diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index e4fbede278..40dde2aed3 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -22,6 +22,9 @@
 
 namespace webrtc {
 
+MediaTransportSettings::MediaTransportSettings() = default;
+MediaTransportSettings::~MediaTransportSettings() = default;
+
 MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
 
 MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
@@ -76,4 +79,20 @@ MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
 MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
     MediaTransportEncodedVideoFrame&&) = default;
 
+RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
+MediaTransportFactory::CreateMediaTransport(
+    rtc::PacketTransportInternal* packet_transport,
+    rtc::Thread* network_thread,
+    bool is_caller) {
+  return std::unique_ptr<MediaTransportInterface>(nullptr);
+}
+
+RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
+MediaTransportFactory::CreateMediaTransport(
+    rtc::PacketTransportInternal* packet_transport,
+    rtc::Thread* network_thread,
+    const MediaTransportSettings settings) {
+  return std::unique_ptr<MediaTransportInterface>(nullptr);
+}
+
 }  // namespace webrtc
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index fc0922cdc1..b388ac9218 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -18,9 +18,11 @@
 #define API_MEDIA_TRANSPORT_INTERFACE_H_
 
 #include <memory>
+#include <string>
 #include <utility>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/rtcerror.h"
 #include "api/video/encoded_image.h"
@@ -33,6 +35,19 @@ class Thread;
 
 namespace webrtc {
 
+// A collection of settings for creation of media transport.
+struct MediaTransportSettings final {
+  MediaTransportSettings();
+  ~MediaTransportSettings();
+
+  // Group calls are not currently supported, in 1:1 call one side must set
+  // is_caller = true and another is_caller = false.
+  bool is_caller;
+
+  // Must be set if a pre-shared key is used for the call.
+  absl::optional<std::string> pre_shared_key;
+};
+
 // 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 final {
@@ -244,10 +259,21 @@ class MediaTransportFactory {
   // - Does not take ownership of packet_transport or network_thread.
   // - Does not support group calls, in 1:1 call one side must set
   //   is_caller = true and another is_caller = false.
+  // TODO(bugs.webrtc.org/9938) This constructor will be removed and replaced
+  // with the one below.
   virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
   CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
                        rtc::Thread* network_thread,
-                       bool is_caller) = 0;
+                       bool is_caller);
+
+  // Creates media transport.
+  // - Does not take ownership of packet_transport or network_thread.
+  // TODO(bugs.webrtc.org/9938): remove default implementation once all children
+  // override it.
+  virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
+  CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
+                       rtc::Thread* network_thread,
+                       const MediaTransportSettings settings);
 };
 
 }  // namespace webrtc