diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index aa53ff89a3..56c23d905f 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -334,9 +334,6 @@ RtpVideoSender::RtpVideoSender(
     : send_side_bwe_with_overhead_(absl::StartsWith(
           field_trials_.Lookup("WebRTC-SendSideBwe-WithOverhead"),
           "Enabled")),
-      account_for_packetization_overhead_(!absl::StartsWith(
-          field_trials_.Lookup("WebRTC-SubtractPacketizationOverhead"),
-          "Disabled")),
       has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
       use_deferred_fec_(!absl::StartsWith(
           field_trials_.Lookup("WebRTC-DeferredFecGeneration"),
@@ -786,16 +783,13 @@ void RtpVideoSender::OnBitrateUpdated(BitrateAllocationUpdate update,
     // since |fec_allowed_| may be toggled back on at any moment.
   }
 
-  uint32_t packetization_rate_bps = 0;
-  if (account_for_packetization_overhead_) {
     // Subtract packetization overhead from the encoder target. If target rate
     // is really low, cap the overhead at 50%. This also avoids the case where
     // |encoder_target_rate_bps_| is 0 due to encoder pause event while the
     // packetization rate is positive since packets are still flowing.
-    packetization_rate_bps =
-        std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
-    encoder_target_rate_bps_ -= packetization_rate_bps;
-  }
+  uint32_t packetization_rate_bps =
+      std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
+  encoder_target_rate_bps_ -= packetization_rate_bps;
 
   loss_mask_vector_.clear();
 
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 33dd5b744a..5eaaa29f2d 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -171,7 +171,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
 
   const FieldTrialBasedConfig field_trials_;
   const bool send_side_bwe_with_overhead_;
-  const bool account_for_packetization_overhead_;
   const bool has_packet_feedback_;
   const bool use_deferred_fec_;
 
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 6428207e62..721738393b 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -319,7 +319,6 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
   // test, due to the packetization overhead and encoder pushback.
   webrtc::test::ScopedFieldTrials field_trials(
       std::string(field_trial::GetFieldTrialString()) +
-      "WebRTC-SubtractPacketizationOverhead/Disabled/"
       "WebRTC-VideoRateControl/bitrate_adjuster:false/");
   class EncoderRateStatsTest : public test::EndToEndTest,
                                public test::FakeEncoder {
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 24795bb2cd..02e8b2bd7f 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -2936,10 +2936,9 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
   static const int kIncreasedStartBitrateKbps = 451;
   static const int kIncreasedMaxBitrateKbps = 597;
   // If these fields trial are on, we get lower bitrates than expected by this
-  // test, due to the packetization overhead and encoder pushback.
+  // test, due to encoder pushback.
   webrtc::test::ScopedFieldTrials field_trials(
       std::string(field_trial::GetFieldTrialString()) +
-      "WebRTC-SubtractPacketizationOverhead/Disabled/"
       "WebRTC-VideoRateControl/bitrate_adjuster:false/");
 
   class EncoderBitrateThresholdObserver : public test::SendTest,