diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 826ef65..da94de5 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -93,6 +93,7 @@
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
+      "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../system_wrappers",
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 0e64269..e594dcc 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -26,6 +26,7 @@
 #include "webrtc/base/location.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
@@ -97,7 +98,7 @@
              public BitrateAllocator::LimitObserver {
  public:
   Call(const Call::Config& config,
-       std::unique_ptr<RtpTransportControllerSend> transport_send);
+       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
   virtual ~Call();
 
   // Implements webrtc::Call.
@@ -298,16 +299,21 @@
 }
 
 Call* Call::Create(const Call::Config& config) {
-  return new internal::Call(
-      config, std::unique_ptr<RtpTransportControllerSend>(
-                  new RtpTransportControllerSend(Clock::GetRealTimeClock(),
-                                                 config.event_log)));
+  return new internal::Call(config,
+                            rtc::MakeUnique<RtpTransportControllerSend>(
+                                Clock::GetRealTimeClock(), config.event_log));
+}
+
+Call* Call::Create(
+    const Call::Config& config,
+    std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
+  return new internal::Call(config, std::move(transport_send));
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config,
-           std::unique_ptr<RtpTransportControllerSend> transport_send)
+           std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
@@ -342,7 +348,7 @@
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
-  transport_send->RegisterNetworkObserver(this);
+  transport_send->send_side_cc()->RegisterNetworkObserver(this);
   transport_send_ = std::move(transport_send);
   transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
   transport_send_->send_side_cc()->SetBweBitrates(
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index caf0ee2..f67b690 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -10,6 +10,7 @@
 #ifndef WEBRTC_CALL_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -20,6 +21,7 @@
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -98,6 +100,11 @@
 
   static Call* Create(const Call::Config& config);
 
+  // Allows mocking |transport_send| for testing.
+  static Call* Create(
+      const Call::Config& config,
+      std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 27d2329..564f6bd 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -11,11 +11,15 @@
 #include <list>
 #include <map>
 #include <memory>
+#include <utility>
 
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
+#include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_transport.h"
@@ -305,4 +309,28 @@
   }
 }
 
+// TODO(zstein): This is just a motivating example for
+// MockSendSideCongestionController. It should be deleted once we have more
+// meaningful tests.
+TEST(CallTest, MockSendSideCongestionControllerExample) {
+  RtcEventLogNullImpl event_log;
+  Call::Config config(&event_log);
+
+  SimulatedClock clock(123456);
+  PacketRouter packet_router;
+  testing::NiceMock<test::MockSendSideCongestionController> mock_cc(
+      &clock, &event_log, &packet_router);
+  auto transport_send =
+      rtc::MakeUnique<FakeRtpTransportControllerSend>(&mock_cc);
+  std::unique_ptr<Call> call(Call::Create(config, std::move(transport_send)));
+
+  Call::Config::BitrateConfig bitrate_config;
+  bitrate_config.min_bitrate_bps = 1;
+  bitrate_config.start_bitrate_bps = 2;
+  bitrate_config.max_bitrate_bps = 3;
+
+  EXPECT_CALL(mock_cc, SetBweBitrates(1, 2, 3));
+  call->SetBitrateConfig(bitrate_config);
+}
+
 }  // namespace webrtc
diff --git a/webrtc/call/fake_rtp_transport_controller_send.h b/webrtc/call/fake_rtp_transport_controller_send.h
new file mode 100644
index 0000000..42462d9
--- /dev/null
+++ b/webrtc/call/fake_rtp_transport_controller_send.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 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
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#define WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
+
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
+#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
+#include "webrtc/modules/pacing/packet_router.h"
+
+namespace webrtc {
+
+class FakeRtpTransportControllerSend
+    : public RtpTransportControllerSendInterface {
+ public:
+  explicit FakeRtpTransportControllerSend(
+      SendSideCongestionController* send_side_cc)
+      : send_side_cc_(send_side_cc) {
+    RTC_DCHECK(send_side_cc);
+  }
+
+  PacketRouter* packet_router() override { return &packet_router_; }
+
+  SendSideCongestionController* send_side_cc() override {
+    return send_side_cc_;
+  }
+
+  TransportFeedbackObserver* transport_feedback_observer() override {
+    return send_side_cc_;
+  }
+
+  RtpPacketSender* packet_sender() override { return send_side_cc_->pacer(); }
+
+ private:
+  PacketRouter packet_router_;
+  SendSideCongestionController* send_side_cc_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
diff --git a/webrtc/call/rtp_transport_controller_send.cc b/webrtc/call/rtp_transport_controller_send.cc
index a006f28..1040218 100644
--- a/webrtc/call/rtp_transport_controller_send.cc
+++ b/webrtc/call/rtp_transport_controller_send.cc
@@ -18,10 +18,4 @@
     : send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) {
 }
 
-void RtpTransportControllerSend::RegisterNetworkObserver(
-    SendSideCongestionController::Observer* observer) {
-  // Must be called only once.
-  send_side_cc_.RegisterNetworkObserver(observer);
-}
-
 }  // namespace webrtc
diff --git a/webrtc/call/rtp_transport_controller_send.h b/webrtc/call/rtp_transport_controller_send.h
index 2bbb547..b626fa8 100644
--- a/webrtc/call/rtp_transport_controller_send.h
+++ b/webrtc/call/rtp_transport_controller_send.h
@@ -26,9 +26,6 @@
  public:
   RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
 
-  void RegisterNetworkObserver(
-      SendSideCongestionController::Observer* observer);
-
   // Implements RtpTransportControllerSendInterface
   PacketRouter* packet_router() override { return &packet_router_; }
   SendSideCongestionController* send_side_cc() override {