diff --git a/api/BUILD.gn b/api/BUILD.gn
index 7b8ae83ba3..2359ebc99f 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -133,6 +133,7 @@ rtc_static_library("libjingle_peerconnection_api") {
     "audio_codecs:audio_codecs_api",
     "transport:bitrate_settings",
     "transport:network_control",
+    "units:data_rate",
     "video:encoded_image",
     "video:video_frame",
     "//third_party/abseil-cpp/absl/strings",
diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index c1a83bb2be..eccb74d79c 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -188,4 +188,7 @@ size_t MediaTransportInterface::GetAudioPacketOverhead() const {
   return 0;
 }
 
+void MediaTransportInterface::SetAllocatedBitrateLimits(
+    const MediaTransportAllocatedBitrateLimits& limits) {}
+
 }  // namespace webrtc
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index a440860ffc..a46c99446e 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -26,6 +26,7 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/rtc_error.h"
+#include "api/units/data_rate.h"
 #include "api/video/encoded_image.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/deprecation.h"
@@ -49,6 +50,12 @@ class AudioPacketReceivedObserver {
   virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
 };
 
+struct MediaTransportAllocatedBitrateLimits {
+  DataRate min_pacing_rate = DataRate::Zero();
+  DataRate max_padding_bitrate = DataRate::Zero();
+  DataRate max_total_allocated_bitrate = DataRate::Zero();
+};
+
 // A collection of settings for creation of media transport.
 struct MediaTransportSettings final {
   MediaTransportSettings();
@@ -449,6 +456,11 @@ class MediaTransportInterface {
   virtual void SetMediaTransportStateCallback(
       MediaTransportStateCallback* callback) = 0;
 
+  // Updates allocation limits.
+  // TODO(psla): Make abstract when downstream implementation implement it.
+  virtual void SetAllocatedBitrateLimits(
+      const MediaTransportAllocatedBitrateLimits& limits);
+
   // Sends a data buffer to the remote endpoint using the given send parameters.
   // |buffer| may not be larger than 256 KiB. Returns an error if the send
   // fails.
diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index eb6ec93ae2..bc8a320f3d 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -93,6 +93,9 @@ class FakeMediaTransport : public MediaTransportInterface {
     }
   }
 
+  void SetAllocatedBitrateLimits(
+      const MediaTransportAllocatedBitrateLimits& limits) override {}
+
   int target_rate_observers_size() { return target_rate_observers_.size(); }
 
  private:
diff --git a/api/test/loopback_media_transport.cc b/api/test/loopback_media_transport.cc
index 57cc36d81f..e7ccb0afd8 100644
--- a/api/test/loopback_media_transport.cc
+++ b/api/test/loopback_media_transport.cc
@@ -81,6 +81,9 @@ class WrapperMediaTransport : public MediaTransportInterface {
     wrapped_->SetDataSink(sink);
   }
 
+  void SetAllocatedBitrateLimits(
+      const MediaTransportAllocatedBitrateLimits& limits) override {}
+
  private:
   MediaTransportInterface* wrapped_;
 };
@@ -372,4 +375,8 @@ void MediaTransportPair::LoopbackMediaTransport::OnStateChanged() {
     state_callback_->OnStateChanged(state_);
   }
 }
+
+void MediaTransportPair::LoopbackMediaTransport::SetAllocatedBitrateLimits(
+    const MediaTransportAllocatedBitrateLimits& limits) {}
+
 }  // namespace webrtc
diff --git a/api/test/loopback_media_transport.h b/api/test/loopback_media_transport.h
index 64b8ced24b..bcfdb63fd5 100644
--- a/api/test/loopback_media_transport.h
+++ b/api/test/loopback_media_transport.h
@@ -127,6 +127,9 @@ class MediaTransportPair {
 
     Stats GetStats();
 
+    void SetAllocatedBitrateLimits(
+        const MediaTransportAllocatedBitrateLimits& limits) override;
+
    private:
     void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame);
 
diff --git a/call/call.cc b/call/call.cc
index 2361c72c08..f3f6cd314b 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1180,6 +1180,17 @@ void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
   transport_send_ptr_->SetAllocatedSendBitrateLimits(
       min_send_bitrate_bps, max_padding_bitrate_bps, total_bitrate_bps);
 
+  {
+    rtc::CritScope lock(&target_observer_crit_);
+    if (media_transport_) {
+      MediaTransportAllocatedBitrateLimits limits;
+      limits.min_pacing_rate = DataRate::bps(min_send_bitrate_bps);
+      limits.max_padding_bitrate = DataRate::bps(max_padding_bitrate_bps);
+      limits.max_total_allocated_bitrate = DataRate::bps(total_bitrate_bps);
+      media_transport_->SetAllocatedBitrateLimits(limits);
+    }
+  }
+
   rtc::CritScope lock(&bitrate_crit_);
   min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;
   configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps;