diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 6c7912d5cf..f3810de6fd 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -198,6 +198,7 @@
             'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
             'rtp_rtcp/source/rtp_packet_history_unittest.cc',
             'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
+            'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
             'rtp_rtcp/source/rtp_utility_unittest.cc',
             'rtp_rtcp/source/rtp_header_extension_unittest.cc',
             'rtp_rtcp/source/rtp_sender_unittest.cc',
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 82bfdbbe80..7962e97fdf 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -1864,7 +1864,6 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
       if (_IJ && !statisticians.empty()) {
         rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
       }
-      _lastRTCPTime[0] = Clock::NtpToMs(NTPsec, NTPfrac);
     }
   }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
new file mode 100644
index 0000000000..f9add41840
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+
+namespace webrtc {
+namespace {
+
+class SendTransport : public Transport,
+                      public NullRtpData {
+ public:
+  SendTransport() : rtp_rtcp_impl_(NULL), clock_(NULL), delay_ms_(0) {}
+
+  void SetRtpRtcpModule(ModuleRtpRtcpImpl* rtp_rtcp_impl) {
+    rtp_rtcp_impl_ = rtp_rtcp_impl;
+  }
+  void SimulateNetworkDelay(int delay_ms, SimulatedClock* clock) {
+    clock_ = clock;
+    delay_ms_ = delay_ms;
+  }
+  virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
+    return -1;
+  }
+  virtual int SendRTCPPacket(int /*ch*/, const void *data, int len) {
+    if (clock_) {
+      clock_->AdvanceTimeMilliseconds(delay_ms_);
+    }
+    EXPECT_TRUE(rtp_rtcp_impl_ != NULL);
+    EXPECT_EQ(0, rtp_rtcp_impl_->IncomingRtcpPacket(
+        static_cast<const uint8_t*>(data), len));
+    return len;
+  }
+  ModuleRtpRtcpImpl* rtp_rtcp_impl_;
+  SimulatedClock* clock_;
+  int delay_ms_;
+};
+}  // namespace
+
+class RtpRtcpImplTest : public ::testing::Test {
+ protected:
+  RtpRtcpImplTest()
+      : clock_(1335900000),
+        receive_statistics_(ReceiveStatistics::Create(&clock_)) {
+    RtpRtcp::Configuration configuration;
+    configuration.id = 0;
+    configuration.audio = false;
+    configuration.clock = &clock_;
+    configuration.outgoing_transport = &transport_;
+    configuration.receive_statistics = receive_statistics_.get();
+
+    rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
+    transport_.SetRtpRtcpModule(rtp_rtcp_impl_.get());
+  }
+
+  SimulatedClock clock_;
+  scoped_ptr<ReceiveStatistics> receive_statistics_;
+  scoped_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
+  SendTransport transport_;
+};
+
+TEST_F(RtpRtcpImplTest, Rtt) {
+  const uint32_t kSsrc = 0x12345;
+  RTPHeader header = {};
+  header.timestamp = 1;
+  header.sequenceNumber = 123;
+  header.ssrc = kSsrc;
+  header.headerLength = 12;
+  receive_statistics_->IncomingPacket(header, 100, false);
+
+  rtp_rtcp_impl_->SetRemoteSSRC(kSsrc);
+  EXPECT_EQ(0, rtp_rtcp_impl_->SetSendingStatus(true));
+  EXPECT_EQ(0, rtp_rtcp_impl_->SetRTCPStatus(kRtcpCompound));
+  EXPECT_EQ(0, rtp_rtcp_impl_->SetSSRC(kSsrc));
+
+  // A SR should have been sent and received.
+  EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport));
+
+  // Send new SR. A response to the last SR should be sent.
+  clock_.AdvanceTimeMilliseconds(1000);
+  transport_.SimulateNetworkDelay(100, &clock_);
+  EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport));
+
+  // Verify RTT.
+  uint16_t rtt;
+  uint16_t avg_rtt;
+  uint16_t min_rtt;
+  uint16_t max_rtt;
+  EXPECT_EQ(0, rtp_rtcp_impl_->RTT(kSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
+  EXPECT_EQ(100, rtt);
+  EXPECT_EQ(100, avg_rtt);
+  EXPECT_EQ(100, min_rtt);
+  EXPECT_EQ(100, max_rtt);
+
+  // No RTT from other ssrc.
+  EXPECT_EQ(-1,
+      rtp_rtcp_impl_->RTT(kSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
+}
+
+}  // namespace webrtc