LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri
-                      << " is no longer supported for audio.";
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
diff --git a/webrtc/config.cc b/webrtc/config.cc
index 4b437e264d..a1789a7f1d 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -86,8 +86,7 @@ const char* RtpExtension::kPlayoutDelayUri =
 const int RtpExtension::kPlayoutDelayDefaultId = 6;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
-  return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-         uri == webrtc::RtpExtension::kAudioLevelUri ||
+  return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 3bab92ddc6..3b4670ea3e 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -2071,12 +2071,14 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
 }
 
-// Test support for absolute send time header extension.
-TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
-  TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
+// Test support for transport sequence number header extension.
+TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) {
+  TestSetSendRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
 }
-TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
-  TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
+TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) {
+  TestSetRecvRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
 }
 
 // Test that we can create a channel and start sending on it.