diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 38c1c04..ee5fe13 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -93,6 +93,7 @@
             '<(webrtc_root)/test/test.gyp:frame_generator',
             '<(webrtc_root)/test/test.gyp:rtp_test_utils',
             '<(webrtc_root)/test/test.gyp:test_support_main',
+            '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
             '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
           ],
           'sources': [
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 3f0b047..9652a53 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -18,6 +18,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/null_transport.h"
 #include "webrtc/typedefs.h"
 
 namespace {
@@ -79,6 +80,7 @@
   RTCPSender* rtcp_sender_;
   RTCPReceiver* rtcp_receiver_;
   TestTransport* test_transport_;
+  test::NullTransport null_transport_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
   rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 };
@@ -88,14 +90,13 @@
   configuration.audio = false;
   configuration.clock = system_clock_;
   configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
+  configuration.outgoing_transport = &null_transport_;
   dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
-  rtcp_sender_ =
-      new RTCPSender(false, system_clock_, receive_statistics_.get(), nullptr);
   rtcp_receiver_ = new RTCPReceiver(system_clock_, false, nullptr, nullptr,
                                     nullptr, nullptr, dummy_rtp_rtcp_impl_);
   test_transport_ = new TestTransport(rtcp_receiver_);
-
-  EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
+  rtcp_sender_ = new RTCPSender(false, system_clock_, receive_statistics_.get(),
+                                nullptr, test_transport_);
 }
 
 void RtcpFormatRembTest::TearDown() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 111757b..f250e29 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -135,13 +135,12 @@
     bool audio,
     Clock* clock,
     ReceiveStatistics* receive_statistics,
-    RtcpPacketTypeCounterObserver* packet_type_counter_observer)
+    RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+    Transport* outgoing_transport)
     : audio_(audio),
       clock_(clock),
       method_(kRtcpOff),
-      critical_section_transport_(
-          CriticalSectionWrapper::CreateCriticalSection()),
-      cbTransport_(nullptr),
+      transport_(outgoing_transport),
 
       critical_section_rtcp_sender_(
           CriticalSectionWrapper::CreateCriticalSection()),
@@ -173,6 +172,7 @@
       packet_type_counter_observer_(packet_type_counter_observer) {
   memset(last_send_report_, 0, sizeof(last_send_report_));
   memset(last_rtcp_time_, 0, sizeof(last_rtcp_time_));
+  RTC_DCHECK(transport_ != nullptr);
 
   builders_[kRtcpSr] = &RTCPSender::BuildSR;
   builders_[kRtcpRr] = &RTCPSender::BuildRR;
@@ -196,12 +196,6 @@
 RTCPSender::~RTCPSender() {
 }
 
-int32_t RTCPSender::RegisterSendTransport(Transport* outgoingTransport) {
-  CriticalSectionScoped lock(critical_section_transport_.get());
-  cbTransport_ = outgoingTransport;
-  return 0;
-}
-
 RTCPMethod RTCPSender::Status() const {
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   return method_;
@@ -1115,11 +1109,8 @@
 }
 
 int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
-  CriticalSectionScoped lock(critical_section_transport_.get());
-  if (cbTransport_) {
-    if (cbTransport_->SendRtcp(dataBuffer, length))
-      return 0;
-  }
+  if (transport_->SendRtcp(dataBuffer, length))
+    return 0;
   return -1;
 }
 
@@ -1210,10 +1201,6 @@
 }
 
 bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
-  CriticalSectionScoped lock(critical_section_transport_.get());
-  if (!cbTransport_)
-    return false;
-
   class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
    public:
     Sender(Transport* transport)
@@ -1226,7 +1213,7 @@
 
     Transport* const transport_;
     bool send_failure_;
-  } sender(cbTransport_);
+  } sender(transport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 2051102..2f27989 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -74,11 +74,10 @@
  RTCPSender(bool audio,
             Clock* clock,
             ReceiveStatistics* receive_statistics,
-            RtcpPacketTypeCounterObserver* packet_type_counter_observer);
+            RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+            Transport* outgoing_transport);
  virtual ~RTCPSender();
 
- int32_t RegisterSendTransport(Transport* outgoingTransport);
-
  RTCPMethod Status() const;
  void SetRTCPStatus(RTCPMethod method);
 
@@ -228,8 +227,7 @@
  Clock* const clock_;
  RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_);
 
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_transport_;
- Transport* cbTransport_ GUARDED_BY(critical_section_transport_);
+ Transport* const transport_;
 
  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
  bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 33c3f84..19dbc4d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -230,11 +230,10 @@
     configuration.outgoing_transport = &test_transport_;
 
     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
-    rtcp_sender_.reset(
-        new RTCPSender(false, &clock_, receive_statistics_.get(), nullptr));
+    rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                      nullptr, &test_transport_));
     rtcp_sender_->SetSSRC(kSenderSsrc);
     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-    EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
   }
 
   void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
@@ -667,10 +666,9 @@
 
 TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
   RtcpPacketTypeCounterObserverImpl observer;
-  rtcp_sender_.reset(
-      new RTCPSender(false, &clock_, receive_statistics_.get(), &observer));
+  rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                    &observer, &test_transport_));
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-  EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
   rtcp_sender_->SetRTCPStatus(kRtcpNonCompound);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
   EXPECT_EQ(1, parser()->pli()->num_packets());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index ae4caf7..2e65452 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -72,7 +72,8 @@
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
-                   configuration.rtcp_packet_type_counter_observer),
+                   configuration.rtcp_packet_type_counter_observer,
+                   configuration.outgoing_transport),
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
@@ -99,9 +100,6 @@
       rtt_ms_(0) {
   send_video_codec_.codecType = kVideoCodecUnknown;
 
-  // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
-  rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
-
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 1540e55..f75113a 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
+#include "webrtc/test/null_transport.h"
 
 #include <algorithm>
 #include <vector>
@@ -90,6 +91,7 @@
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock_;
+    configuration.outgoing_transport = &null_transport_;
     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
@@ -105,6 +107,7 @@
   uint16_t test_sequence_number_;
   std::vector<uint32_t> test_csrcs_;
   SimulatedClock fake_clock_;
+  test::NullTransport null_transport_;
 };
 
 TEST_F(RtpRtcpAPITest, Basic) {