diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index b0b37113b8..b49a5d3dca 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -322,6 +322,7 @@
                 'rtp_rtcp/source/rtcp_packet/fir_unittest.cc',
                 'rtp_rtcp/source/rtcp_packet/nack_unittest.cc',
                 'rtp_rtcp/source/rtcp_packet/pli_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc',
                 'rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc',
                 'rtp_rtcp/source/rtcp_packet/remb_unittest.cc',
                 'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc',
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 55f38867b1..511f69de6d 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -66,6 +66,8 @@ source_set("rtp_rtcp") {
     "source/rtcp_packet/pli.h",
     "source/rtcp_packet/psfb.cc",
     "source/rtcp_packet/psfb.h",
+    "source/rtcp_packet/rapid_resync_request.cc",
+    "source/rtcp_packet/rapid_resync_request.h",
     "source/rtcp_packet/receiver_report.cc",
     "source/rtcp_packet/receiver_report.h",
     "source/rtcp_packet/remb.cc",
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index bc2f8c34f9..f72e71030a 100644
--- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -61,6 +61,8 @@
         'source/rtcp_packet/pli.h',
         'source/rtcp_packet/psfb.cc',
         'source/rtcp_packet/psfb.h',
+        'source/rtcp_packet/rapid_resync_request.cc',
+        'source/rtcp_packet/rapid_resync_request.h',
         'source/rtcp_packet/receiver_report.cc',
         'source/rtcp_packet/receiver_report.h',
         'source/rtcp_packet/remb.cc',
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc
new file mode 100644
index 0000000000..394b3cf543
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2016 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 "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+// RFC 4585: Feedback format.
+// Rapid Resynchronisation Request (draft-perkins-avt-rapid-rtp-sync-03).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|  FMT=5  |     PT=205    |         length=2              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool RapidResyncRequest::Parse(const RtcpCommonHeader& header,
+                               const uint8_t* payload) {
+  RTC_CHECK(header.packet_type == kPacketType);
+  RTC_CHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes != kCommonFeedbackLength) {
+    LOG(LS_WARNING) << "Packet payload size should be " << kCommonFeedbackLength
+                    << " instead of " << header.payload_size_bytes
+                    << " to be a valid Rapid Resynchronisation Request";
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+  return true;
+}
+
+bool RapidResyncRequest::Create(
+    uint8_t* packet,
+    size_t* index,
+    size_t max_length,
+    RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
new file mode 100644
index 0000000000..866eb2ce52
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+// draft-perkins-avt-rapid-rtp-sync-03
+class RapidResyncRequest : public Rtpfb {
+ public:
+  static const uint8_t kFeedbackMessageType = 5;
+
+  RapidResyncRequest() {}
+  ~RapidResyncRequest() override {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    return kHeaderLength + kCommonFeedbackLength;
+  }
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RapidResyncRequest);
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
new file mode 100644
index 0000000000..896933d51d
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2016 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 "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+using webrtc::rtcp::RapidResyncRequest;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created packet matching constants above.
+const uint8_t kPacket[] = {0x85, 205,  0x00, 0x02,
+                           0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89};
+const size_t kPacketLength = sizeof(kPacket);
+}  // namespace
+
+TEST(RtcpPacketRapidResyncRequestTest, Parse) {
+  RtcpCommonHeader header;
+  ASSERT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  RapidResyncRequest mutable_parsed;
+  EXPECT_TRUE(mutable_parsed.Parse(
+      header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const RapidResyncRequest& parsed = mutable_parsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, Create) {
+  RapidResyncRequest rrr;
+  rrr.From(kSenderSsrc);
+  rrr.To(kRemoteSsrc);
+
+  rtc::scoped_ptr<RawPacket> packet = rrr.Build();
+
+  EXPECT_THAT(make_tuple(packet->Buffer(), packet->Length()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnWrongSizePacket) {
+  RapidResyncRequest parsed;
+  RtcpCommonHeader header;
+  ASSERT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  const size_t kCorrectPayloadSize = header.payload_size_bytes;
+  const uint8_t* payload = kPacket + RtcpCommonHeader::kHeaderSizeBytes;
+
+  header.payload_size_bytes = kCorrectPayloadSize - 1;
+  EXPECT_FALSE(parsed.Parse(header, payload));
+
+  header.payload_size_bytes = kCorrectPayloadSize + 1;
+  EXPECT_FALSE(parsed.Parse(header, payload));
+}
+}  // namespace webrtc