diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 9759f1799d..bcfa650c02 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -951,6 +951,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
   constexpr uint32_t kFlexfecSsrc = 5678;
   const std::vector<RtpExtension> kNoRtpExtensions;
   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
+
   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                                kNoRtpExtensions, kNoRtpExtensionSizes,
                                nullptr /* rtp_state */, &fake_clock_);
@@ -966,6 +967,10 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
+  // Need extension to be registered for timing frames to be sent.
+  ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                   kRtpExtensionVideoTiming, kVideoTimingExtensionId));
+
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
   params.fec_rate = 15;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 351056e216..94503dcdf1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -304,7 +304,6 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
   auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
 
   size_t fec_packet_overhead;
-  bool is_timing_frame = false;
   bool red_enabled;
   int32_t retransmission_settings;
   {
@@ -336,7 +335,6 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
       if (video_header->video_timing.is_timing_frame) {
         last_packet->SetExtension<VideoTimingExtension>(
             video_header->video_timing);
-        is_timing_frame = true;
       }
     }
 
@@ -396,7 +394,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
 
     bool protect_packet = (packetizer->GetProtectionType() == kProtectedPacket);
     // Put packetization finish timestamp into extension.
-    if (last && is_timing_frame) {
+    if (packet->HasExtension<VideoTimingExtension>()) {
       packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
       // TODO(ilnik): Due to webrtc:7859, packets with timing extensions are not
       // protected by FEC. It reduces FEC efficiency a bit. When FEC is moved