diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 00e374c4c3..4409e8e398 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -418,12 +418,6 @@ class RtpRtcp : public Module {
     */
     virtual int32_t ResetRTT(const uint32_t remoteSSRC)= 0 ;
 
-    /*
-     * Sets the estimated RTT, to be used for receive only modules without
-     * possibility of calculating its own RTT.
-     */
-    virtual void SetRtt(uint32_t rtt) = 0;
-
     /*
     *   Force a send of a RTCP packet
     *   normal SR and RR are triggered via the process function
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 355d79699e..fd7b9a942f 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -145,7 +145,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const uint32_t remoteSSRC, uint16_t* RTT, uint16_t* avgRTT, uint16_t* minRTT, uint16_t* maxRTT));
   MOCK_METHOD1(ResetRTT,
       int32_t(const uint32_t remoteSSRC));
-  MOCK_METHOD1(SetRtt, void(uint32_t rtt));
   MOCK_METHOD1(SendRTCP,
       int32_t(uint32_t rtcpPacketType));
   MOCK_METHOD1(SendRTCPReferencePictureSelection,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index ddeedccd94..fb7adf6b34 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -54,8 +54,7 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock,
     _receivedInfoMap(),
     _packetTimeOutMS(0),
     _lastReceivedRrMs(0),
-    _lastIncreasedSequenceNumberMs(0),
-    _rtt(0) {
+    _lastIncreasedSequenceNumberMs(0) {
     memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
     WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
 }
@@ -228,23 +227,6 @@ bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) {
   return true;
 }
 
-uint16_t RTCPReceiver::RTT() const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-  if (!_receivedReportBlockMap.empty()) {
-    return 0;
-  }
-  return _rtt;
-}
-
-int RTCPReceiver::SetRTT(uint16_t rtt) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-  if (!_receivedReportBlockMap.empty()) {
-    return -1;
-  }
-  _rtt = rtt;
-  return 0;
-}
-
 int32_t
 RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs,
                   uint32_t *ReceivedNTPfrac,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 3da3445701..5c6f732195 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -78,10 +78,6 @@ public:
                 uint16_t* minRTT,
                 uint16_t* maxRTT) const;
 
-    uint16_t RTT() const;
-
-    int SetRTT(uint16_t rtt);
-
     int32_t ResetRTT(const uint32_t remoteSSRC);
 
     int32_t SenderInfoReceived(RTCPSenderInfo* senderInfo) const;
@@ -266,10 +262,6 @@ protected:
   // delivered RTP packet to the remote side.
   int64_t _lastIncreasedSequenceNumberMs;
 
-  // Externally set RTT. This value can only be used if there are no valid
-  // RTT estimates.
-  uint16_t _rtt;
-
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3369b55198..6caa18883e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -910,11 +910,6 @@ int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) {
   return rtcp_receiver_.ResetRTT(remote_ssrc);
 }
 
-void ModuleRtpRtcpImpl:: SetRtt(uint32_t rtt) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRtt(rtt: %u)", rtt);
-  rtcp_receiver_.SetRTT(static_cast<uint16_t>(rtt));
-}
-
 // Reset RTP data counters for the sending side.
 int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
   WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c2fc498d79..d7b2035a02 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -171,8 +171,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Reset RoundTripTime statistics.
   virtual int32_t ResetRTT(const uint32_t remote_ssrc) OVERRIDE;
 
-  virtual void SetRtt(uint32_t rtt) OVERRIDE;
-
   // Force a send of an RTCP packet.
   // Normal SR and RR are triggered via the process function.
   virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE;
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 484408fb27..c1b9cd8454 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -1711,8 +1711,6 @@ bool ViEChannel::ChannelDecodeProcess() {
 
 void ViEChannel::OnRttUpdate(uint32_t rtt) {
   vcm_.SetReceiveChannelParameters(rtt);
-  if (!sender_)
-    rtp_rtcp_->SetRtt(rtt);
 }
 
 int32_t ViEChannel::StartDecodeThread() {