diff --git a/api/DEPS b/api/DEPS
index bc807048f7..f5a0201b18 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -150,6 +150,10 @@ specific_include_rules = {
     "+rtc_base/logging.h",
   ],
 
+  "rtp_parameters\.h": [
+    "+rtc_base/deprecation.h",
+  ],
+
   "rtp_receiver_interface\.h": [
     "+rtc_base/ref_count.h",
   ],
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index 90e4a4cf1c..6138c37320 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -18,6 +18,7 @@
 
 #include "absl/types/optional.h"
 #include "api/media_types.h"
+#include "rtc_base/deprecation.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -258,53 +259,64 @@ struct RtpExtension {
   // Header extension for audio levels, as defined in:
   // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
   static const char kAudioLevelUri[];
-  static const int kAudioLevelDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kAudioLevelDefaultId;
 
   // Header extension for RTP timestamp offset, see RFC 5450 for details:
   // http://tools.ietf.org/html/rfc5450
   static const char kTimestampOffsetUri[];
-  static const int kTimestampOffsetDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kTimestampOffsetDefaultId;
 
   // Header extension for absolute send time, see url for details:
   // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   static const char kAbsSendTimeUri[];
-  static const int kAbsSendTimeDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kAbsSendTimeDefaultId;
 
   // Header extension for coordination of video orientation, see url for
   // details:
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
   static const char kVideoRotationUri[];
-  static const int kVideoRotationDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kVideoRotationDefaultId;
 
   // Header extension for video content type. E.g. default or screenshare.
   static const char kVideoContentTypeUri[];
-  static const int kVideoContentTypeDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kVideoContentTypeDefaultId;
 
   // Header extension for video timing.
   static const char kVideoTimingUri[];
-  static const int kVideoTimingDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kVideoTimingDefaultId;
 
   // Header extension for video frame marking.
   static const char kFrameMarkingUri[];
-  static const int kFrameMarkingDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kFrameMarkingDefaultId;
 
   // Experimental codec agnostic frame descriptor.
   static const char kGenericFrameDescriptorUri[];
-  static const int kGenericFrameDescriptorDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kGenericFrameDescriptorDefaultId;
 
   // Header extension for transport sequence number, see url for details:
   // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
   static const char kTransportSequenceNumberUri[];
   static const char kTransportSequenceNumberV2Uri[];
-  static const int kTransportSequenceNumberDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kTransportSequenceNumberDefaultId;
 
   static const char kPlayoutDelayUri[];
-  static const int kPlayoutDelayDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kPlayoutDelayDefaultId;
 
   // Header extension for identifying media section within a transport.
   // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-49#section-15
   static const char kMidUri[];
-  static const int kMidDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kMidDefaultId;
 
   // Encryption of Header Extensions, see RFC 6904 for details:
   // https://tools.ietf.org/html/rfc6904
@@ -312,15 +324,18 @@ struct RtpExtension {
 
   // Header extension for color space information.
   static const char kColorSpaceUri[];
-  static const int kColorSpaceDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kColorSpaceDefaultId;
 
   // Header extension for RIDs and Repaired RIDs
   // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
   // https://tools.ietf.org/html/draft-ietf-mmusic-rid-15
   static const char kRidUri[];
-  static const int kRidDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kRidDefaultId;
   static const char kRepairedRidUri[];
-  static const int kRepairedRidDefaultId;
+  // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated.
+  RTC_DEPRECATED static const int kRepairedRidDefaultId;
 
   // Inclusive min and max IDs for two-byte header extensions and one-byte
   // header extensions, per RFC8285 Section 4.2-4.3.
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 6f6a198238..d9ac876d59 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -919,22 +919,26 @@ ParsedRtcEventLog::LoggedRtpStreamView::LoggedRtpStreamView(
 //             audio streams. Tracking bug: webrtc:6399
 webrtc::RtpHeaderExtensionMap
 ParsedRtcEventLog::GetDefaultHeaderExtensionMap() {
+  // Values from before the default RTP header extension IDs were removed.
+  constexpr int kAudioLevelDefaultId = 1;
+  constexpr int kTimestampOffsetDefaultId = 2;
+  constexpr int kAbsSendTimeDefaultId = 3;
+  constexpr int kVideoRotationDefaultId = 4;
+  constexpr int kTransportSequenceNumberDefaultId = 5;
+  constexpr int kPlayoutDelayDefaultId = 6;
+  constexpr int kVideoContentTypeDefaultId = 7;
+  constexpr int kVideoTimingDefaultId = 8;
+
   webrtc::RtpHeaderExtensionMap default_map;
-  default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
-  default_map.Register<TransmissionOffset>(
-      webrtc::RtpExtension::kTimestampOffsetDefaultId);
-  default_map.Register<AbsoluteSendTime>(
-      webrtc::RtpExtension::kAbsSendTimeDefaultId);
-  default_map.Register<VideoOrientation>(
-      webrtc::RtpExtension::kVideoRotationDefaultId);
-  default_map.Register<VideoContentTypeExtension>(
-      webrtc::RtpExtension::kVideoContentTypeDefaultId);
-  default_map.Register<VideoTimingExtension>(
-      webrtc::RtpExtension::kVideoTimingDefaultId);
+  default_map.Register<AudioLevel>(kAudioLevelDefaultId);
+  default_map.Register<TransmissionOffset>(kTimestampOffsetDefaultId);
+  default_map.Register<AbsoluteSendTime>(kAbsSendTimeDefaultId);
+  default_map.Register<VideoOrientation>(kVideoRotationDefaultId);
   default_map.Register<TransportSequenceNumber>(
-      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
-  default_map.Register<PlayoutDelayLimits>(
-      webrtc::RtpExtension::kPlayoutDelayDefaultId);
+      kTransportSequenceNumberDefaultId);
+  default_map.Register<PlayoutDelayLimits>(kPlayoutDelayDefaultId);
+  default_map.Register<VideoContentTypeExtension>(kVideoContentTypeDefaultId);
+  default_map.Register<VideoTimingExtension>(kVideoTimingDefaultId);
   return default_map;
 }
 
diff --git a/logging/rtc_event_log/rtc_event_log_unittest.cc b/logging/rtc_event_log/rtc_event_log_unittest.cc
index f44f53e83c..103789d420 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -52,20 +52,6 @@ namespace webrtc {
 
 namespace {
 
-RtpHeaderExtensionMap ExtensionMapWithAllSupportedExtensions() {
-  RtpHeaderExtensionMap all_extensions;
-  all_extensions.Register<AudioLevel>(RtpExtension::kAudioLevelDefaultId);
-  all_extensions.Register<TransmissionOffset>(
-      RtpExtension::kTimestampOffsetDefaultId);
-  all_extensions.Register<AbsoluteSendTime>(
-      RtpExtension::kAbsSendTimeDefaultId);
-  all_extensions.Register<VideoOrientation>(
-      RtpExtension::kVideoRotationDefaultId);
-  all_extensions.Register<TransportSequenceNumber>(
-      RtpExtension::kTransportSequenceNumberDefaultId);
-  return all_extensions;
-}
-
 struct EventCounts {
   size_t audio_send_streams = 0;
   size_t audio_recv_streams = 0;
@@ -247,7 +233,7 @@ void RtcEventLogSession::WriteVideoRecvConfigs(size_t video_recv_streams,
   // Force least one stream to use all header extensions, to ensure
   // (statistically) that every extension is tested in packet creation.
   RtpHeaderExtensionMap all_extensions =
-      ExtensionMapWithAllSupportedExtensions();
+      ParsedRtcEventLog::GetDefaultHeaderExtensionMap();
 
   clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
   uint32_t ssrc = prng_.Rand<uint32_t>();
@@ -276,7 +262,7 @@ void RtcEventLogSession::WriteVideoSendConfigs(size_t video_send_streams,
   // Force least one stream to use all header extensions, to ensure
   // (statistically) that every extension is tested in packet creation.
   RtpHeaderExtensionMap all_extensions =
-      ExtensionMapWithAllSupportedExtensions();
+      ParsedRtcEventLog::GetDefaultHeaderExtensionMap();
 
   clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
   uint32_t ssrc = prng_.Rand<uint32_t>();
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 5d8a173656..88f900045c 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -467,37 +467,28 @@ std::vector<VideoCodec> WebRtcVideoEngine::codecs() const {
 
 RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
   RtpCapabilities capabilities;
+  int id = 1;
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri,
-                           webrtc::RtpExtension::kTimestampOffsetDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
-                           webrtc::RtpExtension::kAbsSendTimeDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri,
-                           webrtc::RtpExtension::kVideoRotationDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri, id++));
   capabilities.header_extensions.push_back(webrtc::RtpExtension(
-      webrtc::RtpExtension::kTransportSequenceNumberUri,
-      webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
+      webrtc::RtpExtension::kTransportSequenceNumberUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri,
-                           webrtc::RtpExtension::kPlayoutDelayDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri,
-                           webrtc::RtpExtension::kVideoContentTypeDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
-                           webrtc::RtpExtension::kVideoTimingDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kFrameMarkingUri,
-                           webrtc::RtpExtension::kFrameMarkingDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kFrameMarkingUri, id++));
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kColorSpaceUri,
-                           webrtc::RtpExtension::kColorSpaceDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kColorSpaceUri, id++));
   if (webrtc::field_trial::IsEnabled("WebRTC-GenericDescriptorAdvertised")) {
     capabilities.header_extensions.push_back(webrtc::RtpExtension(
-        webrtc::RtpExtension::kGenericFrameDescriptorUri,
-        webrtc::RtpExtension::kGenericFrameDescriptorDefaultId));
+        webrtc::RtpExtension::kGenericFrameDescriptorUri, id++));
   }
 
   return capabilities;
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index ef45779c27..482da39fb4 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -243,7 +243,7 @@ class WebRtcVideoEngineTest : public ::testing::Test {
   VideoMediaChannel* SetRecvParamsWithSupportedCodecs(
       const std::vector<VideoCodec>& codecs);
 
-  void TestExtendedEncoderOveruse(bool use_external_encoder);
+  void ExpectRtpCapabilitySupport(const char* uri, bool supported) const;
 
   // Has to be the first one, so it is initialized before the call or there is a
   // race condition in the clock access.
@@ -278,43 +278,43 @@ TEST_F(WebRtcVideoEngineTest, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
 }
 
 TEST_F(WebRtcVideoEngineTest, SupportsTimestampOffsetHeaderExtension) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  EXPECT_THAT(
-      capabilities.header_extensions,
-      testing::Contains(RtpExtension(RtpExtension::kTimestampOffsetUri,
-                                     RtpExtension::kTimestampOffsetDefaultId)));
+  ExpectRtpCapabilitySupport(RtpExtension::kTimestampOffsetUri, true);
 }
 
 TEST_F(WebRtcVideoEngineTest, SupportsAbsoluteSenderTimeHeaderExtension) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  EXPECT_THAT(
-      capabilities.header_extensions,
-      testing::Contains(RtpExtension(RtpExtension::kAbsSendTimeUri,
-                                     RtpExtension::kAbsSendTimeDefaultId)));
+  ExpectRtpCapabilitySupport(RtpExtension::kAbsSendTimeUri, true);
 }
 
 TEST_F(WebRtcVideoEngineTest, SupportsTransportSequenceNumberHeaderExtension) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  EXPECT_THAT(capabilities.header_extensions,
-              testing::Contains(RtpExtension(
-                  RtpExtension::kTransportSequenceNumberUri,
-                  RtpExtension::kTransportSequenceNumberDefaultId)));
+  ExpectRtpCapabilitySupport(RtpExtension::kTransportSequenceNumberUri, true);
 }
 
 TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  EXPECT_THAT(
-      capabilities.header_extensions,
-      testing::Contains(RtpExtension(RtpExtension::kVideoRotationUri,
-                                     RtpExtension::kVideoRotationDefaultId)));
+  ExpectRtpCapabilitySupport(RtpExtension::kVideoRotationUri, true);
+}
+
+TEST_F(WebRtcVideoEngineTest, SupportsPlayoutDelayHeaderExtension) {
+  ExpectRtpCapabilitySupport(RtpExtension::kPlayoutDelayUri, true);
+}
+
+TEST_F(WebRtcVideoEngineTest, SupportsVideoContentTypeHeaderExtension) {
+  ExpectRtpCapabilitySupport(RtpExtension::kVideoContentTypeUri, true);
+}
+
+TEST_F(WebRtcVideoEngineTest, SupportsVideoTimingHeaderExtension) {
+  ExpectRtpCapabilitySupport(RtpExtension::kVideoTimingUri, true);
+}
+
+TEST_F(WebRtcVideoEngineTest, SupportsFrameMarkingHeaderExtension) {
+  ExpectRtpCapabilitySupport(RtpExtension::kFrameMarkingUri, true);
 }
 
 TEST_F(WebRtcVideoEngineTest, SupportsColorSpaceHeaderExtension) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  EXPECT_THAT(
-      capabilities.header_extensions,
-      testing::Contains(RtpExtension(RtpExtension::kColorSpaceUri,
-                                     RtpExtension::kColorSpaceDefaultId)));
+  ExpectRtpCapabilitySupport(RtpExtension::kColorSpaceUri, true);
+}
+
+TEST_F(WebRtcVideoEngineTest, AdvertiseGenericDescriptor) {
+  ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri, false);
 }
 
 class WebRtcVideoEngineTestWithGenericDescriptor
@@ -325,15 +325,7 @@ class WebRtcVideoEngineTestWithGenericDescriptor
 };
 
 TEST_F(WebRtcVideoEngineTestWithGenericDescriptor, AdvertiseGenericDescriptor) {
-  RtpCapabilities capabilities = engine_.GetCapabilities();
-  ASSERT_FALSE(capabilities.header_extensions.empty());
-  for (const RtpExtension& extension : capabilities.header_extensions) {
-    if (extension.uri == RtpExtension::kGenericFrameDescriptorUri) {
-      EXPECT_EQ(RtpExtension::kGenericFrameDescriptorDefaultId, extension.id);
-      return;
-    }
-  }
-  FAIL() << "Generic descriptor extension not in header-extension list.";
+  ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri, true);
 }
 
 TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
@@ -702,6 +694,19 @@ VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs(
   return channel;
 }
 
+void WebRtcVideoEngineTest::ExpectRtpCapabilitySupport(const char* uri,
+                                                       bool supported) const {
+  const RtpCapabilities capabilities = engine_.GetCapabilities();
+  if (supported) {
+    EXPECT_THAT(capabilities.header_extensions,
+                testing::Contains(testing::Field(&RtpExtension::uri, uri)));
+  } else {
+    EXPECT_THAT(
+        capabilities.header_extensions,
+        testing::Each(testing::Field(&RtpExtension::uri, testing::StrNe(uri))));
+  }
+}
+
 TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) {
   encoder_factory_->AddSupportedVideoCodecType("VP8");
 
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 691f7d7bee..07a47d789c 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -556,13 +556,12 @@ const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   RtpCapabilities capabilities;
+  int id = 1;
   capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
-                           webrtc::RtpExtension::kAudioLevelDefaultId));
+      webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, id++));
   if (allocation_settings_.EnableTransportSequenceNumberExtension()) {
     capabilities.header_extensions.push_back(webrtc::RtpExtension(
-        webrtc::RtpExtension::kTransportSequenceNumberUri,
-        webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
+        webrtc::RtpExtension::kTransportSequenceNumberUri, id++));
   }
 
   return capabilities;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 663dd8caba..a3ccee94b9 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -35,13 +35,14 @@
 #include "test/mock_audio_decoder_factory.h"
 #include "test/mock_audio_encoder_factory.h"
 
-using testing::_;
-using testing::ContainerEq;
-using testing::Field;
-using testing::Return;
-using testing::ReturnPointee;
-using testing::SaveArg;
-using testing::StrictMock;
+using ::testing::_;
+using ::testing::ContainerEq;
+using ::testing::Contains;
+using ::testing::Field;
+using ::testing::Return;
+using ::testing::ReturnPointee;
+using ::testing::SaveArg;
+using ::testing::StrictMock;
 
 namespace {
 using webrtc::BitrateConstraints;
@@ -2018,16 +2019,11 @@ class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake {
 
 TEST_F(WebRtcVoiceEngineWithSendSideBweTest,
        SupportsTransportSequenceNumberHeaderExtension) {
-  cricket::RtpCapabilities capabilities = engine_->GetCapabilities();
-  ASSERT_FALSE(capabilities.header_extensions.empty());
-  for (const webrtc::RtpExtension& extension : capabilities.header_extensions) {
-    if (extension.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) {
-      EXPECT_EQ(webrtc::RtpExtension::kTransportSequenceNumberDefaultId,
-                extension.id);
-      return;
-    }
-  }
-  FAIL() << "Transport sequence number extension not in header-extension list.";
+  const cricket::RtpCapabilities capabilities = engine_->GetCapabilities();
+  EXPECT_THAT(capabilities.header_extensions,
+              Contains(testing::Field(
+                  "uri", &RtpExtension::uri,
+                  webrtc::RtpExtension::kTransportSequenceNumberUri)));
 }
 
 // Test support for audio level header extension.
diff --git a/pc/media_session.cc b/pc/media_session.cc
index e18c86a249..19eb769dfc 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -10,6 +10,7 @@
 
 #include "pc/media_session.h"
 
+#include <algorithm>
 #include <functional>
 #include <map>
 #include <memory>
@@ -51,6 +52,7 @@ void GetSupportedSdesCryptoSuiteNames(
     names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
   }
 }
+
 }  // namespace
 
 namespace cricket {
@@ -1347,14 +1349,21 @@ void MediaSessionDescriptionFactory::set_audio_codecs(
 
 static void AddUnifiedPlanExtensions(RtpHeaderExtensions* extensions) {
   RTC_DCHECK(extensions);
+
+  rtc::UniqueNumberGenerator<int> unique_id_generator;
+  unique_id_generator.AddKnownId(0);  // The first valid RTP extension ID is 1.
+  for (const webrtc::RtpExtension& extension : *extensions) {
+    const bool collision_free = unique_id_generator.AddKnownId(extension.id);
+    RTC_DCHECK(collision_free);
+  }
+
   // Unified Plan also offers the MID and RID header extensions.
+  extensions->push_back(webrtc::RtpExtension(webrtc::RtpExtension::kMidUri,
+                                             unique_id_generator()));
+  extensions->push_back(webrtc::RtpExtension(webrtc::RtpExtension::kRidUri,
+                                             unique_id_generator()));
   extensions->push_back(webrtc::RtpExtension(
-      webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
-  extensions->push_back(webrtc::RtpExtension(
-      webrtc::RtpExtension::kRidUri, webrtc::RtpExtension::kRidDefaultId));
-  extensions->push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kRepairedRidUri,
-                           webrtc::RtpExtension::kRepairedRidDefaultId));
+      webrtc::RtpExtension::kRepairedRidUri, unique_id_generator()));
 }
 
 RtpHeaderExtensions
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index b0409eab11..3f7631eccf 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -4118,6 +4118,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, VideoHasRidExtensionsInUnifiedPlan) {
   cricket::RtpHeaderExtensions result = sf.video_rtp_header_extensions();
   // Check to see that RID extensions were added to the extension list
   EXPECT_GE(result.size(), 2u);
+  EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
+                                     RtpExtension::kMidUri)));
   EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
                                      RtpExtension::kRidUri)));
   EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
@@ -4138,6 +4140,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, AudioHasRidExtensionsInUnifiedPlan) {
   cricket::RtpHeaderExtensions result = sf.audio_rtp_header_extensions();
   // Check to see that RID extensions were added to the extension list
   EXPECT_GE(result.size(), 2u);
+  EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
+                                     RtpExtension::kMidUri)));
   EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
                                      RtpExtension::kRidUri)));
   EXPECT_THAT(result, Contains(Field("uri", &RtpExtension::uri,
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 67d9d4b499..cb02610f37 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -553,6 +553,7 @@ if (rtc_include_tests) {
       "../modules/video_coding:webrtc_multiplex",
       "../modules/video_coding:webrtc_vp8",
       "../modules/video_coding:webrtc_vp9",
+      "../rtc_base",
       "../rtc_base:checks",
       "../rtc_base:rate_limiter",
       "../rtc_base:rtc_base_approved",
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 41c039710b..a517064cf2 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -359,9 +359,9 @@ TEST_F(VideoSendStreamImplTest, SetsScreensharePacingFactorWithFeedback) {
   test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
 
   test_queue_.SendTask([this] {
+    constexpr int kId = 1;
     config_.rtp.extensions.emplace_back(
-        RtpExtension::kTransportSequenceNumberUri,
-        RtpExtension::kTransportSequenceNumberDefaultId);
+        RtpExtension::kTransportSequenceNumberUri, kId);
     EXPECT_CALL(transport_controller_,
                 SetPacingFactor(kAlrProbingExperimentPaceMultiplier))
         .Times(1);
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 8338d84205..bd5a03ca30 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -34,6 +34,7 @@
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/time_utils.h"
+#include "rtc_base/unique_id_generator.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
 #include "test/configurable_frame_size_encoder.h"
@@ -3752,10 +3753,14 @@ class PacingFactorObserver : public test::SendTest {
     }
 
     if (configure_send_side_ && !has_send_side) {
+      rtc::UniqueNumberGenerator<int> unique_id_generator;
+      unique_id_generator.AddKnownId(0);  // First valid RTP extension ID is 1.
+      for (const RtpExtension& extension : send_config->rtp.extensions) {
+        unique_id_generator.AddKnownId(extension.id);
+      }
       // Want send side, not present by default, so add it.
       send_config->rtp.extensions.emplace_back(
-          RtpExtension::kTransportSequenceNumberUri,
-          RtpExtension::kTransportSequenceNumberDefaultId);
+          RtpExtension::kTransportSequenceNumberUri, unique_id_generator());
     }
 
     // ALR only enabled for screenshare.