diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 51471ba3d..f8687a552 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -335,6 +335,9 @@ class RtpRtcp : public Module {
         FrameCountObserver* observer) = 0;
     virtual FrameCountObserver* GetSendFrameCountObserver() const = 0;
 
+    virtual bool GetSendSideDelay(int* avg_send_delay_ms,
+                                  int* max_send_delay_ms) const = 0;
+
     /**************************************************************************
     *
     *   RTCP
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 9ca48975a..293412b6f 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -116,6 +116,8 @@ class MockRtpRtcp : public RtpRtcp {
            bool retransmission));
   MOCK_METHOD1(TimeToSendPadding,
       int(int bytes));
+  MOCK_CONST_METHOD2(GetSendSideDelay,
+      bool(int* avg_send_delay_ms, int* max_send_delay_ms));
   MOCK_METHOD3(RegisterRtcpObservers,
       void(RtcpIntraFrameObserver* intraFrameCallback,
            RtcpBandwidthObserver* bandwidthCallback,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4af8cde5f..00e34125c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -732,6 +732,18 @@ int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) {
   return 0;
 }
 
+bool ModuleRtpRtcpImpl::GetSendSideDelay(int* avg_send_delay_ms,
+                                         int* max_send_delay_ms) const {
+  assert(avg_send_delay_ms);
+  assert(max_send_delay_ms);
+
+  if (!child_modules_.empty()) {
+    // This API is only supported for child modules.
+    return false;
+  }
+  return rtp_sender_.GetSendSideDelay(avg_send_delay_ms, max_send_delay_ms);
+}
+
 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const {
   WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "MaxPayloadLength()");
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 0dd25492d..ae070f925 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -134,6 +134,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Returns the number of padding bytes actually sent, which can be more or
   // less than |bytes|.
   virtual int TimeToSendPadding(int bytes) OVERRIDE;
+
+  virtual bool GetSendSideDelay(int* avg_send_delay_ms,
+                                int* max_send_delay_ms) const OVERRIDE;
+
   // RTCP part.
 
   // Get RTCP status.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index c7179646b..787f8717f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -22,6 +22,7 @@ namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 const int kMaxPaddingLength = 224;
+const int kSendSideDelayWindowMs = 1000;
 
 namespace {
 
@@ -127,6 +128,23 @@ uint32_t RTPSender::NackOverheadRate() const {
   return nack_bitrate_.BitrateLast();
 }
 
+bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
+                                 int* max_send_delay_ms) const {
+  CriticalSectionScoped cs(statistics_crit_.get());
+  SendDelayMap::const_iterator it = send_delays_.upper_bound(
+      clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
+  if (!sending_media_ || it == send_delays_.end())
+    return false;
+  int num_delays = 0;
+  for (; it != send_delays_.end(); ++it) {
+    *max_send_delay_ms = std::max(*max_send_delay_ms, it->second);
+    *avg_send_delay_ms += it->second;
+    ++num_delays;
+  }
+  *avg_send_delay_ms = (*avg_send_delay_ms + num_delays / 2) / num_delays;
+  return true;
+}
+
 int32_t RTPSender::SetTransmissionTimeOffset(
     const int32_t transmission_time_offset) {
   if (transmission_time_offset > (0x800000 - 1) ||
@@ -756,6 +774,9 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
     // Packet cannot be found. Allow sending to continue.
     return true;
   }
+  if (!retransmission && capture_time_ms > 0) {
+    UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
+  }
   return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
                               retransmission && (rtx_ & kRtxRetransmitted) > 0);
 }
@@ -871,12 +892,22 @@ int32_t RTPSender::SendToNetwork(
       return 0;
     }
   }
+  if (capture_time_ms > 0) {
+    UpdateDelayStatistics(capture_time_ms, now_ms);
+  }
   if (SendPacketToNetwork(buffer, payload_length + rtp_header_length)) {
     return 0;
   }
   return -1;
 }
 
+void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
+  CriticalSectionScoped cs(statistics_crit_.get());
+  send_delays_[now_ms] = now_ms - capture_time_ms;
+  send_delays_.erase(send_delays_.begin(),
+                     send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs));
+}
+
 void RTPSender::ProcessBitrate() {
   CriticalSectionScoped cs(send_critsect_);
   Bitrate::Process();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index cd8cd2180..becd22ad1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -78,6 +78,10 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
+  // Returns true if the statistics have been calculated, and false if no frame
+  // was sent within the statistics window.
+  bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const;
+
   void SetTargetSendBitrate(const uint32_t bits);
 
   virtual uint16_t MaxDataPayloadLength() const
@@ -272,6 +276,11 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
                            RtpVideoCodecTypes *video_type);
 
  private:
+  // Maps capture time in milliseconds to send-side delay in milliseconds.
+  // Send-side delay is the difference between transmission time and capture
+  // time.
+  typedef std::map<int64_t, int> SendDelayMap;
+
   int CreateRTPHeader(uint8_t* header, int8_t payload_type,
                       uint32_t ssrc, bool marker_bit,
                       uint32_t timestamp, uint16_t sequence_number,
@@ -296,6 +305,8 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
 
   bool SendPacketToNetwork(const uint8_t *packet, uint32_t size);
 
+  void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
+
   int32_t id_;
   const bool audio_configured_;
   RTPSenderAudio *audio_;
@@ -329,6 +340,7 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
   scoped_ptr<CriticalSectionWrapper> statistics_crit_;
   uint32_t packets_sent_;
   uint32_t payload_bytes_sent_;
+  SendDelayMap send_delays_;
 
   // RTP variables
   bool start_time_stamp_forced_;
diff --git a/webrtc/video_engine/include/vie_codec.h b/webrtc/video_engine/include/vie_codec.h
index aaf017d2b..f658aaa24 100644
--- a/webrtc/video_engine/include/vie_codec.h
+++ b/webrtc/video_engine/include/vie_codec.h
@@ -198,6 +198,10 @@ class WEBRTC_DLLEXPORT ViECodec {
   // This is under development; not tested.
   virtual void SuspendBelowMinBitrate(int video_channel) = 0;
 
+  // TODO(holmer): Remove this default implementation when possible.
+  virtual bool GetSendSideDelay(int video_channel, int* avg_delay_ms,
+                                int* max_delay_ms) const { return false; }
+
  protected:
   ViECodec() {}
   virtual ~ViECodec() {}
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
index 0ebfbecbe..773192ec5 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -258,10 +258,16 @@ void ViEAutoTest::ViECodecStandardTest() {
   }
   AutoTestSleep(kAutoTestSleepTimeMs);
 
-  // Verify the delay estimate is larger than 0.
-  int delay_ms = 0;
-  EXPECT_EQ(0, codec->GetReceiveSideDelay(video_channel, &delay_ms));
-  EXPECT_GT(delay_ms, 0);
+  // Verify the delay estimates are larger than 0.
+  int avg_send_delay = 0;
+  int max_send_delay = 0;
+  EXPECT_TRUE(codec->GetSendSideDelay(video_channel, &avg_send_delay,
+                                      &max_send_delay));
+  EXPECT_GT(avg_send_delay, 0);
+  EXPECT_GE(max_send_delay, avg_send_delay);
+  int receive_delay_ms = 0;
+  EXPECT_EQ(0, codec->GetReceiveSideDelay(video_channel, &receive_delay_ms));
+  EXPECT_GT(receive_delay_ms, 0);
 
   EXPECT_EQ(0, base->StopSend(video_channel));
   EXPECT_EQ(0, codec->DeregisterEncoderObserver(video_channel));
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index eaa60447a..8b71a198c 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -1374,6 +1374,40 @@ void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
   }
 }
 
+bool ViEChannel::GetSendSideDelay(int* avg_send_delay,
+                                  int* max_send_delay) const {
+  WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+               __FUNCTION__);
+
+  *avg_send_delay = 0;
+  *max_send_delay = 0;
+  bool valid_estimate = false;
+  int num_send_delays = 0;
+  if (rtp_rtcp_->GetSendSideDelay(avg_send_delay, max_send_delay)) {
+    ++num_send_delays;
+    valid_estimate = true;
+  }
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+       it != simulcast_rtp_rtcp_.end(); it++) {
+    RtpRtcp* rtp_rtcp = *it;
+    int sub_stream_avg_delay = 0;
+    int sub_stream_max_delay = 0;
+    if (rtp_rtcp->GetSendSideDelay(&sub_stream_avg_delay,
+                                   &sub_stream_max_delay)) {
+      *avg_send_delay += sub_stream_avg_delay;
+      *max_send_delay = std::max(*max_send_delay, sub_stream_max_delay);
+      ++num_send_delays;
+    }
+  }
+  if (num_send_delays > 0) {
+    valid_estimate = true;
+    *avg_send_delay = *avg_send_delay / num_send_delays;
+    *avg_send_delay = (*avg_send_delay + num_send_delays / 2) / num_send_delays;
+  }
+  return valid_estimate;
+}
+
 void ViEChannel::GetEstimatedReceiveBandwidth(
     uint32_t* estimated_bandwidth) const {
   vie_receiver_.EstimatedReceiveBandwidth(estimated_bandwidth);
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index a17bb2b92..35981fcca 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -194,6 +194,7 @@ class ViEChannel
                          uint32_t* video_bitrate_sent,
                          uint32_t* fec_bitrate_sent,
                          uint32_t* nackBitrateSent) const;
+  bool GetSendSideDelay(int* avg_send_delay, int* max_send_delay) const;
   void GetEstimatedReceiveBandwidth(uint32_t* estimated_bandwidth) const;
 
   int32_t StartRTPDump(const char file_nameUTF8[1024],
diff --git a/webrtc/video_engine/vie_codec_impl.cc b/webrtc/video_engine/vie_codec_impl.cc
index 364862fe2..b46eb88d3 100644
--- a/webrtc/video_engine/vie_codec_impl.cc
+++ b/webrtc/video_engine/vie_codec_impl.cc
@@ -738,6 +738,20 @@ void ViECodecImpl::SuspendBelowMinBitrate(int video_channel) {
   vie_channel->SetTransmissionSmoothingStatus(true);
 }
 
+bool ViECodecImpl::GetSendSideDelay(int video_channel, int* avg_delay_ms,
+                                   int* max_delay_ms) const {
+  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: No channel %d", __FUNCTION__, video_channel);
+    shared_data_->SetLastError(kViECodecInvalidChannelId);
+    return false;
+  }
+  return vie_channel->GetSendSideDelay(avg_delay_ms, max_delay_ms);
+}
+
 bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
   // Check pl_name matches codec_type.
   if (video_codec.codecType == kVideoCodecRED) {
diff --git a/webrtc/video_engine/vie_codec_impl.h b/webrtc/video_engine/vie_codec_impl.h
index 372ffc997..cee3ed2b2 100644
--- a/webrtc/video_engine/vie_codec_impl.h
+++ b/webrtc/video_engine/vie_codec_impl.h
@@ -71,6 +71,8 @@ class ViECodecImpl
                                   const char* file_name_utf8);
   virtual int StopDebugRecording(int video_channel);
   virtual void SuspendBelowMinBitrate(int video_channel);
+  virtual bool GetSendSideDelay(int video_channel, int* avg_delay_ms,
+                                int* max_delay_ms) const;
 
  protected:
   explicit ViECodecImpl(ViESharedData* shared_data);