diff --git a/third_party/libwebrtc/README.moz-ff-commit b/third_party/libwebrtc/README.moz-ff-commit
index bc7497475177..e4851689f927 100644
--- a/third_party/libwebrtc/README.moz-ff-commit
+++ b/third_party/libwebrtc/README.moz-ff-commit
@@ -8508,3 +8508,6 @@ f1384afda04
 # MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 dc8fc722048
+# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+d0321c5e5ab
diff --git a/third_party/libwebrtc/README.mozilla b/third_party/libwebrtc/README.mozilla
index 670fd2d559ea..62ea2773abd6 100644
--- a/third_party/libwebrtc/README.mozilla
+++ b/third_party/libwebrtc/README.mozilla
@@ -5678,3 +5678,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-08T00:38:14.086550.
 # python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-08T00:39:02.686688.
+# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
+libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-08T00:53:14.131533.
diff --git a/third_party/libwebrtc/api/rtp_parameters.cc b/third_party/libwebrtc/api/rtp_parameters.cc
index f1ef51d99463..2a119cf35147 100644
--- a/third_party/libwebrtc/api/rtp_parameters.cc
+++ b/third_party/libwebrtc/api/rtp_parameters.cc
@@ -114,7 +114,6 @@ std::string RtpExtension::ToString() const {
 
 constexpr char RtpExtension::kEncryptHeaderExtensionsUri[];
 constexpr char RtpExtension::kAudioLevelUri[];
-constexpr char RtpExtension::kCsrcAudioLevelUri[];
 constexpr char RtpExtension::kTimestampOffsetUri[];
 constexpr char RtpExtension::kAbsSendTimeUri[];
 constexpr char RtpExtension::kAbsoluteCaptureTimeUri[];
@@ -132,6 +131,7 @@ constexpr char RtpExtension::kMidUri[];
 constexpr char RtpExtension::kRidUri[];
 constexpr char RtpExtension::kRepairedRidUri[];
 constexpr char RtpExtension::kVideoFrameTrackingIdUri[];
+constexpr char RtpExtension::kCsrcAudioLevelsUri[];
 
 constexpr int RtpExtension::kMinId;
 constexpr int RtpExtension::kMaxId;
@@ -141,14 +141,14 @@ constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize;
 
 bool RtpExtension::IsSupportedForAudio(absl::string_view uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
-         uri == webrtc::RtpExtension::kCsrcAudioLevelUri;
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
          uri == webrtc::RtpExtension::kMidUri ||
          uri == webrtc::RtpExtension::kRidUri ||
-         uri == webrtc::RtpExtension::kRepairedRidUri;
+         uri == webrtc::RtpExtension::kRepairedRidUri ||
+         uri == webrtc::RtpExtension::kCsrcAudioLevelsUri;
 }
 
 bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
diff --git a/third_party/libwebrtc/audio/audio_send_stream.cc b/third_party/libwebrtc/audio/audio_send_stream.cc
index a41d06edbb9f..b3d555293b7a 100644
--- a/third_party/libwebrtc/audio/audio_send_stream.cc
+++ b/third_party/libwebrtc/audio/audio_send_stream.cc
@@ -272,7 +272,7 @@ void AudioSendStream::ConfigureStream(
     rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(
         kRtpExtensionAbsoluteSendTime);
     if (new_ids.abs_send_time) {
-      rtp_rtcp_module_->RegisterRtpHeaderExtension(AbsoluteSendTime::kUri,
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(AbsoluteSendTime::Uri(),
                                                    new_ids.abs_send_time);
     }
   }
@@ -290,7 +290,7 @@ void AudioSendStream::ConfigureStream(
     if (!allocate_audio_without_feedback_ &&
         new_ids.transport_sequence_number != 0) {
       rtp_rtcp_module_->RegisterRtpHeaderExtension(
-          TransportSequenceNumber::kUri, new_ids.transport_sequence_number);
+          TransportSequenceNumber::Uri(), new_ids.transport_sequence_number);
       // Probing in application limited region is only used in combination with
       // send side congestion control, wich depends on feedback packets which
       // requires transport sequence numbers to be enabled.
@@ -308,7 +308,7 @@ void AudioSendStream::ConfigureStream(
   if ((first_time || new_ids.mid != old_ids.mid ||
        new_config.rtp.mid != old_config.rtp.mid) &&
       new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
-    rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpMid::kUri, new_ids.mid);
+    rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpMid::Uri(), new_ids.mid);
     rtp_rtcp_module_->SetMid(new_config.rtp.mid);
   }
 
@@ -318,12 +318,12 @@ void AudioSendStream::ConfigureStream(
        new_config.rtp.rid != old_config.rtp.rid)) {
     if (new_ids.rid != 0 || new_ids.repaired_rid != 0) {
       if (new_config.rtp.rid.empty()) {
-        rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(RtpStreamId::kUri);
+        rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(RtpStreamId::Uri());
       } else if (new_ids.repaired_rid != 0) {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::kUri,
+        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
                                                      new_ids.repaired_rid);
       } else {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::kUri,
+        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
                                                      new_ids.rid);
       }
     }
@@ -335,7 +335,7 @@ void AudioSendStream::ConfigureStream(
         kRtpExtensionAbsoluteCaptureTime);
     if (new_ids.abs_capture_time) {
       rtp_rtcp_module_->RegisterRtpHeaderExtension(
-          AbsoluteCaptureTimeExtension::kUri, new_ids.abs_capture_time);
+          AbsoluteCaptureTimeExtension::Uri(), new_ids.abs_capture_time);
     }
   }
 
diff --git a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
index 58db5254fe7d..6ac5f4e8edcd 100644
--- a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
+++ b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
@@ -233,7 +233,7 @@ struct ConfigHelper {
         .WillRepeatedly(Return(&bandwidth_observer_));
     if (audio_bwe_enabled) {
       EXPECT_CALL(rtp_rtcp_,
-                  RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
+                  RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
                                              kTransportSequenceNumberId))
           .Times(1);
       EXPECT_CALL(*channel_send_,
@@ -802,7 +802,7 @@ TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
     ConfigHelper::AddBweToConfig(&new_config);
 
     EXPECT_CALL(*helper.rtp_rtcp(),
-                RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
+                RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
                                            kTransportSequenceNumberId))
         .Times(1);
     {
diff --git a/third_party/libwebrtc/audio/channel_send.cc b/third_party/libwebrtc/audio/channel_send.cc
index 34b5bb9c6557..3ab6c9a7301f 100644
--- a/third_party/libwebrtc/audio/channel_send.cc
+++ b/third_party/libwebrtc/audio/channel_send.cc
@@ -738,9 +738,9 @@ void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   _includeAudioLevelIndication = enable;
   if (enable) {
-    rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::kUri, id);
+    rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::Uri(), id);
   } else {
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::kUri);
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::Uri());
   }
 }
 
diff --git a/third_party/libwebrtc/call/flexfec_receive_stream_unittest.cc b/third_party/libwebrtc/call/flexfec_receive_stream_unittest.cc
index 312fe0c90738..9885971a8bde 100644
--- a/third_party/libwebrtc/call/flexfec_receive_stream_unittest.cc
+++ b/third_party/libwebrtc/call/flexfec_receive_stream_unittest.cc
@@ -66,7 +66,7 @@ TEST(FlexfecReceiveStreamConfigTest, IsCompleteAndEnabled) {
   config.rtp.local_ssrc = 18374743;
   config.rtcp_mode = RtcpMode::kCompound;
   config.rtp.transport_cc = true;
-  config.rtp.extensions.emplace_back(TransportSequenceNumber::kUri, 7);
+  config.rtp.extensions.emplace_back(TransportSequenceNumber::Uri(), 7);
   EXPECT_FALSE(config.IsCompleteAndEnabled());
 
   config.payload_type = 123;
diff --git a/third_party/libwebrtc/call/rtp_video_sender_unittest.cc b/third_party/libwebrtc/call/rtp_video_sender_unittest.cc
index e58ee3c5c04a..29ae31107e34 100644
--- a/third_party/libwebrtc/call/rtp_video_sender_unittest.cc
+++ b/third_party/libwebrtc/call/rtp_video_sender_unittest.cc
@@ -103,7 +103,7 @@ VideoSendStream::Config CreateVideoSendStreamConfig(
   config.rtp.nack.rtp_history_ms = 1000;
   config.rtp.extensions.emplace_back(RtpExtension::kTransportSequenceNumberUri,
                                      kTransportsSequenceExtensionId);
-  config.rtp.extensions.emplace_back(RtpDependencyDescriptorExtension::kUri,
+  config.rtp.extensions.emplace_back(RtpDependencyDescriptorExtension::Uri(),
                                      kDependencyDescriptorExtensionId);
   config.rtp.extmap_allow_mixed = true;
   return config;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
index a746d8a07642..3c2fa3d66f82 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
@@ -15,6 +15,7 @@
 
 #include <string>
 
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -35,7 +36,7 @@ class RtpHeaderExtensionMap {
 
   template <typename Extension>
   bool Register(int id) {
-    return Register(id, Extension::kId, Extension::kUri);
+    return Register(id, Extension::kId, Extension::Uri());
   }
   bool RegisterByType(int id, RTPExtensionType type);
   bool RegisterByUri(int id, absl::string_view uri);
@@ -64,7 +65,7 @@ class RtpHeaderExtensionMap {
   }
 
  private:
-  bool Register(int id, RTPExtensionType type, const char* uri);
+  bool Register(int id, RTPExtensionType type, absl::string_view uri);
 
   uint8_t ids_[kRtpExtensionNumberOfExtensions];
   bool extmap_allow_mixed_;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc
index f1fe71d19846..071829f1c0eb 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -47,13 +47,13 @@ RtpHeaderExtensionMap RegisterSupportedExtensions(
     const std::vector<RtpExtension>& rtp_header_extensions) {
   RtpHeaderExtensionMap map;
   for (const auto& extension : rtp_header_extensions) {
-    if (extension.uri == TransportSequenceNumber::kUri) {
+    if (extension.uri == TransportSequenceNumber::Uri()) {
       map.Register<TransportSequenceNumber>(extension.id);
-    } else if (extension.uri == AbsoluteSendTime::kUri) {
+    } else if (extension.uri == AbsoluteSendTime::Uri()) {
       map.Register<AbsoluteSendTime>(extension.id);
-    } else if (extension.uri == TransmissionOffset::kUri) {
+    } else if (extension.uri == TransmissionOffset::Uri()) {
       map.Register<TransmissionOffset>(extension.id);
-    } else if (extension.uri == RtpMid::kUri) {
+    } else if (extension.uri == RtpMid::Uri()) {
       map.Register<RtpMid>(extension.id);
     } else {
       RTC_LOG(LS_INFO)
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
index de16eeab2a36..5a89fae9cb58 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
@@ -13,7 +13,10 @@
 #include <bitset>
 #include <cstdint>
 
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
+#include "api/rtp_parameters.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
@@ -25,10 +28,12 @@ namespace webrtc {
 class RtpDependencyDescriptorExtension {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor02;
-  // TODO(bugs.webrtc.org/10342): Use uri from the spec when there is one.
-  static constexpr char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") char kUri[] =
       "https://aomediacodec.github.io/av1-rtp-spec/"
       "#dependency-descriptor-rtp-header-extension";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kDependencyDescriptorUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     const FrameDependencyStructure* structure,
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
index ac7afb489b66..f260a48df7ee 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
@@ -13,7 +13,10 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
+#include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
 
@@ -23,9 +26,12 @@ class RtpGenericFrameDescriptorExtension00 {
  public:
   using value_type = RtpGenericFrameDescriptor;
   static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor00;
-  static constexpr char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/"
       "generic-frame-descriptor-00";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kGenericFrameDescriptorUri00;
+  }
   static constexpr int kMaxSizeBytes = 16;
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
index ddd579344ff5..541e570ed882 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
@@ -10,6 +10,7 @@
 
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 
+#include "absl/strings/string_view.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -23,12 +24,12 @@ namespace {
 
 struct ExtensionInfo {
   RTPExtensionType type;
-  const char* uri;
+  absl::string_view uri;
 };
 
 template <typename Extension>
 constexpr ExtensionInfo CreateExtensionInfo() {
-  return {Extension::kId, Extension::kUri};
+  return {Extension::kId, Extension::Uri()};
 }
 
 constexpr ExtensionInfo kExtensions[] = {
@@ -135,7 +136,7 @@ void RtpHeaderExtensionMap::Deregister(absl::string_view uri) {
 
 bool RtpHeaderExtensionMap::Register(int id,
                                      RTPExtensionType type,
-                                     const char* uri) {
+                                     absl::string_view uri) {
   RTC_DCHECK_GT(type, kRtpExtensionNone);
   RTC_DCHECK_LT(type, kRtpExtensionNumberOfExtensions);
 
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
index f5bb1bb324a8..91bb2527732a 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
@@ -31,7 +31,7 @@ TEST(RtpHeaderExtensionTest, RegisterByType) {
 TEST(RtpHeaderExtensionTest, RegisterByUri) {
   RtpHeaderExtensionMap map;
 
-  EXPECT_TRUE(map.RegisterByUri(3, TransmissionOffset::kUri));
+  EXPECT_TRUE(map.RegisterByUri(3, TransmissionOffset::Uri()));
 
   EXPECT_TRUE(map.IsRegistered(TransmissionOffset::kId));
   EXPECT_EQ(3, map.GetId(TransmissionOffset::kId));
@@ -49,8 +49,8 @@ TEST(RtpHeaderExtensionTest, RegisterWithTrait) {
 }
 
 TEST(RtpHeaderExtensionTest, RegisterDuringContruction) {
-  const std::vector<RtpExtension> config = {{TransmissionOffset::kUri, 1},
-                                            {AbsoluteSendTime::kUri, 3}};
+  const std::vector<RtpExtension> config = {{TransmissionOffset::Uri(), 1},
+                                            {AbsoluteSendTime::Uri(), 3}};
   const RtpHeaderExtensionMap map(config);
 
   EXPECT_EQ(1, map.GetId(TransmissionOffset::kId));
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
index 478eb947fc02..55d282ff8b38 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -16,8 +16,11 @@
 #include <string>
 #include <vector>
 
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
 #include "api/video/color_space.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_rotation.h"
@@ -31,8 +34,11 @@ class AbsoluteSendTime {
   using value_type = uint32_t;
   static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAbsSendTimeUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data, uint32_t* time_24bits);
   static size_t ValueSize(uint32_t time_24bits) { return kValueSizeBytes; }
@@ -50,8 +56,11 @@ class AbsoluteCaptureTimeExtension {
   static constexpr uint8_t kValueSizeBytes = 16;
   static constexpr uint8_t kValueSizeBytesWithoutEstimatedCaptureClockOffset =
       8;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAbsoluteCaptureTimeUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     AbsoluteCaptureTime* extension);
@@ -64,8 +73,11 @@ class AudioLevel {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
   static constexpr uint8_t kValueSizeBytes = 1;
-  static constexpr const char kUri[] =
-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAudioLevelUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     bool* voice_activity,
@@ -83,8 +95,11 @@ class CsrcAudioLevel {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
   static constexpr uint8_t kMaxValueSizeBytes = 15;
-  static constexpr const char kUri[] =
-      "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kCsrcAudioLevelsUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     std::vector<uint8_t>* csrc_audio_levels);
@@ -99,7 +114,11 @@ class TransmissionOffset {
   using value_type = int32_t;
   static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static constexpr const char kUri[] = "urn:ietf:params:rtp-hdrext:toffset";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:toffset";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTimestampOffsetUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data, int32_t* rtp_time);
   static size_t ValueSize(int32_t rtp_time) { return kValueSizeBytes; }
@@ -111,9 +130,13 @@ class TransportSequenceNumber {
   using value_type = uint16_t;
   static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
   static constexpr uint8_t kValueSizeBytes = 2;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.ietf.org/id/"
       "draft-holmer-rmcat-transport-wide-cc-extensions-01";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTransportSequenceNumberUri;
+  }
+
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     uint16_t* transport_sequence_number);
   static size_t ValueSize(uint16_t /*transport_sequence_number*/) {
@@ -129,8 +152,12 @@ class TransportSequenceNumberV2 {
       kRtpExtensionTransportSequenceNumber02;
   static constexpr uint8_t kValueSizeBytes = 4;
   static constexpr uint8_t kValueSizeBytesWithoutFeedbackRequest = 2;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTransportSequenceNumberV2Uri;
+  }
+
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     uint16_t* transport_sequence_number,
                     absl::optional<FeedbackRequest>* feedback_request);
@@ -153,7 +180,11 @@ class VideoOrientation {
   using value_type = VideoRotation;
   static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
   static constexpr uint8_t kValueSizeBytes = 1;
-  static constexpr const char kUri[] = "urn:3gpp:video-orientation";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:3gpp:video-orientation";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoRotationUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data, VideoRotation* value);
   static size_t ValueSize(VideoRotation) { return kValueSizeBytes; }
@@ -168,8 +199,11 @@ class PlayoutDelayLimits {
   using value_type = VideoPlayoutDelay;
   static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kPlayoutDelayUri;
+  }
 
   // Playout delay in milliseconds. A playout delay limit (min or max)
   // has 12 bits allocated. This allows a range of 0-4095 values which
@@ -190,8 +224,11 @@ class VideoContentTypeExtension {
   using value_type = VideoContentType;
   static constexpr RTPExtensionType kId = kRtpExtensionVideoContentType;
   static constexpr uint8_t kValueSizeBytes = 1;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoContentTypeUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     VideoContentType* content_type);
@@ -205,8 +242,11 @@ class VideoTimingExtension {
   using value_type = VideoSendTiming;
   static constexpr RTPExtensionType kId = kRtpExtensionVideoTiming;
   static constexpr uint8_t kValueSizeBytes = 13;
-  static constexpr const char kUri[] =
-      "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoTimingUri;
+  }
 
   // Offsets of the fields in the RTP header extension, counting from the first
   // byte after the one-byte header.
@@ -239,8 +279,11 @@ class ColorSpaceExtension {
   static constexpr RTPExtensionType kId = kRtpExtensionColorSpace;
   static constexpr uint8_t kValueSizeBytes = 28;
   static constexpr uint8_t kValueSizeBytesWithoutHdrMetadata = 4;
-  static constexpr const char kUri[] =
-      "http://www.webrtc.org/experiments/rtp-hdrext/color-space";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/color-space";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kColorSpaceUri;
+  }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     ColorSpace* color_space);
@@ -303,21 +346,27 @@ class BaseRtpStringExtension {
 class RtpStreamId : public BaseRtpStringExtension {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
-  static constexpr const char kUri[] =
-      "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+  static constexpr absl::string_view Uri() { return RtpExtension::kRidUri; }
 };
 
 class RepairedRtpStreamId : public BaseRtpStringExtension {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionRepairedRtpStreamId;
-  static constexpr const char kUri[] =
-      "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kRepairedRidUri;
+  }
 };
 
 class RtpMid : public BaseRtpStringExtension {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionMid;
-  static constexpr const char kUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid";
+  static constexpr ABSL_DEPRECATED("Use Uri()") const
+      char kUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid";
+  static constexpr absl::string_view Uri() { return RtpExtension::kMidUri; }
 };
 
 class InbandComfortNoiseExtension {
@@ -328,6 +377,7 @@ class InbandComfortNoiseExtension {
   static constexpr uint8_t kValueSizeBytes = 1;
   static constexpr const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/inband-cn";
+  static constexpr absl::string_view Uri() { return kUri; }
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     absl::optional<uint8_t>* level);
@@ -343,8 +393,12 @@ class VideoFrameTrackingIdExtension {
   using value_type = uint16_t;
   static constexpr RTPExtensionType kId = kRtpExtensionVideoFrameTrackingId;
   static constexpr uint8_t kValueSizeBytes = 2;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/video-frame-tracking-id";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoFrameTrackingIdUri;
+  }
+
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     uint16_t* video_frame_tracking_id);
   static size_t ValueSize(uint16_t /*video_frame_tracking_id*/) {
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index 6d6008534582..21bf37c0c351 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -886,7 +886,7 @@ struct UncopyableValue {
 };
 struct UncopyableExtension {
   static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor02;
-  static constexpr char kUri[] = "uri";
+  static constexpr absl::string_view Uri() { return "uri"; }
 
   static size_t ValueSize(const UncopyableValue& value) { return 1; }
   static bool Write(rtc::ArrayView<uint8_t> data,
@@ -899,7 +899,6 @@ struct UncopyableExtension {
   }
 };
 constexpr RTPExtensionType UncopyableExtension::kId;
-constexpr char UncopyableExtension::kUri[];
 
 TEST(RtpPacketTest, SetUncopyableExtension) {
   RtpPacket::ExtensionManager extensions;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
index 599264e7c5df..8592e42173ed 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -957,7 +957,7 @@ TEST_P(RtpRtcpImpl2Test, PaddingTimestampMatchesMedia) {
 }
 
 TEST_P(RtpRtcpImpl2Test, AssignsTransportSequenceNumber) {
-  sender_.RegisterHeaderExtension(TransportSequenceNumber::kUri,
+  sender_.RegisterHeaderExtension(TransportSequenceNumber::Uri(),
                                   kTransportSequenceNumberExtensionId);
 
   EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
@@ -974,7 +974,7 @@ TEST_P(RtpRtcpImpl2Test, AssignsTransportSequenceNumber) {
 }
 
 TEST_P(RtpRtcpImpl2Test, AssignsAbsoluteSendTime) {
-  sender_.RegisterHeaderExtension(AbsoluteSendTime::kUri,
+  sender_.RegisterHeaderExtension(AbsoluteSendTime::Uri(),
                                   kAbsoluteSendTimeExtensionId);
 
   EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
@@ -982,7 +982,7 @@ TEST_P(RtpRtcpImpl2Test, AssignsAbsoluteSendTime) {
 }
 
 TEST_P(RtpRtcpImpl2Test, AssignsTransmissionTimeOffset) {
-  sender_.RegisterHeaderExtension(TransmissionOffset::kUri,
+  sender_.RegisterHeaderExtension(TransmissionOffset::Uri(),
                                   kTransmissionOffsetExtensionId);
 
   constexpr TimeDelta kOffset = TimeDelta::Millis(100);
@@ -998,7 +998,7 @@ TEST_P(RtpRtcpImpl2Test, AssignsTransmissionTimeOffset) {
 }
 
 TEST_P(RtpRtcpImpl2Test, PropagatesSentPacketInfo) {
-  sender_.RegisterHeaderExtension(TransportSequenceNumber::kUri,
+  sender_.RegisterHeaderExtension(TransportSequenceNumber::Uri(),
                                   kTransportSequenceNumberExtensionId);
   int64_t now_ms = time_controller_.GetClock()->TimeInMilliseconds();
   EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 0221800ea8ac..baa392281bc8 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -106,7 +106,7 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_audio_->SetAudioLevel(kAudioLevel));
-  rtp_module_->RegisterRtpHeaderExtension(AudioLevel::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
                                           kAudioLevelExtensionId);
 
   const char payload_name[] = "PAYLOAD_NAME";
@@ -149,7 +149,7 @@ TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
 }
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
-  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
   constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
   const char payload_name[] = "audio";
@@ -185,7 +185,7 @@ TEST_F(RtpSenderAudioTest,
   rtp_sender_audio_ =
       std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());
 
-  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
   constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
   const char payload_name[] = "audio";
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index d95a639e2155..0f2a218ae4c4 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -212,7 +212,7 @@ TEST_P(RtpSenderEgressTest, TransportFeedbackObserverGetsCorrectByteCount) {
   const uint16_t kTransportSequenceNumber = 17;
 
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   const size_t expected_bytes = GetParam().with_overhead
                                     ? kPayloadSize + kRtpOverheadBytesPerPacket
@@ -266,7 +266,7 @@ TEST_P(RtpSenderEgressTest,
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
 
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
   sender->SendPacket(packet.get(), PacedPacketInfo());
   EXPECT_TRUE(transport_.last_packet()->options.included_in_feedback);
@@ -278,7 +278,7 @@ TEST_P(
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
 
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
   sender->SendPacket(packet.get(), PacedPacketInfo());
   EXPECT_TRUE(transport_.last_packet()->options.included_in_allocation);
@@ -342,7 +342,7 @@ TEST_P(RtpSenderEgressTest, OnSendSideDelayUpdated) {
 TEST_P(RtpSenderEgressTest, WritesPacerExitToTimingExtension) {
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
   header_extensions_.RegisterByUri(kVideoTimingExtensionId,
-                                   VideoTimingExtension::kUri);
+                                   VideoTimingExtension::Uri());
 
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
   packet->SetExtension<VideoTimingExtension>(VideoSendTiming{});
@@ -364,7 +364,7 @@ TEST_P(RtpSenderEgressTest, WritesNetwork2ToTimingExtension) {
   rtp_config.populate_network2_timestamp = true;
   auto sender = std::make_unique<RtpSenderEgress>(rtp_config, &packet_history_);
   header_extensions_.RegisterByUri(kVideoTimingExtensionId,
-                                   VideoTimingExtension::kUri);
+                                   VideoTimingExtension::Uri());
 
   const uint16_t kPacerExitMs = 1234u;
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
@@ -388,7 +388,7 @@ TEST_P(RtpSenderEgressTest, WritesNetwork2ToTimingExtension) {
 TEST_P(RtpSenderEgressTest, OnSendPacketUpdated) {
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   const uint16_t kTransportSequenceNumber = 1;
   EXPECT_CALL(send_packet_observer_,
@@ -402,7 +402,7 @@ TEST_P(RtpSenderEgressTest, OnSendPacketUpdated) {
 TEST_P(RtpSenderEgressTest, OnSendPacketNotUpdatedForRetransmits) {
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   const uint16_t kTransportSequenceNumber = 1;
   EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(0);
@@ -748,11 +748,11 @@ TEST_P(RtpSenderEgressTest, UpdatesDataCounters) {
 
 TEST_P(RtpSenderEgressTest, SendPacketUpdatesExtensions) {
   header_extensions_.RegisterByUri(kVideoTimingExtensionId,
-                                   VideoTimingExtension::kUri);
+                                   VideoTimingExtension::Uri());
   header_extensions_.RegisterByUri(kAbsoluteSendTimeExtensionId,
-                                   AbsoluteSendTime::kUri);
+                                   AbsoluteSendTime::Uri());
   header_extensions_.RegisterByUri(kTransmissionOffsetExtensionId,
-                                   TransmissionOffset::kUri);
+                                   TransmissionOffset::Uri());
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
 
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
@@ -779,7 +779,7 @@ TEST_P(RtpSenderEgressTest, SendPacketSetsPacketOptions) {
   const uint16_t kPacketId = 42;
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
   packet->SetExtension<TransportSequenceNumber>(kPacketId);
@@ -816,7 +816,7 @@ TEST_P(RtpSenderEgressTest, SendPacketUpdatesStats) {
   auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
 
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   const int64_t capture_time_ms = clock_->TimeInMilliseconds();
 
@@ -871,7 +871,7 @@ TEST_P(RtpSenderEgressTest, SendPacketUpdatesStats) {
 TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRetransmission) {
   const uint16_t kTransportSequenceNumber = 17;
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
   std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
   retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
   retransmission->SetExtension<TransportSequenceNumber>(
@@ -893,7 +893,7 @@ TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRetransmission) {
 TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRtxRetransmission) {
   const uint16_t kTransportSequenceNumber = 17;
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   std::unique_ptr<RtpPacketToSend> rtx_retransmission = BuildRtpPacket();
   rtx_retransmission->SetSsrc(kRtxSsrc);
@@ -917,7 +917,7 @@ TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRtxRetransmission) {
 TEST_P(RtpSenderEgressTest, TransportFeedbackObserverPadding) {
   const uint16_t kTransportSequenceNumber = 17;
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
   std::unique_ptr<RtpPacketToSend> padding = BuildRtpPacket();
   padding->SetPadding(224);
   padding->set_packet_type(RtpPacketMediaType::kPadding);
@@ -935,7 +935,7 @@ TEST_P(RtpSenderEgressTest, TransportFeedbackObserverPadding) {
 TEST_P(RtpSenderEgressTest, TransportFeedbackObserverRtxPadding) {
   const uint16_t kTransportSequenceNumber = 17;
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   std::unique_ptr<RtpPacketToSend> rtx_padding = BuildRtpPacket();
   rtx_padding->SetPadding(224);
@@ -955,7 +955,7 @@ TEST_P(RtpSenderEgressTest, TransportFeedbackObserverRtxPadding) {
 TEST_P(RtpSenderEgressTest, TransportFeedbackObserverFec) {
   const uint16_t kTransportSequenceNumber = 17;
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
-                                   TransportSequenceNumber::kUri);
+                                   TransportSequenceNumber::Uri());
 
   std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
   fec_packet->SetSsrc(kFlexFecSsrc);
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 2724cec8ffb9..21c32b377c32 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -267,15 +267,16 @@ class RtpSenderTest : public ::testing::Test {
   // Enable sending of the MID header extension for both the primary SSRC and
   // the RTX SSRC.
   void EnableMidSending(const std::string& mid) {
-    rtp_sender_->RegisterRtpHeaderExtension(RtpMid::kUri, kMidExtensionId);
+    rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
     rtp_sender_->SetMid(mid);
   }
 
   // Enable sending of the RSID header extension for the primary SSRC and the
   // RRSID header extension for the RTX SSRC.
   void EnableRidSending(const std::string& rid) {
-    rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::kUri, kRidExtensionId);
-    rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::kUri,
+    rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
+                                            kRidExtensionId);
+    rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(),
                                             kRepairedRidExtensionId);
     rtp_sender_->SetRid(rid);
   }
@@ -297,15 +298,15 @@ TEST_F(RtpSenderTest, AllocatePacketSetCsrc) {
 TEST_F(RtpSenderTest, AllocatePacketReserveExtensions) {
   // Configure rtp_sender with extensions.
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
-  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(AudioLevel::kUri,
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
                                                       kAudioLevelExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      VideoOrientation::kUri, kVideoRotationExtensionId));
+      VideoOrientation::Uri(), kVideoRotationExtensionId));
 
   auto packet = rtp_sender_->AllocatePacket();
 
@@ -444,7 +445,7 @@ TEST_F(RtpSenderTest, NoPaddingAsFirstPacketWithoutBweExtensions) {
 
 TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
 
   // Padding can't be sent as first packet on media SSRC since we don't know
   // what payload type to assign.
@@ -465,7 +466,7 @@ TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
 
 TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithAbsSendTime) {
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
 
   // Padding can't be sent as first packet on media SSRC since we don't know
   // what payload type to assign.
@@ -518,7 +519,7 @@ TEST_F(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
 
 TEST_F(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
   EnableRtx();
   // Timestamps as set based on capture time in RtpSenderTest.
   const int64_t start_time = clock_->TimeInMilliseconds();
@@ -892,7 +893,7 @@ TEST_F(RtpSenderTest, OnOverheadChanged) {
   // Base RTP overhead is 12B.
   EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
 
-  rtp_sender_->RegisterRtpHeaderExtension(TransmissionOffset::kUri,
+  rtp_sender_->RegisterRtpHeaderExtension(TransmissionOffset::Uri(),
                                           kTransmissionTimeOffsetExtensionId);
 
   // TransmissionTimeOffset extension has a size of 3B, but with the addition
@@ -908,8 +909,8 @@ TEST_F(RtpSenderTest, CountMidOnlyUntilAcked) {
   // Base RTP overhead is 12B.
   EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
 
-  rtp_sender_->RegisterRtpHeaderExtension(RtpMid::kUri, kMidExtensionId);
-  rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::kUri, kRidExtensionId);
+  rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
+  rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), kRidExtensionId);
 
   // Counted only if set.
   EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
@@ -931,15 +932,16 @@ TEST_F(RtpSenderTest, DontCountVolatileExtensionsIntoOverhead) {
   // Base RTP overhead is 12B.
   EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
 
-  rtp_sender_->RegisterRtpHeaderExtension(InbandComfortNoiseExtension::kUri, 1);
-  rtp_sender_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
+  rtp_sender_->RegisterRtpHeaderExtension(InbandComfortNoiseExtension::Uri(),
+                                          1);
+  rtp_sender_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           2);
-  rtp_sender_->RegisterRtpHeaderExtension(VideoOrientation::kUri, 3);
-  rtp_sender_->RegisterRtpHeaderExtension(PlayoutDelayLimits::kUri, 4);
-  rtp_sender_->RegisterRtpHeaderExtension(VideoContentTypeExtension::kUri, 5);
-  rtp_sender_->RegisterRtpHeaderExtension(VideoTimingExtension::kUri, 6);
-  rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::kUri, 7);
-  rtp_sender_->RegisterRtpHeaderExtension(ColorSpaceExtension::kUri, 8);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoOrientation::Uri(), 3);
+  rtp_sender_->RegisterRtpHeaderExtension(PlayoutDelayLimits::Uri(), 4);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoContentTypeExtension::Uri(), 5);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoTimingExtension::Uri(), 6);
+  rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(), 7);
+  rtp_sender_->RegisterRtpHeaderExtension(ColorSpaceExtension::Uri(), 8);
 
   // Still only 12B counted since can't count on above being sent.
   EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
@@ -1006,11 +1008,11 @@ TEST_F(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
   EnableRtx();
 
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
 
   // Put a packet in the history, in order to facilitate payload padding.
   std::unique_ptr<RtpPacketToSend> packet =
@@ -1053,7 +1055,7 @@ TEST_F(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
       RtpPacketHistory::StorageMode::kStoreAndCull, 1);
 
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
 
   const size_t kPayloadPacketSize = kMinPaddingSize;
   std::unique_ptr<RtpPacketToSend> packet =
@@ -1103,7 +1105,7 @@ TEST_F(RtpSenderTest, LimitsPayloadPaddingSize) {
       RtpPacketHistory::StorageMode::kStoreAndCull, 1);
 
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
 
   // Send a dummy video packet so it ends up in the packet history.
   const size_t kPayloadPacketSize = 1234u;
@@ -1139,11 +1141,11 @@ TEST_F(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
   packet_history_->SetStorePacketsStatus(
       RtpPacketHistory::StorageMode::kStoreAndCull, 1);
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
   ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
-      TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
 
   const size_t kPayloadPacketSize = 1234;
   // Send a dummy video packet so it ends up in the packet history. Since we
@@ -1188,8 +1190,8 @@ TEST_F(RtpSenderTest, SupportsPadding) {
   bool kSendingMediaStats[] = {true, false};
   bool kEnableRedundantPayloads[] = {true, false};
   absl::string_view kBweExtensionUris[] = {
-      TransportSequenceNumber::kUri, TransportSequenceNumberV2::kUri,
-      AbsoluteSendTime::kUri, TransmissionOffset::kUri};
+      TransportSequenceNumber::Uri(), TransportSequenceNumberV2::Uri(),
+      AbsoluteSendTime::Uri(), TransmissionOffset::Uri()};
   const int kExtensionsId = 7;
 
   for (bool sending_media : kSendingMediaStats) {
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 3e803fff1f50..ffdd0d8f6c46 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -207,7 +207,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
 
 TEST_P(RtpSenderVideoTest, KeyFrameHasCVO) {
   uint8_t kFrame[kMaxPacketLength];
-  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
                                           kVideoRotationExtensionId);
 
   RTPVideoHeader hdr;
@@ -227,7 +227,7 @@ TEST_P(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
   const int64_t kPacketizationTimeMs = 100;
   const int64_t kEncodeStartDeltaMs = 10;
   const int64_t kEncodeFinishDeltaMs = 50;
-  rtp_module_->RegisterRtpHeaderExtension(VideoTimingExtension::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(VideoTimingExtension::Uri(),
                                           kVideoTimingExtensionId);
 
   const int64_t kCaptureTimestamp = fake_clock_.TimeInMilliseconds();
@@ -252,7 +252,7 @@ TEST_P(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
 
 TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
   uint8_t kFrame[kMaxPacketLength];
-  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
                                           kVideoRotationExtensionId);
 
   RTPVideoHeader hdr;
@@ -276,7 +276,7 @@ TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
 
 TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero) {
   uint8_t kFrame[kMaxPacketLength];
-  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
                                           kVideoRotationExtensionId);
 
   RTPVideoHeader hdr;
@@ -508,7 +508,7 @@ TEST_P(RtpSenderVideoTest, SendsDependencyDescriptorWhenVideoStructureIsSet) {
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   FrameDependencyStructure video_structure;
   video_structure.num_decode_targets = 2;
   video_structure.templates = {
@@ -577,7 +577,7 @@ TEST_P(RtpSenderVideoTest,
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   rtp_module_->SetExtmapAllowMixed(false);
   FrameDependencyStructure video_structure;
   video_structure.num_decode_targets = 2;
@@ -630,7 +630,7 @@ TEST_P(RtpSenderVideoTest, PropagatesChainDiffsIntoDependencyDescriptor) {
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   FrameDependencyStructure video_structure;
   video_structure.num_decode_targets = 2;
   video_structure.num_chains = 1;
@@ -664,7 +664,7 @@ TEST_P(RtpSenderVideoTest,
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   FrameDependencyStructure video_structure;
   video_structure.num_decode_targets = 2;
   video_structure.num_chains = 1;
@@ -698,7 +698,7 @@ TEST_P(RtpSenderVideoTest,
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   FrameDependencyStructure video_structure1;
   video_structure1.num_decode_targets = 2;
   video_structure1.templates = {
@@ -773,7 +773,7 @@ TEST_P(RtpSenderVideoTest,
   uint8_t kFrame[kFrameSize] = {1, 2, 3, 4};
 
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpDependencyDescriptorExtension::kUri, kDependencyDescriptorId);
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
   rtc::scoped_refptr<MockFrameEncryptor> encryptor(
       new rtc::RefCountedObject<NiceMock<MockFrameEncryptor>>);
   ON_CALL(*encryptor, GetMaxCiphertextByteSize).WillByDefault(ReturnArg<1>());
@@ -815,7 +815,7 @@ TEST_P(RtpSenderVideoTest, PopulateGenericFrameDescriptor) {
   const int64_t kFrameId = 100000;
   uint8_t kFrame[100];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpGenericFrameDescriptorExtension00::kUri, kGenericDescriptorId);
+      RtpGenericFrameDescriptorExtension00::Uri(), kGenericDescriptorId);
 
   RTPVideoHeader hdr;
   RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
@@ -847,7 +847,7 @@ void RtpSenderVideoTest::
   uint8_t kFrame[kFrameSize];
 
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpGenericFrameDescriptorExtension00::kUri, kGenericDescriptorId);
+      RtpGenericFrameDescriptorExtension00::Uri(), kGenericDescriptorId);
 
   RTPVideoHeader hdr;
   hdr.codec = kVideoCodecVP8;
@@ -882,7 +882,7 @@ TEST_P(RtpSenderVideoTest, VideoLayersAllocationWithResolutionSentOnKeyFrames) {
   const size_t kFrameSize = 100;
   uint8_t kFrame[kFrameSize];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpVideoLayersAllocationExtension::kUri,
+      RtpVideoLayersAllocationExtension::Uri(),
       kVideoLayersAllocationExtensionId);
 
   VideoLayersAllocation allocation;
@@ -919,7 +919,7 @@ TEST_P(RtpSenderVideoTest,
   const size_t kFrameSize = 100;
   uint8_t kFrame[kFrameSize];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpVideoLayersAllocationExtension::kUri,
+      RtpVideoLayersAllocationExtension::Uri(),
       kVideoLayersAllocationExtensionId);
 
   VideoLayersAllocation allocation;
@@ -967,7 +967,7 @@ TEST_P(RtpSenderVideoTest,
   const size_t kFrameSize = 100;
   uint8_t kFrame[kFrameSize];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpVideoLayersAllocationExtension::kUri,
+      RtpVideoLayersAllocationExtension::Uri(),
       kVideoLayersAllocationExtensionId);
 
   VideoLayersAllocation allocation;
@@ -1012,7 +1012,7 @@ TEST_P(RtpSenderVideoTest, VideoLayersAllocationSentOnDeltaFramesOnlyOnUpdate) {
   const size_t kFrameSize = 100;
   uint8_t kFrame[kFrameSize];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpVideoLayersAllocationExtension::kUri,
+      RtpVideoLayersAllocationExtension::Uri(),
       kVideoLayersAllocationExtensionId);
 
   VideoLayersAllocation allocation;
@@ -1054,7 +1054,7 @@ TEST_P(RtpSenderVideoTest, VideoLayersAllocationNotSentOnHigherTemporalLayers) {
   const size_t kFrameSize = 100;
   uint8_t kFrame[kFrameSize];
   rtp_module_->RegisterRtpHeaderExtension(
-      RtpVideoLayersAllocationExtension::kUri,
+      RtpVideoLayersAllocationExtension::Uri(),
       kVideoLayersAllocationExtensionId);
 
   VideoLayersAllocation allocation;
@@ -1089,7 +1089,7 @@ TEST_P(RtpSenderVideoTest, VideoLayersAllocationNotSentOnHigherTemporalLayers) {
 TEST_P(RtpSenderVideoTest, AbsoluteCaptureTime) {
   constexpr int64_t kAbsoluteCaptureTimestampMs = 12345678;
   uint8_t kFrame[kMaxPacketLength];
-  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
 
   RTPVideoHeader hdr;
@@ -1127,7 +1127,7 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTimeWithCaptureClockOffset) {
 
   constexpr int64_t kAbsoluteCaptureTimestampMs = 12345678;
   uint8_t kFrame[kMaxPacketLength];
-  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
 
   RTPVideoHeader hdr;
@@ -1162,7 +1162,7 @@ TEST_P(RtpSenderVideoTest, PopulatesPlayoutDelay) {
   // Single packet frames.
   constexpr size_t kPacketSize = 123;
   uint8_t kFrame[kPacketSize];
-  rtp_module_->RegisterRtpHeaderExtension(PlayoutDelayLimits::kUri,
+  rtp_module_->RegisterRtpHeaderExtension(PlayoutDelayLimits::Uri(),
                                           kPlayoutDelayExtensionId);
   const VideoPlayoutDelay kExpectedDelay = {10, 20};
 
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
index ff8ea2a7a2bf..a6986349da68 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
@@ -11,6 +11,9 @@
 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_LAYERS_ALLOCATION_EXTENSION_H_
 #define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_LAYERS_ALLOCATION_EXTENSION_H_
 
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
+#include "api/rtp_parameters.h"
 #include "api/video/video_layers_allocation.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
@@ -22,8 +25,12 @@ class RtpVideoLayersAllocationExtension {
  public:
   using value_type = VideoLayersAllocation;
   static constexpr RTPExtensionType kId = kRtpExtensionVideoLayersAllocation;
-  static constexpr const char kUri[] =
+  static constexpr ABSL_DEPRECATED("Use Uri()") const char kUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/video-layers-allocation00";
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoLayersAllocationUri;
+  }
+
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     VideoLayersAllocation* allocation);
   static size_t ValueSize(const VideoLayersAllocation& allocation);