diff --git a/call/BUILD.gn b/call/BUILD.gn
index 6ce4327475..7637f4f969 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -734,13 +734,19 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":rtp_interfaces",
+      "../api:fec_controller_api",
       "../api:frame_transformer_interface",
       "../api:libjingle_peerconnection_api",
+      "../api:scoped_refptr",
       "../api/crypto:frame_encryptor_interface",
       "../api/crypto:options",
+      "../api/transport:bandwidth_estimation_settings",
       "../api/transport:bitrate_settings",
+      "../api/transport:network_control",
+      "../api/units:timestamp",
       "../modules/pacing",
       "../modules/rtp_rtcp",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
       "../rtc_base:network_route",
       "../rtc_base:rate_limiter",
       "../rtc_base/network:sent_packet",
diff --git a/call/call.cc b/call/call.cc
index 2b7751742f..d2e8764c8c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -278,6 +278,9 @@ class Call final : public webrtc::Call,
 
   Stats GetStats() const override;
 
+  void EnableSendCongestionControlFeedbackAccordingToRfc8888() override;
+  int FeedbackAccordingToRfc8888Count() override;
+
   const FieldTrialsView& trials() const override;
 
   TaskQueueBase* network_thread() const override;
@@ -1189,6 +1192,14 @@ Call::Stats Call::GetStats() const {
   return stats;
 }
 
+void Call::EnableSendCongestionControlFeedbackAccordingToRfc8888() {
+  receive_side_cc_.EnableSendCongestionControlFeedbackAccordingToRfc8888();
+}
+
+int Call::FeedbackAccordingToRfc8888Count() {
+  return transport_send_->ReceivedCongestionControlFeedbackCount();
+}
+
 const FieldTrialsView& Call::trials() const {
   return env_.field_trials();
 }
diff --git a/call/call.h b/call/call.h
index 341d55d52b..c69e37416f 100644
--- a/call/call.h
+++ b/call/call.h
@@ -153,6 +153,9 @@ class Call {
   virtual void SetClientBitratePreferences(
       const BitrateSettings& preferences) = 0;
 
+  virtual void EnableSendCongestionControlFeedbackAccordingToRfc8888() = 0;
+  virtual int FeedbackAccordingToRfc8888Count() = 0;
+
   virtual const FieldTrialsView& trials() const = 0;
 
   virtual TaskQueueBase* network_thread() const = 0;
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index 81754f9666..8232d3c741 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -656,6 +656,7 @@ void RtpTransportControllerSend::OnCongestionControlFeedback(
     Timestamp receive_time,
     const rtcp::CongestionControlFeedback& feedback) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
+  ++feedback_count_;
   // TODO: bugs.webrtc.org/42225697 - update feedback demuxer for RFC 8888.
   // Suggest feedback_demuxer_.OnTransportFeedback use TransportPacketFeedback
   // instead. See usage in OnTransportFeedback.
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 51f8faee04..f4f29b06a8 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -142,6 +142,11 @@ class RtpTransportControllerSend final
     return controller_.get();
   }
 
+  int ReceivedCongestionControlFeedbackCount() const override {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    return feedback_count_;
+  }
+
  private:
   void MaybeCreateControllers() RTC_RUN_ON(sequence_checker_);
   void UpdateNetworkAvailability() RTC_RUN_ON(sequence_checker_);
@@ -228,6 +233,8 @@ class RtpTransportControllerSend final
 
   DataSize congestion_window_size_ RTC_GUARDED_BY(sequence_checker_);
   bool is_congested_ RTC_GUARDED_BY(sequence_checker_);
+  // Count of feedback messages received.
+  int feedback_count_ RTC_GUARDED_BY(sequence_checker_) = 0;
 
   // Protected by internal locks.
   RateLimiter retransmission_rate_limiter_;
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index a80dd83bc5..dc2e072254 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -161,6 +161,8 @@ class RtpTransportControllerSendInterface {
 
   virtual void EnsureStarted() = 0;
   virtual NetworkControllerInterface* GetNetworkController() = 0;
+  // Count of RFC8888 feedback reports received
+  virtual int ReceivedCongestionControlFeedbackCount() const = 0;
 };
 
 }  // namespace webrtc
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index 84caccf8f2..cfdcb91e79 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -11,22 +11,28 @@
 #ifndef CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
 #define CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
 
+#include <cstddef>
+#include <cstdint>
 #include <map>
 #include <memory>
-#include <string>
-#include <vector>
+#include <optional>
 
 #include "absl/strings/string_view.h"
-#include "api/crypto/crypto_options.h"
-#include "api/crypto/frame_encryptor_interface.h"
+#include "api/fec_controller.h"
 #include "api/frame_transformer_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/bandwidth_estimation_settings.h"
 #include "api/transport/bitrate_settings.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "call/rtp_config.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
-#include "rtc_base/rate_limiter.h"
 #include "test/gmock.h"
 
 namespace webrtc {
@@ -109,6 +115,10 @@ class MockRtpTransportControllerSend
               GetNetworkController,
               (),
               (override));
+  MOCK_METHOD(int,
+              ReceivedCongestionControlFeedbackCount,
+              (),
+              (const, override));
 };
 }  // namespace webrtc
 #endif  // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 0029b1d23c..ce2ec2057c 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -469,6 +469,8 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
   const webrtc::FieldTrialsView& trials() const override {
     return env_.field_trials();
   }
+  void EnableSendCongestionControlFeedbackAccordingToRfc8888() override {}
+  int FeedbackAccordingToRfc8888Count() { return 0; }
 
  private:
   webrtc::AudioSendStream* CreateAudioSendStream(
diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index 2c00735235..bc1810e9bc 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -18,6 +18,7 @@ rtc_library("congestion_controller") {
   ]
 
   deps = [
+    "..:module_api",
     "../../api:rtp_parameters",
     "../../api:sequence_checker",
     "../../api/environment",
@@ -28,6 +29,7 @@ rtc_library("congestion_controller") {
     "../../api/units:timestamp",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
+    "../../rtc_base/experiments:field_trial_parser",
     "../../rtc_base/synchronization:mutex",
     "../pacing",
     "../remote_bitrate_estimator",
@@ -56,6 +58,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../api/units:data_size",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
+      "../../rtc_base:buffer",
       "../../system_wrappers",
       "../../test:explicit_key_value_config",
       "../../test:test_support",
diff --git a/modules/congestion_controller/include/receive_side_congestion_controller.h b/modules/congestion_controller/include/receive_side_congestion_controller.h
index 7dec287660..b23ea4c86c 100644
--- a/modules/congestion_controller/include/receive_side_congestion_controller.h
+++ b/modules/congestion_controller/include/receive_side_congestion_controller.h
@@ -11,15 +11,19 @@
 #ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
 #define MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
 
+#include <cstdint>
 #include <memory>
 
 #include "absl/base/nullability.h"
 #include "api/environment/environment.h"
+#include "api/media_types.h"
 #include "api/sequence_checker.h"
 #include "api/transport/network_control.h"
 #include "api/units/data_rate.h"
+#include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "modules/congestion_controller/remb_throttler.h"
+#include "modules/include/module_common_types.h"
 #include "modules/remote_bitrate_estimator/congestion_control_feedback_generator.h"
 #include "modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -44,7 +48,7 @@ class ReceiveSideCongestionController : public CallStatsObserver {
 
   ~ReceiveSideCongestionController() override = default;
 
-  void EnablSendCongestionControlFeedbackAccordingToRfc8888();
+  void EnableSendCongestionControlFeedbackAccordingToRfc8888();
 
   void OnReceivedPacket(const RtpPacketReceived& packet, MediaType media_type);
 
diff --git a/modules/congestion_controller/receive_side_congestion_controller.cc b/modules/congestion_controller/receive_side_congestion_controller.cc
index a3d0e38867..df3a9a3b57 100644
--- a/modules/congestion_controller/receive_side_congestion_controller.cc
+++ b/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -11,6 +11,7 @@
 #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
 
 #include <algorithm>
+#include <cstdint>
 #include <memory>
 #include <utility>
 
@@ -18,13 +19,21 @@
 #include "api/environment/environment.h"
 #include "api/media_types.h"
 #include "api/sequence_checker.h"
+#include "api/transport/network_control.h"
 #include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/remb_throttler.h"
 #include "modules/remote_bitrate_estimator/congestion_control_feedback_generator.h"
 #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
 
 namespace webrtc {
 
@@ -96,12 +105,12 @@ ReceiveSideCongestionController::ReceiveSideCongestionController(
       {&force_send_rfc8888_feedback},
       env.field_trials().Lookup("WebRTC-RFC8888CongestionControlFeedback"));
   if (force_send_rfc8888_feedback) {
-    EnablSendCongestionControlFeedbackAccordingToRfc8888();
+    EnableSendCongestionControlFeedbackAccordingToRfc8888();
   }
 }
 
 void ReceiveSideCongestionController::
-    EnablSendCongestionControlFeedbackAccordingToRfc8888() {
+    EnableSendCongestionControlFeedbackAccordingToRfc8888() {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   send_rfc8888_congestion_feedback_ = true;
 }
diff --git a/modules/congestion_controller/receive_side_congestion_controller_unittest.cc b/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
index 385a6b446c..e954011a09 100644
--- a/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
+++ b/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
 
+#include <cstdint>
 #include <memory>
 #include <vector>
 
@@ -22,13 +23,18 @@
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/buffer.h"
 #include "system_wrappers/include/clock.h"
 #include "test/explicit_key_value_config.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/scenario/scenario.h"
+#include "test/scenario/scenario_config.h"
 
 namespace webrtc {
 namespace test {
@@ -133,7 +139,7 @@ TEST(ReceiveSideCongestionControllerTest, SendsRfc8888FeedbackIfEnabled) {
   ReceiveSideCongestionController controller(
       CreateEnvironment(&clock), rtcp_sender.AsStdFunction(),
       remb_sender.AsStdFunction(), nullptr);
-  controller.EnablSendCongestionControlFeedbackAccordingToRfc8888();
+  controller.EnableSendCongestionControlFeedbackAccordingToRfc8888();
 
   // Expect that RTCP feedback is sent.
   EXPECT_CALL(rtcp_sender, Call)
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9b41ba32ca..058fd37b72 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1128,6 +1128,7 @@ rtc_source_set("peer_connection") {
     "../api/transport:bitrate_settings",
     "../api/transport:datagram_transport_interface",
     "../api/transport:enums",
+    "../api/transport:network_control",
     "../api/units:time_delta",
     "../api/video:video_codec_constants",
     "../call:call_interfaces",
@@ -2567,6 +2568,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:ice_transport_interface",
       "../api:libjingle_logging_api",
       "../api:libjingle_peerconnection_api",
+      "../api:make_ref_counted",
       "../api:media_stream_interface",
       "../api:mock_async_dns_resolver",
       "../api:mock_rtp",
@@ -2577,6 +2579,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:rtp_sender_interface",
       "../api:rtp_transceiver_direction",
       "../api:scoped_refptr",
+      "../api:sequence_checker",
       "../api/audio:audio_device",
       "../api/audio:audio_mixer_api",
       "../api/audio:audio_processing",
@@ -2584,6 +2587,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/crypto:frame_decryptor_interface",
       "../api/crypto:frame_encryptor_interface",
       "../api/crypto:options",
+      "../api/metronome",
       "../api/rtc_event_log",
       "../api/rtc_event_log:rtc_event_log_factory",
       "../api/task_queue",
@@ -2630,6 +2634,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:rtc_json",
       "../rtc_base:safe_conversions",
       "../rtc_base:socket_address",
+      "../rtc_base:socket_factory",
+      "../rtc_base:socket_server",
       "../rtc_base:ssl",
       "../rtc_base:ssl_adapter",
       "../rtc_base:task_queue_for_test",
@@ -2647,6 +2653,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../test/pc/sctp:fake_sctp_transport",
       "../test/time_controller",
       "//third_party/abseil-cpp/absl/algorithm:container",
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
diff --git a/pc/congestion_control_integrationtest.cc b/pc/congestion_control_integrationtest.cc
index a60553fb5e..634c52b629 100644
--- a/pc/congestion_control_integrationtest.cc
+++ b/pc/congestion_control_integrationtest.cc
@@ -66,4 +66,21 @@ TEST_F(PeerConnectionCongestionControlTest, ReceiveOfferSetsCcfbFlag) {
   }
 }
 
+TEST_F(PeerConnectionCongestionControlTest, CcfbGetsUsed) {
+  test::ScopedFieldTrials trials(
+      "WebRTC-RFC8888CongestionControlFeedback/Enabled/");
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  MediaExpectations media_expectations;
+  media_expectations.CalleeExpectsSomeAudio();
+  media_expectations.CalleeExpectsSomeVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  auto pc_internal = caller()->pc_internal();
+  EXPECT_TRUE_WAIT(pc_internal->FeedbackAccordingToRfc8888CountForTesting() > 0,
+                   kDefaultTimeout);
+}
+
 }  // namespace webrtc
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 1171bc8929..a9559c3a6f 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -3031,6 +3031,13 @@ void PeerConnection::RequestUsagePatternReportForTesting() {
       /* delay_ms= */ 0);
 }
 
+int PeerConnection::FeedbackAccordingToRfc8888CountForTesting() const {
+  return worker_thread()->BlockingCall([this]() {
+    RTC_DCHECK_RUN_ON(worker_thread());
+    return call_->FeedbackAccordingToRfc8888Count();
+  });
+}
+
 std::function<void(const rtc::CopyOnWriteBuffer& packet,
                    int64_t packet_time_us)>
 PeerConnection::InitializeRtcpCallback() {
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 16e43c75d4..923b9be4ba 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -21,8 +21,10 @@
 #include <string>
 #include <vector>
 
+#include "absl/strings/string_view.h"
 #include "api/adaptation/resource.h"
 #include "api/async_dns_resolver.h"
+#include "api/audio/audio_device.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/data_channel_interface.h"
@@ -35,8 +37,8 @@
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "api/rtc_event_log/rtc_event_log.h"
 #include "api/rtc_event_log_output.h"
+#include "api/rtp_parameters.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_interface.h"
@@ -47,11 +49,14 @@
 #include "api/set_remote_description_observer_interface.h"
 #include "api/stats/rtc_stats_collector_callback.h"
 #include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/bandwidth_estimation_settings.h"
 #include "api/transport/bitrate_settings.h"
 #include "api/transport/data_channel_transport_interface.h"
 #include "api/transport/enums.h"
+#include "api/transport/network_control.h"
 #include "api/turn_customizer.h"
 #include "call/call.h"
+#include "call/payload_type_picker.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "p2p/base/ice_transport_internal.h"
 #include "p2p/base/port.h"
@@ -70,7 +75,6 @@
 #include "pc/rtp_transceiver.h"
 #include "pc/rtp_transmission_manager.h"
 #include "pc/rtp_transport_internal.h"
-#include "pc/sctp_data_channel.h"
 #include "pc/sdp_offer_answer.h"
 #include "pc/session_description.h"
 #include "pc/transceiver_list.h"
@@ -438,6 +442,7 @@ class PeerConnection : public PeerConnectionInternal,
     return_histogram_very_quickly_ = true;
   }
   void RequestUsagePatternReportForTesting();
+  int FeedbackAccordingToRfc8888CountForTesting() const;
 
   NetworkControllerInterface* GetNetworkController() override {
     if (!worker_thread()->IsCurrent()) {
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 0542910941..3f7e74bb1e 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -4892,6 +4892,8 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
     std::vector<
         std::pair<cricket::ChannelInterface*, const MediaContentDescription*>>
         channels;
+    bool use_ccfb = false;
+    bool seen_ccfb = false;
     for (const auto& transceiver : rtp_transceivers) {
       const ContentInfo* content_info =
           FindMediaSectionForTransceiver(transceiver, sdesc);
@@ -4904,6 +4906,17 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
       if (!content_desc) {
         continue;
       }
+      // RFC 8888 says that the ccfb must be consistent across the description.
+      if (seen_ccfb) {
+        if (use_ccfb != content_desc->rtcp_fb_ack_ccfb()) {
+          RTC_LOG(LS_ERROR)
+              << "Warning: Inconsistent CCFB flag - CCFB turned off";
+          use_ccfb = false;
+        }
+      } else {
+        use_ccfb = content_desc->rtcp_fb_ack_ccfb();
+        seen_ccfb = true;
+      }
 
       transceiver->OnNegotiationUpdate(type, content_desc);
       channels.push_back(std::make_pair(channel, content_desc));
@@ -4930,6 +4943,17 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
         LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
       }
     }
+    // If local and remote are both set, we assume that it's safe to trigger
+    // CCFB.
+    if (context_->env().field_trials().IsEnabled(
+            "WebRTC-RFC8888CongestionControlFeedback")) {
+      if (use_ccfb && local_description() && remote_description()) {
+        // The call and the congestion controller live on the worker thread.
+        context_->worker_thread()->PostTask([call = pc_->call_ptr()] {
+          call->EnableSendCongestionControlFeedbackAccordingToRfc8888();
+        });
+      }
+    }
   }
   // Need complete offer/answer with an SCTP m= section before starting SCTP,
   // according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc
index 9033e65e29..332f084747 100644
--- a/pc/test/integration_test_helpers.cc
+++ b/pc/test/integration_test_helpers.cc
@@ -10,7 +10,37 @@
 
 #include "pc/test/integration_test_helpers.h"
 
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
 #include "api/audio/builtin_audio_processing_builder.h"
+#include "api/enable_media_with_defaults.h"
+#include "api/jsep.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/sequence_checker.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/base/port_allocator.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "pc/peer_connection_factory.h"
+#include "pc/test/fake_audio_capture_module.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_network.h"
+#include "rtc_base/socket_server.h"
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 6dbbd0b1ee..7d9216e752 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -18,7 +18,6 @@
 #include <algorithm>
 #include <functional>
 #include <limits>
-#include <list>
 #include <map>
 #include <memory>
 #include <optional>
@@ -27,66 +26,47 @@
 #include <utility>
 #include <vector>
 
-#include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/audio/audio_device.h"
-#include "api/audio/audio_processing.h"
 #include "api/audio_options.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/data_channel_interface.h"
-#include "api/enable_media_with_defaults.h"
 #include "api/field_trials_view.h"
 #include "api/ice_transport_interface.h"
 #include "api/jsep.h"
+#include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/metronome/metronome.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "api/rtc_event_log/rtc_event_log_factory.h"
-#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "api/rtc_event_log_output.h"
+#include "api/rtp_parameters.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/scoped_refptr.h"
-#include "api/stats/rtc_stats.h"
+#include "api/sequence_checker.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/stats/rtcstats_objects.h"
-#include "api/task_queue/default_task_queue_factory.h"
 #include "api/task_queue/pending_task_safety_flag.h"
-#include "api/task_queue/task_queue_factory.h"
 #include "api/test/mock_async_dns_resolver.h"
-#include "api/transport/field_trial_based_config.h"
-#include "api/uma_metrics.h"
 #include "api/units/time_delta.h"
 #include "api/video/video_rotation.h"
-#include "api/video_codecs/sdp_video_format.h"
-#include "api/video_codecs/video_decoder_factory.h"
-#include "api/video_codecs/video_encoder_factory.h"
-#include "call/call.h"
 #include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
-#include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
-#include "media/engine/fake_webrtc_video_engine.h"
 #include "p2p/base/fake_ice_transport.h"
 #include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/base/port_interface.h"
-#include "p2p/base/test_stun_server.h"
 #include "p2p/base/test_turn_customizer.h"
 #include "p2p/base/test_turn_server.h"
-#include "p2p/client/basic_port_allocator.h"
-#include "pc/dtmf_sender.h"
-#include "pc/local_audio_source.h"
-#include "pc/media_session.h"
 #include "pc/peer_connection.h"
 #include "pc/peer_connection_factory.h"
 #include "pc/peer_connection_proxy.h"
-#include "pc/rtp_media_utils.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
 #include "pc/test/fake_periodic_video_source.h"
@@ -97,28 +77,23 @@
 #include "pc/video_track_source.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
-#include "rtc_base/event.h"
-#include "rtc_base/fake_clock.h"
 #include "rtc_base/fake_mdns_responder.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
 #include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/mdns_responder_interface.h"
-#include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/socket_factory.h"
+#include "rtc_base/socket_server.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/task_queue_for_test.h"
-#include "rtc_base/task_utils/repeating_task.h"
-#include "rtc_base/test_certificate_verifier.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
+#include "test/gtest.h"
 #include "test/scoped_key_value_config.h"
 
 namespace webrtc {
@@ -266,6 +241,15 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
 
   PeerConnectionInterface* pc() const { return peer_connection_.get(); }
 
+  // Return the PC implementation, so that non-public interfaces
+  // can be used in tests.
+  PeerConnection* pc_internal() const {
+    auto* pci =
+        static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(
+            pc());
+    return static_cast<PeerConnection*>(pci->internal());
+  }
+
   // If a signaling message receiver is set (via ConnectFakeSignaling), this
   // will set the whole offer/answer exchange in motion. Just need to wait for
   // the signaling state to reach "stable".