diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 931f9e39a2..1b6df005d6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -29,6 +29,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ -1293,10 +1294,16 @@ void RTPSender::SetTransportOverhead(int transport_overhead) {
 void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
                                              const RtpPacketToSend& packet,
                                              int probe_cluster_id) {
+  size_t packet_size = packet.payload_size() + packet.padding_size();
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
+      "Enabled") {
+    rtc::CritScope lock(&send_critsect_);
+    packet_size = packet.size() + transport_overhead_bytes_per_packet_;
+  }
+
   if (transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(
-        packet_id, packet.payload_size() + packet.padding_size(),
-        probe_cluster_id);
+    transport_feedback_observer_->AddPacket(packet_id, packet_size,
+                                            probe_cluster_id);
   }
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index a9e9485e5c..69c1c00807 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -27,6 +27,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/stl_util.h"
+#include "webrtc/test/field_trial.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_transport.h"
@@ -1428,4 +1429,34 @@ TEST_F(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
   SendGenericPayload();
 }
 
+TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) {
+  constexpr int kTransportOverheadBytesPerPacket = 28;
+  constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
+      &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
+      nullptr, &retransmission_rate_limiter_, &mock_overhead_observer));
+  rtp_sender_->SetTransportOverhead(kTransportOverheadBytesPerPacket);
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                   kRtpExtensionTransportSequenceNumber,
+                   kTransportSequenceNumberExtensionId));
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(feedback_observer_,
+              AddPacket(kTransportSequenceNumber,
+                        sizeof(kPayloadData) + kGenericHeaderLength +
+                            kRtpOverheadBytesPerPacket +
+                            kTransportOverheadBytesPerPacket,
+                        PacketInfo::kNotAProbe))
+      .Times(1);
+  EXPECT_CALL(mock_overhead_observer,
+              OnOverheadChanged(kTransportOverheadBytesPerPacket +
+                                kRtpOverheadBytesPerPacket))
+      .Times(1);
+  SendGenericPayload();
+}
+
 }  // namespace webrtc