diff --git a/BUILD.gn b/BUILD.gn
index 95d02f77ae..9207a1d690 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -504,6 +504,7 @@ if (rtc_include_tests) {
       "rtc_base/synchronization:sequence_checker_unittests",
       "rtc_base/task_utils:to_queued_task_unittests",
       "sdk:sdk_tests",
+      "test:rtp_test_utils",
       "test:test_main",
       "test/network:network_emulation_unittests",
     ]
diff --git a/call/BUILD.gn b/call/BUILD.gn
index dc545cd6b9..c044a806e5 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -460,6 +460,7 @@ if (rtc_include_tests) {
       "../test:fileutils",
       "../test:null_transport",
       "../test:perf_test",
+      "../test:rtp_test_utils",
       "../test:test_common",
       "../test:test_support",
       "../test:video_test_common",
diff --git a/call/call.cc b/call/call.cc
index 8771380194..62a4378a6c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -40,9 +40,9 @@
 #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "modules/utility/include/process_thread.h"
 #include "modules/video_coding/fec_controller_default.h"
 #include "rtc_base/checks.h"
@@ -155,6 +155,11 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
   return rtclog_config;
 }
 
+bool IsRtcp(const uint8_t* packet, size_t length) {
+  RtpUtility::RtpHeaderParser rtp_parser(packet, length);
+  return rtp_parser.RTCP();
+}
+
 }  // namespace
 
 namespace internal {
@@ -1322,7 +1327,7 @@ PacketReceiver::DeliveryStatus Call::DeliverPacket(
     rtc::CopyOnWriteBuffer packet,
     int64_t packet_time_us) {
   RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
-  if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size()))
+  if (IsRtcp(packet.cdata(), packet.size()))
     return DeliverRtcp(media_type, packet.cdata(), packet.size());
 
   return DeliverRtp(media_type, std::move(packet), packet_time_us);
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 321f69c2a7..5da1fae2c5 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -27,7 +27,6 @@
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/metrics.h"
@@ -41,6 +40,7 @@
 #include "test/frame_generator_capturer.h"
 #include "test/gtest.h"
 #include "test/null_transport.h"
+#include "test/rtp_header_parser.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/single_threaded_task_queue.h"
 #include "test/testsupport/file_utils.h"
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index ce60643e0c..0d54d303c2 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -665,6 +665,7 @@ if (is_linux || is_win) {
       "../system_wrappers:field_trial",
       "../test:field_trial",
       "../test:platform_video_capturer",
+      "../test:rtp_test_utils",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 4635175f14..426b2a0452 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -558,6 +558,7 @@ if (rtc_include_tests) {
       "../rtc_base/third_party/sigslot",
       "../test:audio_codec_mocks",
       "../test:field_trial",
+      "../test:rtp_test_utils",
       "../test:test_main",
       "../test:test_support",
       "../test:video_test_common",
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 48e90f5f45..ba1f671d2a 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -51,7 +51,6 @@
 #include "media/engine/fake_webrtc_video_engine.h"
 #include "media/engine/simulcast.h"
 #include "media/engine/webrtc_voice_engine.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/fake_clock.h"
 #include "rtc_base/gunit.h"
@@ -60,6 +59,7 @@
 #include "test/field_trial.h"
 #include "test/frame_generator.h"
 #include "test/gmock.h"
+#include "test/rtp_header_parser.h"
 
 using ::testing::Field;
 using ::testing::IsEmpty;
@@ -1417,7 +1417,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
   static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
                              webrtc::RTPHeader* header) {
     std::unique_ptr<webrtc::RtpHeaderParser> parser(
-        webrtc::RtpHeaderParser::Create());
+        webrtc::RtpHeaderParser::CreateForTest());
     return parser->Parse(p->cdata(), p->size(), header);
   }
 
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index e28964dd1f..7d72a43e6d 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1099,6 +1099,7 @@ rtc_source_set("neteq_test_tools") {
     "../../test:rtp_test_utils",
     "../rtp_rtcp",
     "../rtp_rtcp:rtp_rtcp_format",
+    "//third_party/abseil-cpp/absl/memory:memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 8545f7aea2..d029c60b41 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -71,12 +71,12 @@ const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
 const uint8_t kPayloadType = 111;
 }  // namespace
 
-class RtpUtility {
+class RtpData {
  public:
-  RtpUtility(int samples_per_packet, uint8_t payload_type)
+  RtpData(int samples_per_packet, uint8_t payload_type)
       : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
 
-  virtual ~RtpUtility() {}
+  virtual ~RtpData() {}
 
   void Populate(RTPHeader* rtp_header) {
     rtp_header->sequenceNumber = 0xABCD;
@@ -163,7 +163,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
 class AudioCodingModuleTestOldApi : public ::testing::Test {
  protected:
   AudioCodingModuleTestOldApi()
-      : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+      : rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)),
         clock_(Clock::GetRealTimeClock()) {}
 
   ~AudioCodingModuleTestOldApi() {}
@@ -239,7 +239,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
     VerifyEncoding();
   }
 
-  std::unique_ptr<RtpUtility> rtp_utility_;
+  std::unique_ptr<RtpData> rtp_utility_;
   std::unique_ptr<AudioCodingModule> acm_;
   PacketizationCallbackStubOldApi packet_cb_;
   RTPHeader rtp_header_;
diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc
index 4e2102da2d..6ed6f98755 100644
--- a/modules/audio_coding/neteq/tools/packet.cc
+++ b/modules/audio_coding/neteq/tools/packet.cc
@@ -14,81 +14,53 @@
 
 #include <memory>
 
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
 namespace test {
 
-Packet::Packet(uint8_t* packet_memory,
-               size_t allocated_bytes,
-               double time_ms,
-               const RtpHeaderParser& parser)
-    : payload_memory_(packet_memory),
-      payload_(NULL),
-      packet_length_bytes_(allocated_bytes),
-      payload_length_bytes_(0),
-      virtual_packet_length_bytes_(allocated_bytes),
-      virtual_payload_length_bytes_(0),
-      time_ms_(time_ms) {
-  valid_header_ = ParseHeader(parser);
-}
+using webrtc::RtpUtility::RtpHeaderParser;
 
 Packet::Packet(uint8_t* packet_memory,
                size_t allocated_bytes,
                size_t virtual_packet_length_bytes,
                double time_ms,
-               const RtpHeaderParser& parser)
+               const RtpUtility::RtpHeaderParser& parser,
+               const RtpHeaderExtensionMap* extension_map /*= nullptr*/)
     : payload_memory_(packet_memory),
-      payload_(NULL),
       packet_length_bytes_(allocated_bytes),
-      payload_length_bytes_(0),
       virtual_packet_length_bytes_(virtual_packet_length_bytes),
       virtual_payload_length_bytes_(0),
-      time_ms_(time_ms) {
-  valid_header_ = ParseHeader(parser);
-}
+      time_ms_(time_ms),
+      valid_header_(ParseHeader(parser, extension_map)) {}
 
 Packet::Packet(const RTPHeader& header,
                size_t virtual_packet_length_bytes,
                size_t virtual_payload_length_bytes,
                double time_ms)
     : header_(header),
-      payload_memory_(),
-      payload_(NULL),
-      packet_length_bytes_(0),
-      payload_length_bytes_(0),
       virtual_packet_length_bytes_(virtual_packet_length_bytes),
       virtual_payload_length_bytes_(virtual_payload_length_bytes),
       time_ms_(time_ms),
       valid_header_(true) {}
 
 Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms)
-    : payload_memory_(packet_memory),
-      payload_(NULL),
-      packet_length_bytes_(allocated_bytes),
-      payload_length_bytes_(0),
-      virtual_packet_length_bytes_(allocated_bytes),
-      virtual_payload_length_bytes_(0),
-      time_ms_(time_ms) {
-  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-  valid_header_ = ParseHeader(*parser);
-}
+    : Packet(packet_memory,
+             allocated_bytes,
+             allocated_bytes,
+             time_ms,
+             RtpUtility::RtpHeaderParser(packet_memory, allocated_bytes)) {}
 
 Packet::Packet(uint8_t* packet_memory,
                size_t allocated_bytes,
                size_t virtual_packet_length_bytes,
                double time_ms)
-    : payload_memory_(packet_memory),
-      payload_(NULL),
-      packet_length_bytes_(allocated_bytes),
-      payload_length_bytes_(0),
-      virtual_packet_length_bytes_(virtual_packet_length_bytes),
-      virtual_payload_length_bytes_(0),
-      time_ms_(time_ms) {
-  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-  valid_header_ = ParseHeader(*parser);
-}
+    : Packet(packet_memory,
+             allocated_bytes,
+             virtual_packet_length_bytes,
+             time_ms,
+             RtpUtility::RtpHeaderParser(packet_memory, allocated_bytes)) {}
 
 Packet::~Packet() = default;
 
@@ -139,9 +111,10 @@ void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
   }
 }
 
-bool Packet::ParseHeader(const RtpHeaderParser& parser) {
-  bool valid_header = parser.Parse(
-      payload_memory_.get(), static_cast<int>(packet_length_bytes_), &header_);
+bool Packet::ParseHeader(const RtpHeaderParser& parser,
+                         const RtpHeaderExtensionMap* extension_map) {
+  bool valid_header = parser.Parse(&header_, extension_map);
+
   // Special case for dummy packets that have padding marked in the RTP header.
   // This causes the RTP header parser to report failure, but is fine in this
   // context.
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index 5748ba2590..f4189aae10 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -15,11 +15,14 @@
 #include <memory>
 
 #include "api/rtp_headers.h"  // NOLINT(build/include)
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
+namespace RtpUtility {
 class RtpHeaderParser;
+}  // namespace RtpUtility
 
 namespace test {
 
@@ -32,22 +35,17 @@ class Packet {
   // when the Packet object is deleted. The |time_ms| is an extra time
   // associated with this packet, typically used to denote arrival time.
   // The first bytes in |packet_memory| will be parsed using |parser|.
-  Packet(uint8_t* packet_memory,
-         size_t allocated_bytes,
-         double time_ms,
-         const RtpHeaderParser& parser);
-
-  // Same as above, but with the extra argument |virtual_packet_length_bytes|.
-  // This is typically used when reading RTP dump files that only contain the
-  // RTP headers, and no payload (a.k.a RTP dummy files or RTP light). The
-  // |virtual_packet_length_bytes| tells what size the packet had on wire,
-  // including the now discarded payload, whereas |allocated_bytes| is the
-  // length of the remaining payload (typically only the RTP header).
+  // |virtual_packet_length_bytes| is typically used when reading RTP dump files
+  // that only contain the RTP headers, and no payload (a.k.a RTP dummy files or
+  // RTP light). The |virtual_packet_length_bytes| tells what size the packet
+  // had on wire, including the now discarded payload, whereas |allocated_bytes|
+  // is the length of the remaining payload (typically only the RTP header).
   Packet(uint8_t* packet_memory,
          size_t allocated_bytes,
          size_t virtual_packet_length_bytes,
          double time_ms,
-         const RtpHeaderParser& parser);
+         const RtpUtility::RtpHeaderParser& parser,
+         const RtpHeaderExtensionMap* extension_map = nullptr);
 
   // Same as above, but creates the packet from an already parsed RTPHeader.
   // This is typically used when reading RTP dump files that only contain the
@@ -98,25 +96,25 @@ class Packet {
 
   const RTPHeader& header() const { return header_; }
 
-  void set_time_ms(double time) { time_ms_ = time; }
   double time_ms() const { return time_ms_; }
   bool valid_header() const { return valid_header_; }
 
  private:
-  bool ParseHeader(const RtpHeaderParser& parser);
+  bool ParseHeader(const webrtc::RtpUtility::RtpHeaderParser& parser,
+                   const RtpHeaderExtensionMap* extension_map);
   void CopyToHeader(RTPHeader* destination) const;
 
   RTPHeader header_;
-  std::unique_ptr<uint8_t[]> payload_memory_;
-  const uint8_t* payload_;            // First byte after header.
-  const size_t packet_length_bytes_;  // Total length of packet.
-  size_t payload_length_bytes_;  // Length of the payload, after RTP header.
-                                 // Zero for dummy RTP packets.
+  const std::unique_ptr<uint8_t[]> payload_memory_;
+  const uint8_t* payload_ = nullptr;      // First byte after header.
+  const size_t packet_length_bytes_ = 0;  // Total length of packet.
+  size_t payload_length_bytes_ = 0;  // Length of the payload, after RTP header.
+                                     // Zero for dummy RTP packets.
   // Virtual lengths are used when parsing RTP header files (dummy RTP files).
   const size_t virtual_packet_length_bytes_;
-  size_t virtual_payload_length_bytes_;
-  double time_ms_;     // Used to denote a packet's arrival time.
-  bool valid_header_;  // Set by the RtpHeaderParser.
+  size_t virtual_payload_length_bytes_ = 0;
+  const double time_ms_;     // Used to denote a packet's arrival time.
+  const bool valid_header_;  // Set by the RtpHeaderParser.
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
 };
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc
index eda2b3e31f..410af27809 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -18,8 +18,8 @@
 
 #include <memory>
 
+#include "absl/memory/memory.h"
 #include "modules/audio_coding/neteq/tools/packet.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/checks.h"
 #include "test/rtp_file_reader.h"
 
@@ -49,8 +49,7 @@ RtpFileSource::~RtpFileSource() {}
 
 bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
                                                uint8_t id) {
-  assert(parser_.get());
-  return parser_->RegisterRtpHeaderExtension(type, id);
+  return rtp_header_extension_map_.RegisterByType(id, type);
 }
 
 std::unique_ptr<Packet> RtpFileSource::NextPacket() {
@@ -66,9 +65,11 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
     }
     std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
     memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
-    std::unique_ptr<Packet> packet(new Packet(
+    RtpUtility::RtpHeaderParser parser(packet_memory.get(), temp_packet.length);
+    auto packet = absl::make_unique<Packet>(
         packet_memory.release(), temp_packet.length,
-        temp_packet.original_length, temp_packet.time_ms, *parser_.get()));
+        temp_packet.original_length, temp_packet.time_ms, parser,
+        &rtp_header_extension_map_);
     if (!packet->valid_header()) {
       continue;
     }
@@ -83,7 +84,6 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
 
 RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
     : PacketSource(),
-      parser_(RtpHeaderParser::Create()),
       ssrc_filter_(ssrc_filter) {}
 
 bool RtpFileSource::OpenFile(const std::string& file_name) {
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index 77e435ae3a..953e2fae5f 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -19,12 +19,11 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
-class RtpHeaderParser;
-
 namespace test {
 
 class RtpFileReader;
@@ -58,8 +57,8 @@ class RtpFileSource : public PacketSource {
   bool OpenFile(const std::string& file_name);
 
   std::unique_ptr<RtpFileReader> rtp_reader_;
-  std::unique_ptr<RtpHeaderParser> parser_;
   const absl::optional<uint32_t> ssrc_filter_;
+  RtpHeaderExtensionMap rtp_header_extension_map_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
 };
diff --git a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index aa60b15388..c0b3a37ba5 100644
--- a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -20,8 +20,8 @@
 #include "absl/flags/parse.h"
 #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "test/rtp_file_reader.h"
+#include "test/rtp_header_parser.h"
 
 ABSL_FLAG(std::string,
           extension_type,
@@ -65,14 +65,14 @@ std::set<uint32_t> SsrcFilter() {
   return ssrcs;
 }
 
-bool ParseArgsAndSetupEstimator(int argc,
-                                char** argv,
-                                webrtc::Clock* clock,
-                                webrtc::RemoteBitrateObserver* observer,
-                                webrtc::test::RtpFileReader** rtp_reader,
-                                webrtc::RtpHeaderParser** parser,
-                                webrtc::RemoteBitrateEstimator** estimator,
-                                std::string* estimator_used) {
+std::unique_ptr<webrtc::RtpHeaderParser> ParseArgsAndSetupEstimator(
+    int argc,
+    char** argv,
+    webrtc::Clock* clock,
+    webrtc::RemoteBitrateObserver* observer,
+    std::unique_ptr<webrtc::test::RtpFileReader>* rtp_reader,
+    std::unique_ptr<webrtc::RemoteBitrateEstimator>* estimator,
+    std::string* estimator_used) {
   absl::ParseCommandLine(argc, argv);
   std::string filename = InputFile();
 
@@ -84,16 +84,16 @@ bool ParseArgsAndSetupEstimator(int argc,
   fprintf(stderr, "\n");
   if (filename.substr(filename.find_last_of('.')) == ".pcap") {
     fprintf(stderr, "Opening as pcap\n");
-    *rtp_reader = webrtc::test::RtpFileReader::Create(
-        webrtc::test::RtpFileReader::kPcap, filename.c_str(), SsrcFilter());
+    rtp_reader->reset(webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kPcap, filename.c_str(), SsrcFilter()));
   } else {
     fprintf(stderr, "Opening as rtp\n");
-    *rtp_reader = webrtc::test::RtpFileReader::Create(
-        webrtc::test::RtpFileReader::kRtpDump, filename.c_str());
+    rtp_reader->reset(webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kRtpDump, filename.c_str()));
   }
   if (!*rtp_reader) {
     fprintf(stderr, "Cannot open input file %s\n", filename.c_str());
-    return false;
+    return nullptr;
   }
   fprintf(stderr, "Input file: %s\n\n", filename.c_str());
 
@@ -105,29 +105,31 @@ bool ParseArgsAndSetupEstimator(int argc,
     fprintf(stderr, "Extension: abs\n");
   } else {
     fprintf(stderr, "Unknown extension type\n");
-    return false;
+    return nullptr;
   }
 
   // Setup the RTP header parser and the bitrate estimator.
-  *parser = webrtc::RtpHeaderParser::Create();
-  (*parser)->RegisterRtpHeaderExtension(extension, ExtensionId());
+  auto parser = webrtc::RtpHeaderParser::CreateForTest();
+  parser->RegisterRtpHeaderExtension(extension, ExtensionId());
   if (estimator) {
     switch (extension) {
       case webrtc::kRtpExtensionAbsoluteSendTime: {
-        *estimator =
-            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock);
+        estimator->reset(
+            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock));
         *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
         break;
       }
       case webrtc::kRtpExtensionTransmissionTimeOffset: {
-        *estimator =
-            new webrtc::RemoteBitrateEstimatorSingleStream(observer, clock);
+        estimator->reset(
+            new webrtc::RemoteBitrateEstimatorSingleStream(observer, clock));
         *estimator_used = "RemoteBitrateEstimator";
         break;
       }
       default:
         assert(false);
+        return nullptr;
     }
   }
-  return true;
+
+  return parser;
 }
diff --git a/modules/remote_bitrate_estimator/tools/bwe_rtp.h b/modules/remote_bitrate_estimator/tools/bwe_rtp.h
index 57484fd95d..4285f926b5 100644
--- a/modules/remote_bitrate_estimator/tools/bwe_rtp.h
+++ b/modules/remote_bitrate_estimator/tools/bwe_rtp.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
 #define MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
 
+#include <memory>
 #include <string>
 
 namespace webrtc {
@@ -23,13 +24,13 @@ class RtpFileReader;
 }
 }  // namespace webrtc
 
-bool ParseArgsAndSetupEstimator(int argc,
-                                char** argv,
-                                webrtc::Clock* clock,
-                                webrtc::RemoteBitrateObserver* observer,
-                                webrtc::test::RtpFileReader** rtp_reader,
-                                webrtc::RtpHeaderParser** parser,
-                                webrtc::RemoteBitrateEstimator** estimator,
-                                std::string* estimator_used);
+std::unique_ptr<webrtc::RtpHeaderParser> ParseArgsAndSetupEstimator(
+    int argc,
+    char** argv,
+    webrtc::Clock* clock,
+    webrtc::RemoteBitrateObserver* observer,
+    std::unique_ptr<webrtc::test::RtpFileReader>* rtp_reader,
+    std::unique_ptr<webrtc::RemoteBitrateEstimator>* estimator,
+    std::string* estimator_used);
 
 #endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
diff --git a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index c362623067..7f1e009793 100644
--- a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -13,21 +13,20 @@
 #include <memory>
 
 #include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/format_macros.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/rtp_file_reader.h"
+#include "test/rtp_header_parser.h"
 
 int main(int argc, char* argv[]) {
-  webrtc::test::RtpFileReader* reader;
-  webrtc::RtpHeaderParser* parser;
-  if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser,
-                                  NULL, NULL)) {
+  std::unique_ptr<webrtc::test::RtpFileReader> reader;
+  std::unique_ptr<webrtc::RtpHeaderParser> parser(ParseArgsAndSetupEstimator(
+      argc, argv, nullptr, nullptr, &reader, nullptr, nullptr));
+  if (!parser)
     return -1;
-  }
+
   bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
-  std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
-  std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
+
   fprintf(stdout,
           "seqnum timestamp ts_offset abs_sendtime recvtime "
           "markerbit ssrc size original_size\n");
@@ -35,7 +34,7 @@ int main(int argc, char* argv[]) {
   int non_zero_abs_send_time = 0;
   int non_zero_ts_offsets = 0;
   webrtc::test::RtpPacket packet;
-  while (rtp_reader->NextPacket(&packet)) {
+  while (reader->NextPacket(&packet)) {
     webrtc::RTPHeader header;
     parser->Parse(packet.data, packet.length, &header);
     if (header.extension.absoluteSendTime != 0)
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 06ed9bd8e5..24ed0d2206 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -129,7 +129,6 @@ rtc_static_library("rtp_rtcp") {
     "include/flexfec_sender.h",
     "include/receive_statistics.h",
     "include/remote_ntp_time_estimator.h",
-    "include/rtp_header_parser.h",
     "include/rtp_rtcp.h",
     "include/ulpfec_receiver.h",
     "source/absolute_capture_time_receiver.cc",
@@ -175,7 +174,6 @@ rtc_static_library("rtp_rtcp") {
     "source/rtp_format_vp9.h",
     "source/rtp_header_extension_size.cc",
     "source/rtp_header_extension_size.h",
-    "source/rtp_header_parser.cc",
     "source/rtp_packet_history.cc",
     "source/rtp_packet_history.h",
     "source/rtp_rtcp_config.h",
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index e6f8db130a..f55e4f81d2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -17,7 +17,6 @@
 #include "absl/memory/memory.h"
 #include "api/transport/field_trial_based_config.h"
 #include "api/video_codecs/video_codec.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "modules/rtp_rtcp/source/rtcp_packet.h"
@@ -28,6 +27,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"
+#include "test/rtp_header_parser.h"
 
 using ::testing::_;
 using ::testing::ElementsAre;
@@ -72,7 +72,7 @@ class SendTransport : public Transport {
                size_t len,
                const PacketOptions& options) override {
     RTPHeader header;
-    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::CreateForTest());
     EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
     ++rtp_packets_sent_;
     last_rtp_header_ = header;
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 125a0b8994..ad501df57e 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -21,7 +21,6 @@
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/include/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
@@ -39,6 +38,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_transport.h"
+#include "test/rtp_header_parser.h"
 
 namespace webrtc {
 
diff --git a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
index cd1798bbea..32f3bbb2e2 100644
--- a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
@@ -15,7 +15,6 @@
 #include <list>
 #include <memory>
 
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -23,6 +22,7 @@
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/rtp_header_parser.h"
 
 namespace webrtc {
 
@@ -177,7 +177,7 @@ void UlpfecReceiverTest::SurvivesMaliciousPacket(const uint8_t* data,
                                                  size_t length,
                                                  uint8_t ulpfec_payload_type) {
   RTPHeader header;
-  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::CreateForTest());
   ASSERT_TRUE(parser->Parse(data, length, &header));
 
   NullRecoveredPacketReceiver null_callback;
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 814720f502..64706c1b69 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -558,6 +558,7 @@ if (rtc_include_tests) {
       "../system_wrappers:metrics",
       "../test:field_trial",
       "../test:fileutils",
+      "../test:rtp_test_utils",
       "//third_party/abseil-cpp/absl/algorithm:container",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings",
diff --git a/pc/datagram_rtp_transport.cc b/pc/datagram_rtp_transport.cc
index ebf82a7ab9..02e7a6f022 100644
--- a/pc/datagram_rtp_transport.cc
+++ b/pc/datagram_rtp_transport.cc
@@ -20,7 +20,6 @@
 #include "api/array_view.h"
 #include "api/rtc_error.h"
 #include "media/base/rtp_utils.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
diff --git a/test/BUILD.gn b/test/BUILD.gn
index bc56d898d2..74c52686ef 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -131,15 +131,19 @@ rtc_source_set("rtp_test_utils") {
     "rtp_file_reader.h",
     "rtp_file_writer.cc",
     "rtp_file_writer.h",
+    "rtp_header_parser.cc",
+    "rtp_header_parser.h",
   ]
 
   deps = [
     "../api:array_view",
+    "../api:rtp_parameters",
     "../modules/rtp_rtcp",
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/system:arch",
+    "//third_party/abseil-cpp/absl/memory",
   ]
 }
 
@@ -597,6 +601,7 @@ rtc_source_set("direct_transport") {
     "direct_transport.h",
   ]
   deps = [
+    ":rtp_test_utils",
     "../api:simulated_network_api",
     "../api:transport_api",
     "../call:call_interfaces",
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index 4638652163..7ca5bb1e65 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -12,8 +12,8 @@
 #include "absl/memory/memory.h"
 #include "call/call.h"
 #include "call/fake_network_pipe.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/time_utils.h"
+#include "test/rtp_header_parser.h"
 #include "test/single_threaded_task_queue.h"
 
 namespace webrtc {
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index cd8e21742f..36183036f2 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -616,7 +616,7 @@ webrtc_fuzzer_test("rtp_header_parser_fuzzer") {
     "rtp_header_parser_fuzzer.cc",
   ]
   deps = [
-    "../../modules/rtp_rtcp",
+    "../:rtp_test_utils",
   ]
 }
 
@@ -625,7 +625,7 @@ webrtc_fuzzer_test("ssl_certificate_fuzzer") {
     "rtp_header_parser_fuzzer.cc",
   ]
   deps = [
-    "../../modules/rtp_rtcp",
+    "../:rtp_test_utils",
   ]
 }
 
diff --git a/test/fuzzers/rtp_header_parser_fuzzer.cc b/test/fuzzers/rtp_header_parser_fuzzer.cc
index 6d95fdc2dd..d6af5ca3ce 100644
--- a/test/fuzzers/rtp_header_parser_fuzzer.cc
+++ b/test/fuzzers/rtp_header_parser_fuzzer.cc
@@ -15,7 +15,7 @@
 #include <memory>
 #include <string>
 
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "test/rtp_header_parser.h"
 
 namespace webrtc {
 
@@ -24,7 +24,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   RtpHeaderParser::GetSsrc(data, size);
   RTPHeader rtp_header;
 
-  std::unique_ptr<RtpHeaderParser> rtp_header_parser(RtpHeaderParser::Create());
+  std::unique_ptr<RtpHeaderParser> rtp_header_parser(
+      RtpHeaderParser::CreateForTest());
 
   rtp_header_parser->Parse(data, size, &rtp_header);
   for (int i = 1; i < kRtpExtensionNumberOfExtensions; ++i) {
diff --git a/test/fuzzers/utils/BUILD.gn b/test/fuzzers/utils/BUILD.gn
index 307cbe170a..007c750b60 100644
--- a/test/fuzzers/utils/BUILD.gn
+++ b/test/fuzzers/utils/BUILD.gn
@@ -23,7 +23,6 @@ rtc_source_set("rtp_replayer") {
     "../../../call:call_interfaces",
     "../../../common_video",
     "../../../media:rtc_internal_video_codecs",
-    "../../../modules/rtp_rtcp",
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:rtc_json",
diff --git a/test/fuzzers/utils/rtp_replayer.cc b/test/fuzzers/utils/rtp_replayer.cc
index e430d40d86..0656f4c8f6 100644
--- a/test/fuzzers/utils/rtp_replayer.cc
+++ b/test/fuzzers/utils/rtp_replayer.cc
@@ -16,7 +16,6 @@
 
 #include "absl/memory/memory.h"
 #include "api/task_queue/default_task_queue_factory.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/strings/json.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/sleep.h"
@@ -24,6 +23,7 @@
 #include "test/encoder_settings.h"
 #include "test/fake_decoder.h"
 #include "test/rtp_file_reader.h"
+#include "test/rtp_header_parser.h"
 
 namespace webrtc {
 namespace test {
@@ -158,7 +158,8 @@ void RtpReplayer::ReplayPackets(Call* call, test::RtpFileReader* rtp_reader) {
         break;
       case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
         RTPHeader header;
-        std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+        std::unique_ptr<RtpHeaderParser> parser(
+            RtpHeaderParser::CreateForTest());
 
         parser->Parse(packet.data, packet.length, &header);
         if (unknown_packets[header.ssrc] == 0) {
@@ -171,7 +172,8 @@ void RtpReplayer::ReplayPackets(Call* call, test::RtpFileReader* rtp_reader) {
         RTC_LOG(LS_ERROR)
             << "Packet error, corrupt packets or incorrect setup?";
         RTPHeader header;
-        std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+        std::unique_ptr<RtpHeaderParser> parser(
+            RtpHeaderParser::CreateForTest());
         parser->Parse(packet.data, packet.length, &header);
         RTC_LOG(LS_ERROR) << "Packet packet_length=" << packet.length
                           << " payload_type=" << header.payloadType
diff --git a/modules/rtp_rtcp/source/rtp_header_parser.cc b/test/rtp_header_parser.cc
similarity index 93%
rename from modules/rtp_rtcp/source/rtp_header_parser.cc
rename to test/rtp_header_parser.cc
index 65431d1397..1a4ba42ff7 100644
--- a/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/test/rtp_header_parser.cc
@@ -7,10 +7,11 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "test/rtp_header_parser.h"
 
-#include <string.h>
+#include <memory>
 
+#include "absl/memory/memory.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "rtc_base/critical_section.h"
@@ -39,8 +40,8 @@ class RtpHeaderParserImpl : public RtpHeaderParser {
       RTC_GUARDED_BY(critical_section_);
 };
 
-RtpHeaderParser* RtpHeaderParser::Create() {
-  return new RtpHeaderParserImpl;
+std::unique_ptr<RtpHeaderParser> RtpHeaderParser::CreateForTest() {
+  return absl::make_unique<RtpHeaderParserImpl>();
 }
 
 RtpHeaderParserImpl::RtpHeaderParserImpl() {}
diff --git a/modules/rtp_rtcp/include/rtp_header_parser.h b/test/rtp_header_parser.h
similarity index 89%
rename from modules/rtp_rtcp/include/rtp_header_parser.h
rename to test/rtp_header_parser.h
index 0afcb71fd6..851ccf3bc2 100644
--- a/modules/rtp_rtcp/include/rtp_header_parser.h
+++ b/test/rtp_header_parser.h
@@ -7,8 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
-#define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#ifndef TEST_RTP_HEADER_PARSER_H_
+#define TEST_RTP_HEADER_PARSER_H_
+
+#include <memory>
 
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -19,7 +21,7 @@ struct RTPHeader;
 
 class RtpHeaderParser {
  public:
-  static RtpHeaderParser* Create();
+  static std::unique_ptr<RtpHeaderParser> CreateForTest();
   virtual ~RtpHeaderParser() {}
 
   // Returns true if the packet is an RTCP packet, false otherwise.
@@ -48,4 +50,4 @@ class RtpHeaderParser {
   virtual bool DeregisterRtpHeaderExtension(RtpExtension extension) = 0;
 };
 }  // namespace webrtc
-#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#endif  // TEST_RTP_HEADER_PARSER_H_
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index 830c2f12bb..57630397ac 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -18,12 +18,12 @@
 #include "api/test/simulated_network.h"
 #include "call/simulated_packet_receiver.h"
 #include "call/video_send_stream.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "system_wrappers/include/field_trial.h"
 #include "test/direct_transport.h"
 #include "test/gtest.h"
+#include "test/rtp_header_parser.h"
 
 namespace {
 const int kShortTimeoutMs = 500;
@@ -71,7 +71,8 @@ class RtpRtcpObserver {
  protected:
   RtpRtcpObserver() : RtpRtcpObserver(0) {}
   explicit RtpRtcpObserver(int event_timeout_ms)
-      : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {}
+      : parser_(RtpHeaderParser::CreateForTest()),
+        timeout_ms_(event_timeout_ms) {}
 
   rtc::Event observation_complete_;
   const std::unique_ptr<RtpHeaderParser> parser_;
diff --git a/test/scenario/BUILD.gn b/test/scenario/BUILD.gn
index 979227101d..dce7775c47 100644
--- a/test/scenario/BUILD.gn
+++ b/test/scenario/BUILD.gn
@@ -76,6 +76,7 @@ if (rtc_include_tests) {
       ":column_printer",
       "../:fake_video_codecs",
       "../:fileutils",
+      "../:rtp_test_utils",
       "../:test_common",
       "../:test_support",
       "../:video_test_common",
diff --git a/test/scenario/call_client.cc b/test/scenario/call_client.cc
index 31435bb21a..1654afc348 100644
--- a/test/scenario/call_client.cc
+++ b/test/scenario/call_client.cc
@@ -203,7 +203,7 @@ CallClient::CallClient(
       clock_(time_controller->GetClock()),
       log_writer_factory_(std::move(log_writer_factory)),
       network_controller_factory_(log_writer_factory_.get(), config.transport),
-      header_parser_(RtpHeaderParser::Create()),
+      header_parser_(RtpHeaderParser::CreateForTest()),
       task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
           "CallClient",
           TaskQueueFactory::Priority::NORMAL)) {
diff --git a/test/scenario/call_client.h b/test/scenario/call_client.h
index d2603a87c7..78c302d5a8 100644
--- a/test/scenario/call_client.h
+++ b/test/scenario/call_client.h
@@ -20,11 +20,11 @@
 #include "call/call.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/constructor_magic.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/logging/log_writer.h"
 #include "test/network/network_emulation.h"
+#include "test/rtp_header_parser.h"
 #include "test/scenario/column_printer.h"
 #include "test/scenario/network_node.h"
 #include "test/scenario/scenario_config.h"
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index e3ba87a652..ce1d86739e 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -611,7 +611,8 @@ TEST_F(StatsEndToEndTest, VerifyNackStats) {
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       if (++sent_rtp_packets_ == kPacketNumberToDrop) {
-        std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+        std::unique_ptr<RtpHeaderParser> parser(
+            RtpHeaderParser::CreateForTest());
         RTPHeader header;
         EXPECT_TRUE(parser->Parse(packet, length, &header));
         dropped_rtp_packet_ = header.sequenceNumber;
diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc
index d8ae23014d..da59405293 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -50,7 +50,7 @@ TEST_F(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
                                   BuiltInNetworkBehaviorConfig())),
                           sender_call,
                           payload_type_map),
-          parser_(RtpHeaderParser::Create()),
+          parser_(RtpHeaderParser::CreateForTest()),
           first_media_ssrc_(first_media_ssrc),
           rtx_to_media_ssrcs_(ssrc_map),
           padding_observed_(false),
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index a1a91b4e5b..33f9898e71 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -24,7 +24,6 @@
 #include "call/rtp_transport_controller_send.h"
 #include "call/simulated_network.h"
 #include "call/video_send_stream.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtcp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
@@ -53,6 +52,7 @@
 #include "test/gtest.h"
 #include "test/null_transport.h"
 #include "test/rtcp_packet_parser.h"
+#include "test/rtp_header_parser.h"
 #include "test/testsupport/perf_test.h"
 #include "test/video_encoder_proxy_factory.h"
 #include "video/send_statistics_proxy.h"