diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index 40dde2aed3..c583fb4c8d 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -23,6 +23,10 @@
 namespace webrtc {
 
 MediaTransportSettings::MediaTransportSettings() = default;
+MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
+    default;
+MediaTransportSettings& MediaTransportSettings::operator=(
+    const MediaTransportSettings&) = default;
 MediaTransportSettings::~MediaTransportSettings() = default;
 
 MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
@@ -84,14 +88,16 @@ MediaTransportFactory::CreateMediaTransport(
     rtc::PacketTransportInternal* packet_transport,
     rtc::Thread* network_thread,
     bool is_caller) {
-  return std::unique_ptr<MediaTransportInterface>(nullptr);
+  MediaTransportSettings settings;
+  settings.is_caller = is_caller;
+  return CreateMediaTransport(packet_transport, network_thread, settings);
 }
 
 RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
 MediaTransportFactory::CreateMediaTransport(
     rtc::PacketTransportInternal* packet_transport,
     rtc::Thread* network_thread,
-    const MediaTransportSettings settings) {
+    const MediaTransportSettings& settings) {
   return std::unique_ptr<MediaTransportInterface>(nullptr);
 }
 
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index 0e6a3fc270..6580370a0e 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -39,6 +39,8 @@ namespace webrtc {
 // A collection of settings for creation of media transport.
 struct MediaTransportSettings final {
   MediaTransportSettings();
+  MediaTransportSettings(const MediaTransportSettings&);
+  MediaTransportSettings& operator=(const MediaTransportSettings&);
   ~MediaTransportSettings();
 
   // Group calls are not currently supported, in 1:1 call one side must set
@@ -280,7 +282,7 @@ class MediaTransportFactory {
   virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
   CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
                        rtc::Thread* network_thread,
-                       const MediaTransportSettings settings);
+                       const MediaTransportSettings& settings);
 };
 
 }  // namespace webrtc