diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 2095aad..a5b0d30 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -118,6 +118,7 @@
     "source/rtp_packet.h",
     "source/rtp_packet_history.cc",
     "source/rtp_packet_history.h",
+    "source/rtp_packet_received.cc",
     "source/rtp_packet_received.h",
     "source/rtp_packet_to_send.h",
     "source/rtp_payload_registry.cc",
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 2c9d0ae..7df6005 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -16,7 +16,6 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/random.h"
@@ -145,43 +144,6 @@
   return csrcs;
 }
 
-void Packet::GetHeader(RTPHeader* header) const {
-  header->markerBit = Marker();
-  header->payloadType = PayloadType();
-  header->sequenceNumber = SequenceNumber();
-  header->timestamp = Timestamp();
-  header->ssrc = Ssrc();
-  std::vector<uint32_t> csrcs = Csrcs();
-  header->numCSRCs = csrcs.size();
-  for (size_t i = 0; i < csrcs.size(); ++i) {
-    header->arrOfCSRCs[i] = csrcs[i];
-  }
-  header->paddingLength = padding_size();
-  header->headerLength = headers_size();
-  header->payload_type_frequency = 0;
-  header->extension.hasTransmissionTimeOffset =
-      GetExtension<TransmissionOffset>(
-          &header->extension.transmissionTimeOffset);
-  header->extension.hasAbsoluteSendTime =
-      GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
-  header->extension.hasTransportSequenceNumber =
-      GetExtension<TransportSequenceNumber>(
-          &header->extension.transportSequenceNumber);
-  header->extension.hasAudioLevel = GetExtension<AudioLevel>(
-      &header->extension.voiceActivity, &header->extension.audioLevel);
-  header->extension.hasVideoRotation =
-      GetExtension<VideoOrientation>(&header->extension.videoRotation);
-  header->extension.hasVideoContentType =
-      GetExtension<VideoContentTypeExtension>(
-          &header->extension.videoContentType);
-  header->extension.has_video_timing =
-      GetExtension<VideoTimingExtension>(&header->extension.video_timing);
-  GetExtension<RtpStreamId>(&header->extension.stream_id);
-  GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
-  GetExtension<RtpMid>(&header->extension.mid);
-  GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
-}
-
 size_t Packet::headers_size() const {
   return payload_offset_;
 }
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 8797417..fb19815 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -18,7 +18,6 @@
 #include "webrtc/rtc_base/copyonwritebuffer.h"
 
 namespace webrtc {
-struct RTPHeader;
 class RtpHeaderExtensionMap;
 class Random;
 
@@ -49,10 +48,6 @@
   uint32_t Ssrc() const;
   std::vector<uint32_t> Csrcs() const;
 
-  // TODO(danilchap): Remove this function when all code update to use RtpPacket
-  // directly. Function is there just for easier backward compatibilty.
-  void GetHeader(RTPHeader* header) const;
-
   size_t headers_size() const;
 
   // Payload.
@@ -127,7 +122,7 @@
   explicit Packet(const ExtensionManager* extensions);
   Packet(const Packet&);
   Packet(const ExtensionManager* extensions, size_t capacity);
-  virtual ~Packet();
+  ~Packet();
 
   Packet& operator=(const Packet&) = default;
 
diff --git a/modules/rtp_rtcp/source/rtp_packet_received.cc b/modules/rtp_rtcp/source/rtp_packet_received.cc
new file mode 100644
index 0000000..82a4f68
--- /dev/null
+++ b/modules/rtp_rtcp/source/rtp_packet_received.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 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/rtp_packet_received.h"
+
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+namespace webrtc {
+
+void RtpPacketReceived::GetHeader(RTPHeader* header) const {
+  header->markerBit = Marker();
+  header->payloadType = PayloadType();
+  header->sequenceNumber = SequenceNumber();
+  header->timestamp = Timestamp();
+  header->ssrc = Ssrc();
+  std::vector<uint32_t> csrcs = Csrcs();
+  header->numCSRCs = csrcs.size();
+  for (size_t i = 0; i < csrcs.size(); ++i) {
+    header->arrOfCSRCs[i] = csrcs[i];
+  }
+  header->paddingLength = padding_size();
+  header->headerLength = headers_size();
+  header->payload_type_frequency = payload_type_frequency();
+  header->extension.hasTransmissionTimeOffset =
+      GetExtension<TransmissionOffset>(
+          &header->extension.transmissionTimeOffset);
+  header->extension.hasAbsoluteSendTime =
+      GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
+  header->extension.hasTransportSequenceNumber =
+      GetExtension<TransportSequenceNumber>(
+          &header->extension.transportSequenceNumber);
+  header->extension.hasAudioLevel = GetExtension<AudioLevel>(
+      &header->extension.voiceActivity, &header->extension.audioLevel);
+  header->extension.hasVideoRotation =
+      GetExtension<VideoOrientation>(&header->extension.videoRotation);
+  header->extension.hasVideoContentType =
+      GetExtension<VideoContentTypeExtension>(
+          &header->extension.videoContentType);
+  header->extension.has_video_timing =
+      GetExtension<VideoTimingExtension>(&header->extension.video_timing);
+  GetExtension<RtpStreamId>(&header->extension.stream_id);
+  GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
+  GetExtension<RtpMid>(&header->extension.mid);
+  GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
+}
+
+}  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet_received.h b/modules/rtp_rtcp/source/rtp_packet_received.h
index 68c33e9..14d30c9 100644
--- a/modules/rtp_rtcp/source/rtp_packet_received.h
+++ b/modules/rtp_rtcp/source/rtp_packet_received.h
@@ -22,10 +22,9 @@
   explicit RtpPacketReceived(const ExtensionManager* extensions)
       : Packet(extensions) {}
 
-  void GetHeader(RTPHeader* header) const {
-    Packet::GetHeader(header);
-    header->payload_type_frequency = payload_type_frequency();
-  }
+  // TODO(danilchap): Remove this function when all code update to use RtpPacket
+  // directly. Function is there just for easier backward compatibilty.
+  void GetHeader(RTPHeader* header) const;
 
   // Time in local time base as close as it can to packet arrived on the
   // network.
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 5a47759..860f193 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -500,9 +500,6 @@
   packet->SetExtension<VideoTimingExtension>(kVideoTiming);
   EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
   size_t packet_size = packet->size();
-  webrtc::RTPHeader rtp_header;
-
-  packet->GetHeader(&rtp_header);
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
@@ -513,10 +510,10 @@
   EXPECT_EQ(1, transport_.packets_sent());
   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
 
-  transport_.last_sent_packet().GetHeader(&rtp_header);
-  EXPECT_TRUE(rtp_header.extension.has_video_timing);
-  EXPECT_EQ(kStoredTimeInMs,
-            rtp_header.extension.video_timing.pacer_exit_delta_ms);
+  VideoSendTiming video_timing;
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
+      &video_timing));
+  EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
 
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
   rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
@@ -525,10 +522,9 @@
   EXPECT_EQ(2, transport_.packets_sent());
   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
 
-  transport_.last_sent_packet().GetHeader(&rtp_header);
-  EXPECT_TRUE(rtp_header.extension.has_video_timing);
-  EXPECT_EQ(kStoredTimeInMs * 2,
-            rtp_header.extension.video_timing.pacer_exit_delta_ms);
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
+      &video_timing));
+  EXPECT_EQ(kStoredTimeInMs * 2, video_timing.pacer_exit_delta_ms);
 }
 
 TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {