diff --git a/webrtc/common_types.cc b/webrtc/common_types.cc
index b67c18c986..7b99f3c5a8 100644
--- a/webrtc/common_types.cc
+++ b/webrtc/common_types.cc
@@ -28,6 +28,7 @@ RTPHeaderExtension::RTPHeaderExtension()
       hasTransportSequenceNumber(false),
       transportSequenceNumber(0),
       hasAudioLevel(false),
+      voiceActivity(false),
       audioLevel(0),
       hasVideoRotation(false),
       videoRotation(0) {
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index ac56eb1cf1..39ece41197 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -804,6 +804,7 @@ struct RTPHeaderExtension {
   // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
   // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
   bool hasAudioLevel;
+  bool voiceActivity;
   uint8_t audioLevel;
 
   // For Coordination of Video Orientation. See
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
index d6d1b6560a..78f0497fff 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -123,11 +123,10 @@ int main(int argc, char* argv[]) {
             packet->header().markerBit,
             packet->header().ssrc);
     if (print_audio_level && packet->header().extension.hasAudioLevel) {
-      // |audioLevel| consists of one bit for "V" and then 7 bits level.
       fprintf(out_file,
               " %5u (%1i)",
-              packet->header().extension.audioLevel & 0x7F,
-              (packet->header().extension.audioLevel & 0x80) == 0 ? 0 : 1);
+              packet->header().extension.audioLevel,
+              packet->header().extension.voiceActivity);
     }
     if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
       if (cycles == -1) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index ea6fb6adf3..409be1a66c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -320,6 +320,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
   EXPECT_FALSE(rtp_header.extension.hasAudioLevel);
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime);
+  EXPECT_FALSE(rtp_header.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header.extension.videoRotation);
 }
@@ -504,9 +505,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
-  // Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to
-  // UpdateAudioLevel(), above.
-  EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
@@ -516,6 +516,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
+  EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
 }
 
@@ -566,7 +567,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
   EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
   EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
-  EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
   EXPECT_EQ(kTransportSequenceNumber,
             rtp_header.extension.transportSequenceNumber);
 
@@ -584,6 +586,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
 
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
+  EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index 0d083bd92a..2727e7b8bc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -312,6 +312,7 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
 
   // May not be present in packet.
   header.extension.hasAudioLevel = false;
+  header.extension.voiceActivity = false;
   header.extension.audioLevel = 0;
 
   // May not be present in packet.
@@ -423,14 +424,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=0 |V|   level     |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //
-
-          // Parse out the fields but only use it for debugging for now.
-          // const uint8_t V = (*ptr & 0x80) >> 7;
-          // const uint8_t level = (*ptr & 0x7f);
-          // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
-          // level=%u", ID, len, V, level);
-
-          header.extension.audioLevel = ptr[0];
+          header.extension.audioLevel = ptr[0] & 0x7f;
+          header.extension.voiceActivity = (ptr[0] & 0x80) != 0;
           header.extension.hasAudioLevel = true;
           break;
         }