diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 31080fcee..54aadf5d3 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -510,6 +510,8 @@ class RtpRtcp : public Module {
     */
     virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
 
+    virtual bool RtcpXrRrtrStatus() const = 0;
+
     /*
     *  (REMB) Receiver Estimated Max Bitrate
     */
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 09c3ca623..90ea4f848 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -170,6 +170,8 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const RTCPVoIPMetric* VoIPMetric));
   MOCK_METHOD1(SetRtcpXrRrtrStatus,
       void(bool enable));
+  MOCK_CONST_METHOD0(RtcpXrRrtrStatus,
+      bool());
   MOCK_CONST_METHOD0(REMB,
       bool());
   MOCK_METHOD1(SetREMBStatus,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 7962e97fd..a407a42c9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -2182,6 +2182,11 @@ void RTCPSender::SendRtcpXrReceiverReferenceTime(bool enable) {
   xrSendReceiverReferenceTimeEnabled_ = enable;
 }
 
+bool RTCPSender::RtcpXrReceiverReferenceTime() const {
+  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  return xrSendReceiverReferenceTimeEnabled_;
+}
+
 // called under critsect _criticalSectionRTCPSender
 int32_t RTCPSender::WriteAllReportBlocksToBuffer(
     uint8_t* rtcpbuffer,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 106a78dda..9ed582444 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -171,6 +171,8 @@ public:
 
     void SendRtcpXrReceiverReferenceTime(bool enable);
 
+    bool RtcpXrReceiverReferenceTime() const;
+
     int32_t SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize],
                      const uint8_t arrLength);
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 7f1596686..6c5aff9f5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -963,6 +963,10 @@ void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
   return rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
 }
 
+bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
+  return rtcp_sender_.RtcpXrReceiverReferenceTime();
+}
+
 int32_t ModuleRtpRtcpImpl::DataCountersRTP(
     uint32_t* bytes_sent,
     uint32_t* packets_sent) const {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 05ad6c303..940a65633 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -259,6 +259,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // (XR) Receiver reference time report.
   virtual void SetRtcpXrRrtrStatus(bool enable) OVERRIDE;
 
+  virtual bool RtcpXrRrtrStatus() const OVERRIDE;
+
   // Audio part.
 
   // Set audio packet size, used to determine when it's time to send a DTMF
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index deb428f5b..6248f4970 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -124,6 +124,12 @@ TEST_F(RtpRtcpImplTest, Rtt) {
       rtp_rtcp_impl_->RTT(kSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
 }
 
+TEST_F(RtpRtcpImplTest, SetRtcpXrRrtrStatus) {
+  EXPECT_FALSE(rtp_rtcp_impl_->RtcpXrRrtrStatus());
+  rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true);
+  EXPECT_TRUE(rtp_rtcp_impl_->RtcpXrRrtrStatus());
+}
+
 TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true);
   EXPECT_EQ(0, rtp_rtcp_impl_->SetSendingStatus(false));
diff --git a/webrtc/video_engine/include/vie_rtp_rtcp.h b/webrtc/video_engine/include/vie_rtp_rtcp.h
index 6f97ba586..7937d3552 100644
--- a/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ b/webrtc/video_engine/include/vie_rtp_rtcp.h
@@ -254,6 +254,11 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
                                                bool enable,
                                                int id) = 0;
 
+  // Enables/disables RTCP Receiver Reference Time Report Block extension/
+  // DLRR Report Block extension (RFC 3611).
+  // TODO(asapersson): Remove default implementation.
+  virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) { return -1; }
+
   // Enables transmission smoothening, i.e. packets belonging to the same frame
   // will be sent over a longer period of time instead of sending them
   // back-to-back.
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 9e26a89c6..aa86673c9 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -406,6 +406,7 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
         rtp_rtcp->DeregisterSendRtpHeaderExtension(
             kRtpExtensionAbsoluteSendTime);
       }
+      rtp_rtcp->SetRtcpXrRrtrStatus(rtp_rtcp_->RtcpXrRrtrStatus());
     }
     // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
     // modules can be deleted after this step.
@@ -922,6 +923,15 @@ bool ViEChannel::GetReceiveAbsoluteSendTimeStatus() const {
   return receive_absolute_send_time_enabled_;
 }
 
+void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
+  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+       it != simulcast_rtp_rtcp_.end(); it++) {
+    (*it)->SetRtcpXrRrtrStatus(enable);
+  }
+}
+
 void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
   assert(paced_sender_ && "No paced sender registered.");
   paced_sender_->SetStatus(enable);
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 01c60f35c..3b3714050 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -125,6 +125,7 @@ class ViEChannel
   int SetSendAbsoluteSendTimeStatus(bool enable, int id);
   int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
   bool GetReceiveAbsoluteSendTimeStatus() const;
+  void SetRtcpXrRrtrStatus(bool enable);
   void SetTransmissionSmoothingStatus(bool enable);
   int32_t EnableTMMBR(const bool enable);
   int32_t EnableKeyFrameRequestCallback(const bool enable);
diff --git a/webrtc/video_engine/vie_rtp_rtcp_impl.cc b/webrtc/video_engine/vie_rtp_rtcp_impl.cc
index f824b3c72..7d3c8ad25 100644
--- a/webrtc/video_engine/vie_rtp_rtcp_impl.cc
+++ b/webrtc/video_engine/vie_rtp_rtcp_impl.cc
@@ -804,6 +804,25 @@ int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
   return 0;
 }
 
+int ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(int video_channel, bool enable) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+               ViEId(shared_data_->instance_id(), video_channel),
+               "ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(%d, %d)",
+               video_channel, enable);
+
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(video_channel);
+  if (!vie_channel) {
+    WEBRTC_TRACE(kTraceError, kTraceVideo,
+                 ViEId(shared_data_->instance_id(), video_channel),
+                 "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  vie_channel->SetRtcpXrRrtrStatus(enable);
+  return 0;
+}
+
 int ViERTP_RTCPImpl::SetTransmissionSmoothingStatus(int video_channel,
                                                     bool enable) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
diff --git a/webrtc/video_engine/vie_rtp_rtcp_impl.h b/webrtc/video_engine/vie_rtp_rtcp_impl.h
index d7c76616c..caabd2cda 100644
--- a/webrtc/video_engine/vie_rtp_rtcp_impl.h
+++ b/webrtc/video_engine/vie_rtp_rtcp_impl.h
@@ -88,6 +88,7 @@ class ViERTP_RTCPImpl
   virtual int SetReceiveAbsoluteSendTimeStatus(int video_channel,
                                                bool enable,
                                                int id);
+  virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable);
   virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
   virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
                                               RtcpStatistics& basic_stats,