diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
index 55de03d..8a83554 100644
--- a/audio/audio_send_stream_tests.cc
+++ b/audio/audio_send_stream_tests.cc
@@ -8,8 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <string>
+#include <utility>
+#include <vector>
+
 #include "test/call_test.h"
-#include "test/constants.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"
@@ -18,6 +21,11 @@
 namespace test {
 namespace {
 
+enum : int {  // The first valid value is 1.
+  kAudioLevelExtensionId = 1,
+  kTransportSequenceNumberExtensionId,
+};
+
 class AudioSendTest : public SendTest {
  public:
   AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
@@ -103,8 +111,8 @@
   class AudioLevelObserver : public AudioSendTest {
    public:
     AudioLevelObserver() : AudioSendTest() {
-      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionAudioLevel, test::kAudioLevelExtensionId));
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                      kAudioLevelExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -127,8 +135,8 @@
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAudioLevelUri, test::kAudioLevelExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelExtensionId));
     }
 
     void PerformTest() override {
diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc
index 74eaef0..2e3c158 100644
--- a/audio/test/audio_bwe_integration_test.cc
+++ b/audio/test/audio_bwe_integration_test.cc
@@ -23,6 +23,10 @@
 namespace test {
 
 namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+};
+
 // Wait a second between stopping sending and stopping receiving audio.
 constexpr int kExtraProcessTimeMs = 1000;
 }  // namespace
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 3ffcb6b..7325c15 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -51,8 +51,19 @@
 using webrtc::test::DriftingClock;
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+};
+}  // namespace
 
 class CallPerfTest : public test::CallTest {
+ public:
+  CallPerfTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+  }
+
  protected:
   enum class FecMode { kOn, kOff };
   enum class CreateOrder { kAudioFirst, kVideoFirst };
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 7f9d72f..c5ca7f0 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -22,7 +22,10 @@
 namespace webrtc {
 
 namespace {
-const int kAudioLevelExtensionId = 9;
+enum : int {  // The first valid value is 1.
+  kAudioLevelExtensionId = 1,
+};
+
 const uint16_t kSeqNum = 33;
 const uint32_t kSsrc = 725242;
 const uint8_t kAudioLevel = 0x5a;
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 714fac7..0b881f8 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -40,16 +40,19 @@
 namespace webrtc {
 
 namespace {
-const int kTransmissionTimeOffsetExtensionId = 1;
-const int kAbsoluteSendTimeExtensionId = 14;
-const int kTransportSequenceNumberExtensionId = 13;
-const int kVideoTimingExtensionId = 12;
-const int kMidExtensionId = 11;
-const int kGenericDescriptorId = 10;
-const int kAudioLevelExtensionId = 9;
-const int kRidExtensionId = 8;
-const int kRepairedRidExtensionId = 7;
-const int kVideoRotationExtensionId = 5;
+enum : int {  // The first valid value is 1.
+  kAbsoluteSendTimeExtensionId = 1,
+  kAudioLevelExtensionId,
+  kGenericDescriptorId,
+  kMidExtensionId,
+  kRepairedRidExtensionId,
+  kRidExtensionId,
+  kTransmissionTimeOffsetExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoRotationExtensionId,
+  kVideoTimingExtensionId,
+};
+
 const int kPayload = 100;
 const int kRtxPayload = 98;
 const uint32_t kTimestamp = 10;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 43eb4ed..0cd0572 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -34,13 +34,16 @@
 
 using ::testing::ElementsAre;
 
-constexpr int kTransmissionTimeOffsetExtensionId = 1;
-constexpr int kAbsoluteSendTimeExtensionId = 14;
-constexpr int kTransportSequenceNumberExtensionId = 13;
-constexpr int kVideoTimingExtensionId = 12;
-constexpr int kGenericDescriptorId = 10;
-constexpr int kFrameMarkingExtensionId = 6;
-constexpr int kVideoRotationExtensionId = 5;
+enum : int {  // The first valid value is 1.
+  kAbsoluteSendTimeExtensionId = 1,
+  kFrameMarkingExtensionId,
+  kGenericDescriptorId,
+  kTransmissionTimeOffsetExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoRotationExtensionId,
+  kVideoTimingExtensionId,
+};
+
 constexpr int kPayload = 100;
 constexpr uint32_t kTimestamp = 10;
 constexpr uint16_t kSeqNum = 33;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 99c5467..8acc26d 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -656,8 +656,6 @@
   sources = [
     "call_test.cc",
     "call_test.h",
-    "constants.cc",
-    "constants.h",
     "drifting_clock.cc",
     "drifting_clock.h",
     "encoder_settings.cc",
@@ -748,6 +746,7 @@
     "../video",
     "//testing/gtest",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
   if (!is_android && !build_with_chromium) {
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
diff --git a/test/call_test.cc b/test/call_test.cc
index 57a5d9f..ca0d7bb 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -65,6 +65,29 @@
   });
 }
 
+void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
+  for (const RtpExtension& registered_extension : rtp_extensions_) {
+    if (registered_extension.id == extension.id) {
+      ASSERT_EQ(registered_extension.uri, extension.uri)
+          << "Different URIs associated with ID " << extension.id << ".";
+      ASSERT_EQ(registered_extension.encrypt, extension.encrypt)
+          << "Encryption mismatch associated with ID " << extension.id << ".";
+      return;
+    } else {  // Different IDs.
+      // Different IDs referring to the same extension probably indicate
+      // a mistake in the test.
+      ASSERT_FALSE(registered_extension.uri == extension.uri &&
+                   registered_extension.encrypt == extension.encrypt)
+          << "URI " << extension.uri
+          << (extension.encrypt ? " with " : " without ")
+          << "encryption already registered with a different "
+          << "ID (" << extension.id << " vs. " << registered_extension.id
+          << ").";
+    }
+  }
+  rtp_extensions_.push_back(extension);
+}
+
 void CallTest::RunBaseTest(BaseTest* test) {
   task_queue_.SendTask([this, test]() {
     num_video_streams_ = test->GetNumVideoStreams();
@@ -235,25 +258,23 @@
   video_config->rtp.payload_name = "FAKE";
   video_config->rtp.payload_type = kFakeVideoSendPayloadType;
   video_config->rtp.extmap_allow_mixed = true;
-  video_config->rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                   kTransportSequenceNumberExtensionId));
-  video_config->rtp.extensions.push_back(RtpExtension(
-      RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
-  video_config->rtp.extensions.push_back(RtpExtension(
-      RtpExtension::kGenericFrameDescriptorUri, kGenericDescriptorExtensionId));
+  AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri,
+                       &video_config->rtp.extensions);
   if (video_encoder_configs_.empty()) {
     video_encoder_configs_.emplace_back();
     FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
                              &video_encoder_configs_.back());
   }
-
   for (size_t i = 0; i < num_video_streams; ++i)
     video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
-  video_config->rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
-  video_config->rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
+  AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
+                       &video_config->rtp.extensions);
 }
 
 void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
@@ -666,6 +687,25 @@
   return &flexfec_receive_configs_[0];
 }
 
+absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
+    const std::string& uri) const {
+  for (const auto& extension : rtp_extensions_) {
+    if (extension.uri == uri) {
+      return extension;
+    }
+  }
+  return absl::nullopt;
+}
+
+void CallTest::AddRtpExtensionByUri(
+    const std::string& uri,
+    std::vector<RtpExtension>* extensions) const {
+  const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
+  if (extension) {
+    extensions->push_back(*extension);
+  }
+}
+
 constexpr size_t CallTest::kNumSsrcs;
 const int CallTest::kDefaultWidth;
 const int CallTest::kDefaultHeight;
diff --git a/test/call_test.h b/test/call_test.h
index fb26051..dbe8e07 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/test/video/function_video_decoder_factory.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/video/video_bitrate_allocator_factory.h"
@@ -71,6 +72,8 @@
   static const std::map<uint8_t, MediaType> payload_type_map_;
 
  protected:
+  void RegisterRtpExtension(const RtpExtension& extension);
+
   // RunBaseTest overwrites the audio_state of the send and receive Call configs
   // to simplify test code.
   void RunBaseTest(BaseTest* test);
@@ -216,6 +219,13 @@
   SingleThreadedTaskQueueForTesting task_queue_;
 
  private:
+  absl::optional<RtpExtension> GetRtpExtensionByUri(
+      const std::string& uri) const;
+
+  void AddRtpExtensionByUri(const std::string& uri,
+                            std::vector<RtpExtension>* extensions) const;
+
+  std::vector<RtpExtension> rtp_extensions_;
   rtc::scoped_refptr<AudioProcessing> apm_send_;
   rtc::scoped_refptr<AudioProcessing> apm_recv_;
   rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
diff --git a/test/constants.cc b/test/constants.cc
deleted file mode 100644
index 4f33d25..0000000
--- a/test/constants.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2015 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 "test/constants.h"
-
-namespace webrtc {
-namespace test {
-
-const int kAudioLevelExtensionId = 5;
-const int kTOffsetExtensionId = 6;
-const int kAbsSendTimeExtensionId = 7;
-const int kTransportSequenceNumberExtensionId = 8;
-const int kVideoRotationExtensionId = 9;
-const int kVideoContentTypeExtensionId = 10;
-const int kVideoTimingExtensionId = 11;
-const int kGenericDescriptorExtensionId = 12;
-const int kColorSpaceExtensionId = 13;
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/test/constants.h b/test/constants.h
deleted file mode 100644
index b1b87d6..0000000
--- a/test/constants.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- *  Copyright (c) 2015 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.
- */
-
-namespace webrtc {
-namespace test {
-
-extern const int kAudioLevelExtensionId;
-extern const int kTOffsetExtensionId;
-extern const int kAbsSendTimeExtensionId;
-extern const int kTransportSequenceNumberExtensionId;
-extern const int kVideoRotationExtensionId;
-extern const int kVideoContentTypeExtensionId;
-extern const int kVideoTimingExtensionId;
-extern const int kGenericDescriptorExtensionId;
-extern const int kColorSpaceExtensionId;
-}  // namespace test
-}  // namespace webrtc
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index 7cff645..830c2f1 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -12,6 +12,7 @@
 
 #include <map>
 #include <memory>
+#include <utility>
 #include <vector>
 
 #include "api/test/simulated_network.h"
@@ -21,7 +22,6 @@
 #include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "system_wrappers/include/field_trial.h"
-#include "test/constants.h"
 #include "test/direct_transport.h"
 #include "test/gtest.h"
 
@@ -71,14 +71,7 @@
  protected:
   RtpRtcpObserver() : RtpRtcpObserver(0) {}
   explicit RtpRtcpObserver(int event_timeout_ms)
-      : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {
-    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
-                                        kTOffsetExtensionId);
-    parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
-                                        kAbsSendTimeExtensionId);
-    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
-                                        kTransportSequenceNumberExtensionId);
-  }
+      : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {}
 
   rtc::Event observation_complete_;
   const std::unique_ptr<RtpHeaderParser> parser_;
diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc
index 2f892f0..859626e 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -27,8 +27,13 @@
 namespace webrtc {
 namespace test {
 namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kVideoContentTypeExtensionId,
+  kVideoRotationRtpExtensionId,
+};
+
 constexpr int kDefaultMaxQp = cricket::WebRtcVideoChannel::kDefaultQpMax;
-const int kVideoRotationRtpExtensionId = 4;
 uint8_t CodecTypeToPayloadType(VideoCodecType codec_type) {
   switch (codec_type) {
     case VideoCodecType::kVideoCodecGeneric:
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 17bb985..e7550a4 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -26,6 +26,11 @@
 #include "test/video_encoder_proxy_factory.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kAbsSendTimeExtensionId = 1,
+};
+}  // namespace
 
 class BandwidthEndToEndTest : public test::CallTest {
  public:
@@ -42,8 +47,8 @@
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
       (*receive_configs)[0].rtp.remb = true;
       (*receive_configs)[0].rtp.transport_cc = false;
     }
@@ -87,8 +92,8 @@
       VideoEncoderConfig* encoder_config) override {
     if (!send_side_bwe_) {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
       (*receive_configs)[0].rtp.remb = true;
       (*receive_configs)[0].rtp.transport_cc = false;
     }
diff --git a/video/end_to_end_tests/codec_tests.cc b/video/end_to_end_tests/codec_tests.cc
index ed303ec..eab9ef7 100644
--- a/video/end_to_end_tests/codec_tests.cc
+++ b/video/end_to_end_tests/codec_tests.cc
@@ -27,11 +27,22 @@
 #include "test/gtest.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kColorSpaceExtensionId = 1,
+  kVideoRotationExtensionId,
+};
+}  // namespace
 
 class CodecEndToEndTest : public test::CallTest,
                           public testing::WithParamInterface<std::string> {
  public:
-  CodecEndToEndTest() : field_trial_(GetParam()) {}
+  CodecEndToEndTest() : field_trial_(GetParam()) {
+    RegisterRtpExtension(
+        RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+                                      kVideoRotationExtensionId));
+  }
 
  private:
   test::ScopedFieldTrials field_trial_;
@@ -224,7 +235,10 @@
 class EndToEndTestH264 : public test::CallTest,
                          public testing::WithParamInterface<std::string> {
  public:
-  EndToEndTestH264() : field_trial_(GetParam()) {}
+  EndToEndTestH264() : field_trial_(GetParam()) {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+                                      kVideoRotationExtensionId));
+  }
 
  private:
   test::ScopedFieldTrials field_trial_;
diff --git a/video/end_to_end_tests/extended_reports_tests.cc b/video/end_to_end_tests/extended_reports_tests.cc
index 5b649d5..955f9cd 100644
--- a/video/end_to_end_tests/extended_reports_tests.cc
+++ b/video/end_to_end_tests/extended_reports_tests.cc
@@ -43,8 +43,20 @@
 #include "test/single_threaded_task_queue.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kColorSpaceExtensionId = 1,
+  kTransportSequenceNumberExtensionId,
+};
+}  // namespace
 
-class ExtendedReportsEndToEndTest : public test::CallTest {};
+class ExtendedReportsEndToEndTest : public test::CallTest {
+ public:
+  ExtendedReportsEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+  }
+};
 
 class RtcpXrObserver : public test::EndToEndTest {
  public:
diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc
index 42215d3..dbf0b2c 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -22,10 +22,21 @@
 #include "test/rtcp_packet_parser.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kVideoRotationExtensionId,
+};
+}  // namespace
 
 class FecEndToEndTest : public test::CallTest {
  public:
-  FecEndToEndTest() = default;
+  FecEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+                                      kVideoRotationExtensionId));
+  }
 };
 
 TEST_F(FecEndToEndTest, ReceivesUlpfec) {
diff --git a/video/end_to_end_tests/frame_encryption_tests.cc b/video/end_to_end_tests/frame_encryption_tests.cc
index 98b3a4b..85ad7dd 100644
--- a/video/end_to_end_tests/frame_encryption_tests.cc
+++ b/video/end_to_end_tests/frame_encryption_tests.cc
@@ -17,10 +17,18 @@
 #include "test/gtest.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kGenericDescriptorExtensionId = 1,
+};
+}  // namespace
 
 class FrameEncryptionEndToEndTest : public test::CallTest {
  public:
-  FrameEncryptionEndToEndTest() = default;
+  FrameEncryptionEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kGenericFrameDescriptorUri,
+                                      kGenericDescriptorExtensionId));
+  }
 
  private:
   // GenericDescriptor is required for FrameEncryption to work.
diff --git a/video/end_to_end_tests/histogram_tests.cc b/video/end_to_end_tests/histogram_tests.cc
index 312924c..ef435b4 100644
--- a/video/end_to_end_tests/histogram_tests.cc
+++ b/video/end_to_end_tests/histogram_tests.cc
@@ -16,8 +16,22 @@
 #include "test/gtest.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kVideoContentTypeExtensionId,
+};
+}  // namespace
 
 class HistogramTest : public test::CallTest {
+ public:
+  HistogramTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
+                                      kVideoContentTypeExtensionId));
+  }
+
  protected:
   void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
 };
diff --git a/video/end_to_end_tests/probing_tests.cc b/video/end_to_end_tests/probing_tests.cc
index 80c9ebd..dba3c3e 100644
--- a/video/end_to_end_tests/probing_tests.cc
+++ b/video/end_to_end_tests/probing_tests.cc
@@ -15,9 +15,21 @@
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
-namespace webrtc {
 
-class ProbingEndToEndTest : public test::CallTest {};
+namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+};
+}  // namespace
+
+class ProbingEndToEndTest : public test::CallTest {
+ public:
+  ProbingEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+  }
+};
 
 class ProbingTest : public test::EndToEndTest {
  public:
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index 7d31aed..49c3fb4 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -21,9 +21,18 @@
 #include "test/rtcp_packet_parser.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kVideoRotationExtensionId = 1,
+};
+}  // namespace
+
 class RetransmissionEndToEndTest : public test::CallTest {
  public:
-  RetransmissionEndToEndTest() = default;
+  RetransmissionEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+                                      kVideoRotationExtensionId));
+  }
 
  protected:
   void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index bef15d2..d89a024 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -18,6 +18,11 @@
 #include "test/rtcp_packet_parser.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+};
+}  // namespace
 
 class RtpRtcpEndToEndTest : public test::CallTest {
  protected:
@@ -530,7 +535,7 @@
     flexfec_receive_config.transport_cc = true;
     flexfec_receive_config.rtp_header_extensions.emplace_back(
         RtpExtension::kTransportSequenceNumberUri,
-        test::kTransportSequenceNumberExtensionId);
+        kTransportSequenceNumberExtensionId);
     flexfec_receive_configs_.push_back(flexfec_receive_config);
 
     CreateFlexfecStreams();
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index 53b181a..bad04b4 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -24,7 +24,19 @@
 #include "test/rtcp_packet_parser.h"
 
 namespace webrtc {
-class StatsEndToEndTest : public test::CallTest {};
+namespace {
+enum : int {  // The first valid value is 1.
+  kVideoContentTypeExtensionId = 1,
+};
+}  // namespace
+
+class StatsEndToEndTest : public test::CallTest {
+ public:
+  StatsEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
+                                      kVideoContentTypeExtensionId));
+  }
+};
 
 TEST_F(StatsEndToEndTest, GetStats) {
   static const int kStartBitrateBps = 3000000;
diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc
index 9aacd9a..b607bf6 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -21,12 +21,21 @@
 #include "video/end_to_end_tests/multi_stream_tester.h"
 
 namespace webrtc {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+};
+}  // namespace
 
-class TransportFeedbackEndToEndTest : public test::CallTest {};
+class TransportFeedbackEndToEndTest : public test::CallTest {
+ public:
+  TransportFeedbackEndToEndTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+  }
+};
 
 TEST_F(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
-  static const int kExtensionId = 5;
-
   class RtpExtensionHeaderObserver : public test::DirectTransport {
    public:
     RtpExtensionHeaderObserver(
@@ -50,7 +59,7 @@
           retransmit_observed_(false),
           started_(false) {
       parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
-                                          kExtensionId);
+                                          kTransportSequenceNumberExtensionId);
     }
     virtual ~RtpExtensionHeaderObserver() {}
 
@@ -174,8 +183,9 @@
         VideoEncoderConfig* encoder_config,
         test::FrameGeneratorCapturer** frame_generator) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                       kTransportSequenceNumberExtensionId));
 
       // Force some padding to be sent. Note that since we do send media
       // packets we can not guarantee that a padding only packet is sent.
@@ -201,8 +211,9 @@
         VideoReceiveStream::Config* receive_config) override {
       receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       receive_config->rtp.extensions.clear();
-      receive_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      receive_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                       kTransportSequenceNumberExtensionId));
       receive_config->renderer = &fake_renderer_;
     }
 
@@ -291,14 +302,14 @@
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
     send_config->rtp.extensions.clear();
     send_config->rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                     kTransportSequenceNumberExtensionId));
     (*receive_configs)[0].rtp.extensions.clear();
     (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
   }
 
  private:
-  static const int kExtensionId = 5;
   const bool feedback_enabled_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
@@ -426,7 +437,6 @@
 
 TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) {
   test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
-  static constexpr int kExtensionId = 8;
   static constexpr size_t kMinPacketsToWaitFor = 50;
   class TransportSequenceNumberTest : public test::EndToEndTest {
    public:
@@ -435,7 +445,7 @@
           video_observed_(false),
           audio_observed_(false) {
       parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
-                                          kExtensionId);
+                                          kTransportSequenceNumberExtensionId);
     }
 
     size_t GetNumVideoStreams() const override { return 1; }
@@ -445,8 +455,9 @@
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                       kTransportSequenceNumberExtensionId));
       (*receive_configs)[0].rtp.extensions.clear();
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     }
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index ad3224d..4941827 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -47,6 +47,14 @@
 namespace webrtc {
 
 namespace {
+enum : int {  // The first valid value is 1.
+  kAbsSendTimeExtensionId = 1,
+  kGenericFrameDescriptorExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoContentTypeExtensionId,
+  kVideoTimingExtensionId,
+};
+
 constexpr char kSyncGroup[] = "av_sync";
 constexpr int kOpusMinBitrateBps = 6000;
 constexpr int kOpusBitrateFbBps = 32000;
@@ -710,10 +718,10 @@
     if (params_.call.send_side_bwe) {
       video_send_configs_[video_idx].rtp.extensions.emplace_back(
           RtpExtension::kTransportSequenceNumberUri,
-          test::kTransportSequenceNumberExtensionId);
+          kTransportSequenceNumberExtensionId);
     } else {
       video_send_configs_[video_idx].rtp.extensions.emplace_back(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId);
+          RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId);
     }
 
     if (params_.call.generic_descriptor) {
@@ -724,13 +732,13 @@
 
       video_send_configs_[video_idx].rtp.extensions.emplace_back(
           RtpExtension::kGenericFrameDescriptorUri,
-          test::kGenericDescriptorExtensionId);
+          kGenericFrameDescriptorExtensionId);
     }
 
     video_send_configs_[video_idx].rtp.extensions.emplace_back(
-        RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId);
+        RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId);
     video_send_configs_[video_idx].rtp.extensions.emplace_back(
-        RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId);
+        RtpExtension::kVideoTimingUri, kVideoTimingExtensionId);
 
     video_encoder_configs_[video_idx].video_format.name =
         params_.video[video_idx].codec;
@@ -878,10 +886,10 @@
     if (params_.call.send_side_bwe) {
       GetFlexFecConfig()->rtp_header_extensions.push_back(
           RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
+                       kTransportSequenceNumberExtensionId));
     } else {
-      GetFlexFecConfig()->rtp_header_extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      GetFlexFecConfig()->rtp_header_extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     }
   }
 
@@ -919,10 +927,10 @@
     if (params_.call.send_side_bwe) {
       thumbnail_send_config.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
+                       kTransportSequenceNumberExtensionId));
     } else {
-      thumbnail_send_config.rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      thumbnail_send_config.rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     }
 
     VideoEncoderConfig thumbnail_encoder_config;
@@ -1354,7 +1362,7 @@
   if (params_.call.send_side_bwe) {
     audio_send_config.rtp.extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
-                             test::kTransportSequenceNumberExtensionId));
+                             kTransportSequenceNumberExtensionId));
     audio_send_config.min_bitrate_bps = kOpusMinBitrateBps;
     audio_send_config.max_bitrate_bps = kOpusBitrateFbBps;
     audio_send_config.send_codec_spec->transport_cc_enabled = true;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 057fd95..0beefec 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -73,6 +73,15 @@
 }  // namespace test
 
 namespace {
+enum : int {  // The first valid value is 1.
+  kAbsSendTimeExtensionId = 1,
+  kTimestampOffsetExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoContentTypeExtensionId,
+  kVideoRotationExtensionId,
+  kVideoTimingExtensionId,
+};
+
 constexpr int64_t kRtcpIntervalMs = 1000;
 
 enum VideoFormat {
@@ -84,6 +93,12 @@
 VideoFrame CreateVideoFrame(int width, int height, uint8_t data);
 
 class VideoSendStreamTest : public test::CallTest {
+ public:
+  VideoSendStreamTest() {
+    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                                      kTransportSequenceNumberExtensionId));
+  }
+
  protected:
   void TestNackRetransmission(uint32_t retransmit_ssrc,
                               uint8_t retransmit_payload_type);
@@ -163,7 +178,7 @@
    public:
     AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionAbsoluteSendTime, test::kAbsSendTimeExtensionId));
+          kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -193,8 +208,8 @@
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     }
 
     void PerformTest() override {
@@ -215,7 +230,7 @@
                 Clock::GetRealTimeClock(), kEncodeDelayMs);
           }) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
+          kRtpExtensionTransmissionTimeOffset, kTimestampOffsetExtensionId));
     }
 
    private:
@@ -239,7 +254,7 @@
       send_config->encoder_settings.encoder_factory = &encoder_factory_;
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
+          RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
     }
 
     void PerformTest() override {
@@ -253,7 +268,7 @@
 }
 
 TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
-  static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+  static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
   class TransportWideSequenceNumberObserver : public test::SendTest {
    public:
     TransportWideSequenceNumberObserver()
@@ -301,7 +316,7 @@
    public:
     VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionVideoRotation, test::kVideoRotationExtensionId));
+          kRtpExtensionVideoRotation, kVideoRotationExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -322,7 +337,7 @@
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kVideoRotationUri, test::kVideoRotationExtensionId));
+          RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
     }
 
     void OnFrameGeneratorCapturerCreated(
@@ -344,7 +359,7 @@
     VideoContentTypeObserver()
         : SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionVideoContentType, test::kVideoContentTypeExtensionId));
+          kRtpExtensionVideoContentType, kVideoContentTypeExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -367,9 +382,8 @@
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kVideoContentTypeUri,
-                       test::kVideoContentTypeExtensionId));
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
       encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
     }
 
@@ -389,8 +403,8 @@
    public:
     VideoTimingObserver()
         : SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
-      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionVideoTiming, test::kVideoTimingExtensionId));
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionVideoTiming,
+                                                      kVideoTimingExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -412,8 +426,8 @@
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kVideoTimingUri, kVideoTimingExtensionId));
     }
 
     void PerformTest() override {
@@ -464,7 +478,12 @@
         expect_ulpfec_(expect_ulpfec),
         sent_media_(false),
         sent_ulpfec_(false),
-        header_extensions_enabled_(header_extensions_enabled) {}
+        header_extensions_enabled_(header_extensions_enabled) {
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                        kAbsSendTimeExtensionId);
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                        kTransportSequenceNumberExtensionId);
+  }
 
   // Some of the test cases are expected to time out and thus we are using
   // a shorter timeout window than the default here.
@@ -564,12 +583,11 @@
         VideoSendStreamTest::kRedPayloadType;
     send_config->rtp.ulpfec.ulpfec_payload_type =
         VideoSendStreamTest::kUlpfecPayloadType;
-    EXPECT_FALSE(send_config->rtp.extensions.empty());
     if (!header_extensions_enabled_) {
       send_config->rtp.extensions.clear();
     } else {
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     }
     encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
     (*receive_configs)[0].rtp.red_payload_type =
@@ -687,7 +705,14 @@
         sent_media_(false),
         sent_flexfec_(false),
         header_extensions_enabled_(header_extensions_enabled),
-        num_video_streams_(num_video_streams) {}
+        num_video_streams_(num_video_streams) {
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                        kAbsSendTimeExtensionId);
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
+                                        kTimestampOffsetExtensionId);
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                        kTransportSequenceNumberExtensionId);
+  }
 
   size_t GetNumFlexfecStreams() const override { return 1; }
   size_t GetNumVideoStreams() const override { return num_video_streams_; }
@@ -751,10 +776,10 @@
     send_config->encoder_settings.encoder_factory = encoder_factory_;
     send_config->rtp.payload_name = payload_name_;
     if (header_extensions_enabled_) {
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
       send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
+          RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
     } else {
       send_config->rtp.extensions.clear();
     }
@@ -1628,7 +1653,7 @@
 TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
   static const int kStartBitrateBps = 300000;
   static const int kNewMaxBitrateBps = 1234567;
-  static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+  static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
   class ChangingNetworkRouteTest : public test::EndToEndTest {
    public:
     explicit ChangingNetworkRouteTest(