diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index 13bc98297f..fc97b0b503 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
+
 MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
     int sampling_rate_hz,
     int starting_sample_index,
@@ -35,7 +37,19 @@ MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
       payload_type_(payload_type),
       encoded_data_(std::move(encoded_data)) {}
 
-MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() = default;
+MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
+    const MediaTransportEncodedAudioFrame&) = default;
+
+MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
+    MediaTransportEncodedAudioFrame&&) = default;
+
+MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
+    const MediaTransportEncodedAudioFrame&) = default;
+
+MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
+    MediaTransportEncodedAudioFrame&&) = default;
+
+MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() {}
 
 MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
     int64_t frame_id,
@@ -47,6 +61,16 @@ MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
       frame_id_(frame_id),
       referenced_frame_ids_(std::move(referenced_frame_ids)) {}
 
-MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() = default;
+MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
+    const MediaTransportEncodedVideoFrame&) = default;
+
+MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
+    MediaTransportEncodedVideoFrame&&) = default;
+
+MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
+    const MediaTransportEncodedVideoFrame&) = default;
+
+MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
+    MediaTransportEncodedVideoFrame&&) = default;
 
 }  // namespace webrtc
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index 154fe4a825..33fa2adf6c 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -1,5 +1,4 @@
-/*
- *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+/* Copyright 2018 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -35,7 +34,7 @@ namespace webrtc {
 
 // Represents encoded audio frame in any encoding (type of encoding is opaque).
 // To avoid copying of encoded data use move semantics when passing by value.
-class MediaTransportEncodedAudioFrame {
+class MediaTransportEncodedAudioFrame final {
  public:
   enum class FrameType {
     // Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
@@ -79,6 +78,12 @@ class MediaTransportEncodedAudioFrame {
       std::vector<uint8_t> encoded_data);
 
   ~MediaTransportEncodedAudioFrame();
+  MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
+  MediaTransportEncodedAudioFrame& operator=(
+      const MediaTransportEncodedAudioFrame& other);
+  MediaTransportEncodedAudioFrame& operator=(
+      MediaTransportEncodedAudioFrame&& other);
+  MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
 
   // Getters.
   int sampling_rate_hz() const { return sampling_rate_hz_; }
@@ -121,13 +126,19 @@ class MediaTransportAudioSinkInterface {
 };
 
 // Represents encoded video frame, along with the codec information.
-class MediaTransportEncodedVideoFrame {
+class MediaTransportEncodedVideoFrame final {
  public:
   MediaTransportEncodedVideoFrame(int64_t frame_id,
                                   std::vector<int64_t> referenced_frame_ids,
                                   VideoCodecType codec_type,
                                   const webrtc::EncodedImage& encoded_image);
   ~MediaTransportEncodedVideoFrame();
+  MediaTransportEncodedVideoFrame(const MediaTransportEncodedVideoFrame&);
+  MediaTransportEncodedVideoFrame& operator=(
+      const MediaTransportEncodedVideoFrame& other);
+  MediaTransportEncodedVideoFrame& operator=(
+      MediaTransportEncodedVideoFrame&& other);
+  MediaTransportEncodedVideoFrame(MediaTransportEncodedVideoFrame&&);
 
   VideoCodecType codec_type() const { return codec_type_; }
   const webrtc::EncodedImage& encoded_image() const { return encoded_image_; }
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index c6850e17ee..3d8a9c1e41 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -80,6 +80,7 @@
 #include "api/datachannelinterface.h"
 #include "api/fec_controller.h"
 #include "api/jsep.h"
+#include "api/media_transport_interface.h"
 #include "api/mediastreaminterface.h"
 #include "api/rtcerror.h"
 #include "api/rtceventlogoutput.h"
@@ -560,6 +561,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     // correctly. This flag will be deprecated soon. Do not rely on it.
     bool active_reset_srtp_params = false;
 
+    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
+    // informs PeerConnection that it should use the MediaTransportInterface.
+    // It's invalid to set it to |true| if the MediaTransportFactory wasn't
+    // provided.
+    bool use_media_transport = false;
+
     //
     // Don't forget to update operator== if adding something.
     //
@@ -1153,6 +1160,7 @@ struct PeerConnectionFactoryDependencies final {
   std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
   std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
+  std::unique_ptr<MediaTransportFactory> media_transport_factory;
 };
 
 // PeerConnectionFactoryInterface is the factory interface used for creating
diff --git a/pc/createpeerconnectionfactory.cc b/pc/createpeerconnectionfactory.cc
index bbc4ebbe73..b1db69091b 100644
--- a/pc/createpeerconnectionfactory.cc
+++ b/pc/createpeerconnectionfactory.cc
@@ -129,10 +129,14 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
 
   std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
       CreateRtcEventLogFactory();
-
-  return CreateModularPeerConnectionFactory(
-      network_thread, worker_thread, signaling_thread, std::move(media_engine),
-      std::move(call_factory), std::move(event_log_factory));
+  PeerConnectionFactoryDependencies dependencies;
+  dependencies.network_thread = network_thread;
+  dependencies.worker_thread = worker_thread;
+  dependencies.signaling_thread = signaling_thread;
+  dependencies.media_engine = std::move(media_engine);
+  dependencies.call_factory = std::move(call_factory);
+  dependencies.event_log_factory = std::move(event_log_factory);
+  return CreateModularPeerConnectionFactory(std::move(dependencies));
 }
 
 #if defined(USE_BUILTIN_SW_CODECS)
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index f920ddf72c..7430898912 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -705,6 +705,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
     SdpSemantics sdp_semantics;
     absl::optional<rtc::AdapterType> network_preference;
     bool active_reset_srtp_params;
+    bool use_media_transport;
   };
   static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
                 "Did you add something to RTCConfiguration and forget to "
@@ -752,7 +753,8 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
          turn_customizer == o.turn_customizer &&
          sdp_semantics == o.sdp_semantics &&
          network_preference == o.network_preference &&
-         active_reset_srtp_params == o.active_reset_srtp_params;
+         active_reset_srtp_params == o.active_reset_srtp_params &&
+         use_media_transport == o.use_media_transport;
 }
 
 bool PeerConnectionInterface::RTCConfiguration::operator!=(
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 37c9d73bd4..b42d620107 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "api/media_transport_interface.h"
 #include "api/peerconnectioninterface.h"
 #include "api/turncustomizer.h"
 #include "pc/iceserverparsing.h"
@@ -1026,6 +1027,10 @@ class PeerConnection : public PeerConnectionInternal,
   // List of content names for which the remote side triggered an ICE restart.
   std::set<std::string> pending_ice_restarts_;
 
+  // Optional media transport for sending / receiving encoded frames.
+  // If available, media transport will be used instead of RTP / SRTP.
+  std::unique_ptr<MediaTransportFactory> media_transport_factory_;
+
   std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
 
   // Member variables for caching global options.
diff --git a/pc/peerconnectionfactory.cc b/pc/peerconnectionfactory.cc
index b78ed8db1e..edc13034a3 100644
--- a/pc/peerconnectionfactory.cc
+++ b/pc/peerconnectionfactory.cc
@@ -15,6 +15,7 @@
 
 #include "absl/memory/memory.h"
 #include "api/fec_controller.h"
+#include "api/media_transport_interface.h"
 #include "api/mediaconstraintsinterface.h"
 #include "api/mediastreamproxy.h"
 #include "api/mediastreamtrackproxy.h"
@@ -187,7 +188,9 @@ PeerConnectionFactory::PeerConnectionFactory(
           std::move(dependencies.call_factory),
           std::move(dependencies.event_log_factory),
           std::move(dependencies.fec_controller_factory),
-          std::move(dependencies.network_controller_factory)) {}
+          std::move(dependencies.network_controller_factory)) {
+  media_transport_factory_ = std::move(dependencies.media_transport_factory);
+}
 
 PeerConnectionFactory::~PeerConnectionFactory() {
   RTC_DCHECK(signaling_thread_->IsCurrent());
diff --git a/pc/peerconnectionfactory.h b/pc/peerconnectionfactory.h
index f36a4e0c04..1c1ea94fe2 100644
--- a/pc/peerconnectionfactory.h
+++ b/pc/peerconnectionfactory.h
@@ -15,6 +15,7 @@
 #include <memory>
 #include <string>
 
+#include "api/media_transport_interface.h"
 #include "api/mediastreaminterface.h"
 #include "api/peerconnectioninterface.h"
 #include "media/sctp/sctptransportinternal.h"
@@ -96,6 +97,10 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
   virtual rtc::Thread* network_thread();
   const Options& options() const { return options_; }
 
+  MediaTransportFactory* media_transport_factory() {
+    return media_transport_factory_.get();
+  }
+
  protected:
   PeerConnectionFactory(
       rtc::Thread* network_thread,
@@ -148,6 +153,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
       injected_network_controller_factory_;
   std::unique_ptr<NetworkControllerFactoryInterface>
       bbr_network_controller_factory_;
+  std::unique_ptr<MediaTransportFactory> media_transport_factory_;
 };
 
 }  // namespace webrtc
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index c4bee38d6c..b259f51ef4 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -1170,6 +1170,7 @@ if (is_ios || is_mac) {
           ":videoframebuffer_objc",
           ":videosource_objc",
           ":videotoolbox_objc",
+          "../api:libjingle_peerconnection_api",
           "../api/audio_codecs:audio_codecs_api",
           "../api/audio_codecs:builtin_audio_decoder_factory",
           "../api/audio_codecs:builtin_audio_encoder_factory",
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index 9fb8b4df35..c7998a5aa2 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -168,6 +168,12 @@ RTC_OBJC_EXPORT
  */
 @property(nonatomic, assign) BOOL activeResetSrtpParams;
 
+/**
+ * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
+ * that it should use the MediaTransportInterface.
+ */
+@property(nonatomic, assign) BOOL useMediaTransport;
+
 - (instancetype)init;
 
 @end
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 73a0fa9aa5..bdf4fb6e43 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -48,6 +48,7 @@
 @synthesize sdpSemantics = _sdpSemantics;
 @synthesize turnCustomizer = _turnCustomizer;
 @synthesize activeResetSrtpParams = _activeResetSrtpParams;
+@synthesize useMediaTransport = _useMediaTransport;
 
 - (instancetype)init {
   // Copy defaults.
@@ -117,7 +118,7 @@
 - (NSString *)description {
   static NSString *formatString =
       @"RTCConfiguration: "
-      @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n%d\n%d\n}\n";
+      @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n%d\n%d\n%@\n}\n";
 
   return [NSString
       stringWithFormat:formatString,
@@ -140,7 +141,8 @@
                        _iceRegatherIntervalRange,
                        _disableLinkLocalNetworks,
                        _maxIPv6Networks,
-                       _activeResetSrtpParams];
+                       _activeResetSrtpParams,
+                       _useMediaTransport];
 }
 
 #pragma mark - Private
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
index 5277489747..1ac076b3c2 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnection.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
@@ -29,6 +29,7 @@
 #include <memory>
 
 #include "api/jsepicecandidate.h"
+#include "api/media_transport_interface.h"
 #include "rtc_base/checks.h"
 
 NSString * const kRTCPeerConnectionErrorDomain =
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
index 60aa08f7ca..2752cf4117 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
@@ -17,6 +17,7 @@ namespace webrtc {
 class AudioDeviceModule;
 class AudioEncoderFactory;
 class AudioDecoderFactory;
+class MediaTransportFactory;
 class VideoEncoderFactory;
 class VideoDecoderFactory;
 class AudioProcessing;
@@ -49,6 +50,25 @@ NS_ASSUME_NONNULL_BEGIN
                             audioProcessingModule:
                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
 
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+            nativeAudioDecoderFactory:
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+            nativeVideoEncoderFactory:
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+            nativeVideoDecoderFactory:
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
+                audioProcessingModule:
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+                mediaTransportFactory:
+                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
+
+- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory
+                 mediaTransportFactory:
+                     (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
 @end
 
 NS_ASSUME_NONNULL_END
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
index 403db040fb..48476f2368 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -50,6 +50,7 @@
 // TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
 // API layer.
 #include "absl/memory/memory.h"
+#include "api/media_transport_interface.h"
 #include "media/engine/webrtcmediaengine.h"  // nogncheck
 
 @implementation RTCPeerConnectionFactory {
@@ -80,12 +81,15 @@
                        nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory(
                                                      [[RTCVideoDecoderFactoryH264 alloc] init])
                                audioDeviceModule:[self audioDeviceModule]
-                           audioProcessingModule:nullptr];
+                           audioProcessingModule:nullptr
+                           mediaTransportFactory:nullptr];
 #endif
 }
 
 - (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
-                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory
+                 mediaTransportFactory:
+                     (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
 #ifdef HAVE_NO_MEDIA
   return [self initWithNoMedia];
 #else
@@ -102,9 +106,16 @@
                        nativeVideoEncoderFactory:std::move(native_encoder_factory)
                        nativeVideoDecoderFactory:std::move(native_decoder_factory)
                                audioDeviceModule:[self audioDeviceModule]
-                           audioProcessingModule:nullptr];
+                           audioProcessingModule:nullptr
+                           mediaTransportFactory:std::move(mediaTransportFactory)];
 #endif
 }
+- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
+  return [self initWithEncoderFactory:encoderFactory
+                       decoderFactory:decoderFactory
+                mediaTransportFactory:nullptr];
+}
 
 - (instancetype)initNative {
   if (self = [super init]) {
@@ -152,20 +163,57 @@
                                     (nullable webrtc::AudioDeviceModule *)audioDeviceModule
                             audioProcessingModule:
                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
+  return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
+                       nativeAudioDecoderFactory:audioDecoderFactory
+                       nativeVideoEncoderFactory:std::move(videoEncoderFactory)
+                       nativeVideoDecoderFactory:std::move(videoDecoderFactory)
+                               audioDeviceModule:audioDeviceModule
+                           audioProcessingModule:audioProcessingModule
+                           mediaTransportFactory:nullptr];
+}
+
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+            nativeAudioDecoderFactory:
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+            nativeVideoEncoderFactory:
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+            nativeVideoDecoderFactory:
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
+                audioProcessingModule:
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+                mediaTransportFactory:
+                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
 #ifdef HAVE_NO_MEDIA
   return [self initWithNoMedia];
 #else
   if (self = [self initNative]) {
-    _nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
-                                                         _workerThread.get(),
-                                                         _signalingThread.get(),
-                                                         audioDeviceModule,
-                                                         audioEncoderFactory,
-                                                         audioDecoderFactory,
-                                                         std::move(videoEncoderFactory),
-                                                         std::move(videoDecoderFactory),
-                                                         nullptr,  // audio mixer
-                                                         audioProcessingModule);
+    if (!audioProcessingModule) audioProcessingModule = webrtc::AudioProcessingBuilder().Create();
+
+    std::unique_ptr<cricket::MediaEngineInterface> media_engine =
+        cricket::WebRtcMediaEngineFactory::Create(audioDeviceModule,
+                                                  audioEncoderFactory,
+                                                  audioDecoderFactory,
+                                                  std::move(videoEncoderFactory),
+                                                  std::move(videoDecoderFactory),
+                                                  nullptr,  // audio mixer
+                                                  audioProcessingModule);
+
+    std::unique_ptr<webrtc::CallFactoryInterface> call_factory = webrtc::CreateCallFactory();
+
+    std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory =
+        webrtc::CreateRtcEventLogFactory();
+    webrtc::PeerConnectionFactoryDependencies dependencies;
+    dependencies.network_thread = _networkThread.get();
+    dependencies.worker_thread = _workerThread.get();
+    dependencies.signaling_thread = _signalingThread.get();
+    dependencies.media_engine = std::move(media_engine);
+    dependencies.call_factory = std::move(call_factory);
+    dependencies.event_log_factory = std::move(event_log_factory);
+    dependencies.media_transport_factory = std::move(mediaTransportFactory);
+    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
   }
   return self;
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
index a26a639e28..0adaa30e4f 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
@@ -13,6 +13,7 @@
 
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/media_transport_interface.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -25,6 +26,7 @@
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
   rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
   rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
+  std::unique_ptr<webrtc::MediaTransportFactory> _mediaTransportFactory;
 }
 
 + (RTCPeerConnectionFactoryBuilder *)builder {
@@ -38,7 +40,8 @@
                           nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
                           nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
                                   audioDeviceModule:_audioDeviceModule
-                              audioProcessingModule:_audioProcessingModule];
+                              audioProcessingModule:_audioProcessingModule
+                              mediaTransportFactory:std::move(_mediaTransportFactory)];
 }
 
 - (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
diff --git a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
index 9bb5179521..5f889a699d 100644
--- a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
+++ b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
@@ -22,6 +22,7 @@ extern "C" {
 
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/media_transport_interface.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -49,7 +50,8 @@ extern "C" {
               nativeVideoEncoderFactory:nullptr
               nativeVideoDecoderFactory:nullptr
                       audioDeviceModule:nullptr
-                  audioProcessingModule:nullptr]);
+                  audioProcessingModule:nullptr
+                  mediaTransportFactory:nullptr]);
 #endif
   RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
   RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];
@@ -69,7 +71,8 @@ extern "C" {
               nativeVideoEncoderFactory:nullptr
               nativeVideoDecoderFactory:nullptr
                       audioDeviceModule:nullptr
-                  audioProcessingModule:nullptr]);
+                  audioProcessingModule:nullptr
+                  mediaTransportFactory:nullptr]);
 #endif
   RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
   RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];