diff --git a/test/BUILD.gn b/test/BUILD.gn
index 9172eb3227..d645621e79 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -517,6 +517,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":test_support_test_artifacts",
       ":video_test_common",
       ":video_test_support",
+      "../api:array_view",
       "../api:create_frame_generator",
       "../api:create_simulcast_test_fixture_api",
       "../api:frame_generator_api",
@@ -530,7 +531,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../call:video_stream_api",
       "../common_video",
       "../media:rtc_media_base",
-      "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:rtp_rtcp_format",
       "../modules/video_coding:simulcast_test_fixture_impl",
       "../modules/video_coding:video_codec_interface",
diff --git a/test/peer_scenario/tests/BUILD.gn b/test/peer_scenario/tests/BUILD.gn
index 042b636690..dd72473bf8 100644
--- a/test/peer_scenario/tests/BUILD.gn
+++ b/test/peer_scenario/tests/BUILD.gn
@@ -21,7 +21,6 @@ if (rtc_include_tests) {
       "../../:field_trial",
       "../../:test_support",
       "../../../media:rtc_media_base",
-      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../pc:rtc_pc_base",
       "../../../pc:session_description",
diff --git a/test/peer_scenario/tests/remote_estimate_test.cc b/test/peer_scenario/tests/remote_estimate_test.cc
index f1d8345fde..429a5b4ef6 100644
--- a/test/peer_scenario/tests/remote_estimate_test.cc
+++ b/test/peer_scenario/tests/remote_estimate_test.cc
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "pc/media_session.h"
 #include "pc/session_description.h"
 #include "test/field_trial.h"
@@ -26,19 +28,6 @@ RtpHeaderExtensionMap AudioExtensions(
   return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
 }
 
-absl::optional<RTPHeaderExtension> GetRtpPacketExtensions(
-    const rtc::ArrayView<const uint8_t> packet,
-    const RtpHeaderExtensionMap& extension_map) {
-  RtpUtility::RtpHeaderParser rtp_parser(packet.data(), packet.size());
-  if (IsRtpPacket(packet)) {
-    RTPHeader header;
-    if (rtp_parser.Parse(&header, &extension_map, true)) {
-      return header.extension;
-    }
-  }
-  return absl::nullopt;
-}
-
 }  // namespace
 
 TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
@@ -106,13 +95,10 @@ TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
         // The dummy packets used by the fake signaling are filled with 0. We
         // want to ignore those and we can do that on the basis that the first
         // byte of RTP packets are guaranteed to not be 0.
-        // TODO(srte): Find a more elegant way to check for RTP traffic.
-        if (packet.size() > 1 && packet.cdata()[0] != 0) {
-          auto extensions = GetRtpPacketExtensions(packet.data, extension_map);
-          if (extensions) {
-            EXPECT_TRUE(extensions->hasAbsoluteSendTime);
-            received_abs_send_time = true;
-          }
+        RtpPacket rtp_packet(&extension_map);
+        if (rtp_packet.Parse(packet.data)) {
+          EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
+          received_abs_send_time = true;
         }
       });
   RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
diff --git a/test/rtp_file_reader_unittest.cc b/test/rtp_file_reader_unittest.cc
index 8dc817de60..995d9fbc9d 100644
--- a/test/rtp_file_reader_unittest.cc
+++ b/test/rtp_file_reader_unittest.cc
@@ -13,7 +13,8 @@
 #include <map>
 #include <memory>
 
-#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
@@ -84,11 +85,9 @@ class TestPcapFileReader : public ::testing::Test {
     PacketsPerSsrc pps;
     test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
-      RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
-      webrtc::RTPHeader header;
-      if (!rtp_header_parser.RTCP() &&
-          rtp_header_parser.Parse(&header, nullptr)) {
-        pps[header.ssrc]++;
+      rtc::ArrayView<const uint8_t> raw(packet.data, packet.length);
+      if (IsRtpPacket(raw)) {
+        pps[ParseRtpSsrc(raw)]++;
       }
     }
     return pps;