diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 891e518..7ef00a9 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -496,6 +496,19 @@
   inbound_audio->fec_packets_discarded =
       voice_receiver_info.fec_packets_discarded;
   inbound_audio->packets_discarded = voice_receiver_info.packets_discarded;
+  inbound_audio->jitter_buffer_flushes =
+      voice_receiver_info.jitter_buffer_flushes;
+  inbound_audio->delayed_packet_outage_samples =
+      voice_receiver_info.delayed_packet_outage_samples;
+  inbound_audio->relative_packet_arrival_delay =
+      voice_receiver_info.relative_packet_arrival_delay_seconds;
+  inbound_audio->interruption_count =
+      voice_receiver_info.interruption_count >= 0
+          ? voice_receiver_info.interruption_count
+          : 0;
+  inbound_audio->total_interruption_duration =
+      static_cast<double>(voice_receiver_info.total_interruption_duration_ms) /
+      rtc::kNumMillisecsPerSec;
   return inbound_audio;
 }
 
@@ -1034,6 +1047,9 @@
       voice_receiver_info.silent_concealed_samples;
   audio_track_stats->concealment_events =
       voice_receiver_info.concealment_events;
+
+  // TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
+  // stats, delete them.
   audio_track_stats->jitter_buffer_flushes =
       voice_receiver_info.jitter_buffer_flushes;
   audio_track_stats->delayed_packet_outage_samples =
@@ -1047,6 +1063,7 @@
   audio_track_stats->total_interruption_duration =
       static_cast<double>(voice_receiver_info.total_interruption_duration_ms) /
       rtc::kNumMillisecsPerSec;
+
   return audio_track_stats;
 }
 
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 3c521c82..31083db 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2234,6 +2234,8 @@
   voice_receiver_info.silent_concealed_samples = 765;
   voice_receiver_info.jitter_buffer_delay_seconds = 3.456;
   voice_receiver_info.jitter_buffer_emitted_count = 13;
+  // TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
+  // stats, no need to test these here.
   voice_receiver_info.jitter_buffer_flushes = 7;
   voice_receiver_info.delayed_packet_outage_samples = 15;
   voice_receiver_info.relative_packet_arrival_delay_seconds = 16;
@@ -2278,6 +2280,8 @@
   expected_remote_audio_track.silent_concealed_samples = 765;
   expected_remote_audio_track.jitter_buffer_delay = 3.456;
   expected_remote_audio_track.jitter_buffer_emitted_count = 13;
+  // TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
+  // stats, delete them.
   expected_remote_audio_track.jitter_buffer_flushes = 7;
   expected_remote_audio_track.delayed_packet_outage_samples = 15;
   expected_remote_audio_track.relative_packet_arrival_delay = 16;
@@ -2471,6 +2475,11 @@
   voice_media_info.receivers[0].audio_level = 14442;  // [0,32767]
   voice_media_info.receivers[0].total_output_energy = 10.0;
   voice_media_info.receivers[0].total_output_duration = 11.0;
+  voice_media_info.receivers[0].jitter_buffer_flushes = 7;
+  voice_media_info.receivers[0].delayed_packet_outage_samples = 15;
+  voice_media_info.receivers[0].relative_packet_arrival_delay_seconds = 16;
+  voice_media_info.receivers[0].interruption_count = 7788;
+  voice_media_info.receivers[0].total_interruption_duration_ms = 778899;
 
   voice_media_info.receivers[0].last_packet_received_timestamp_ms =
       absl::nullopt;
@@ -2526,6 +2535,11 @@
   expected_audio.audio_level = 14442.0 / 32767.0;  // [0,1]
   expected_audio.total_audio_energy = 10.0;
   expected_audio.total_samples_duration = 11.0;
+  expected_audio.jitter_buffer_flushes = 7;
+  expected_audio.delayed_packet_outage_samples = 15;
+  expected_audio.relative_packet_arrival_delay = 16;
+  expected_audio.interruption_count = 7788;
+  expected_audio.total_interruption_duration = 778.899;
 
   ASSERT_TRUE(report->Get(expected_audio.id()));
   EXPECT_EQ(
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 4bb8473..0cecdea 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -656,6 +656,8 @@
           media_stream_track.inserted_samples_for_deceleration);
       verifier.TestMemberIsUndefined(
           media_stream_track.removed_samples_for_acceleration);
+      // TODO(crbug.com/webrtc/14524): These metrics have been moved from
+      // "track" stats, delete them.
       verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
       verifier.TestMemberIsUndefined(
           media_stream_track.delayed_packet_outage_samples);
@@ -691,6 +693,8 @@
             media_stream_track.inserted_samples_for_deceleration);
         verifier.TestMemberIsNonNegative<uint64_t>(
             media_stream_track.removed_samples_for_acceleration);
+        // TODO(crbug.com/webrtc/14524): These metrics have been moved from
+        // "track" stats, delete them.
         verifier.TestMemberIsNonNegative<uint64_t>(
             media_stream_track.jitter_buffer_flushes);
         verifier.TestMemberIsNonNegative<uint64_t>(
@@ -722,6 +726,8 @@
             media_stream_track.inserted_samples_for_deceleration);
         verifier.TestMemberIsUndefined(
             media_stream_track.removed_samples_for_acceleration);
+        // TODO(crbug.com/webrtc/14524): These metrics have been moved from
+        // "track" stats, delete them.
         verifier.TestMemberIsUndefined(
             media_stream_track.jitter_buffer_flushes);
         verifier.TestMemberIsUndefined(
@@ -920,6 +926,14 @@
       // The integration test is not set up to test screen share; don't require
       // this to be present.
       verifier.MarkMemberTested(inbound_stream.content_type, true);
+      verifier.TestMemberIsUndefined(inbound_stream.jitter_buffer_flushes);
+      verifier.TestMemberIsUndefined(
+          inbound_stream.delayed_packet_outage_samples);
+      verifier.TestMemberIsUndefined(
+          inbound_stream.relative_packet_arrival_delay);
+      verifier.TestMemberIsUndefined(inbound_stream.interruption_count);
+      verifier.TestMemberIsUndefined(
+          inbound_stream.total_interruption_duration);
       verifier.TestMemberIsNonNegative<double>(
           inbound_stream.min_playout_delay);
     } else {
@@ -939,6 +953,16 @@
       verifier.TestMemberIsUndefined(inbound_stream.freeze_count);
       verifier.TestMemberIsUndefined(inbound_stream.total_freezes_duration);
       verifier.TestMemberIsUndefined(inbound_stream.content_type);
+      verifier.TestMemberIsNonNegative<uint64_t>(
+          inbound_stream.jitter_buffer_flushes);
+      verifier.TestMemberIsNonNegative<uint64_t>(
+          inbound_stream.delayed_packet_outage_samples);
+      verifier.TestMemberIsNonNegative<double>(
+          inbound_stream.relative_packet_arrival_delay);
+      verifier.TestMemberIsNonNegative<uint32_t>(
+          inbound_stream.interruption_count);
+      verifier.TestMemberIsNonNegative<double>(
+          inbound_stream.total_interruption_duration);
       verifier.TestMemberIsUndefined(inbound_stream.min_playout_delay);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 7e5cc74..1d9882e 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -171,6 +171,8 @@
 // endpoint that only signals a=msid lines to convey stream_ids.
 void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
 
+// TODO(https://crbug.com/webrtc/14175): Stop depending on "track" stats, the
+// metrics we're interested in are already available in "inbound-rtp".
 int FindFirstMediaStatsIndexByKind(
     const std::string& kind,
     const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
@@ -650,34 +652,26 @@
   void StartWatchingDelayStats() {
     // Get the baseline numbers for audio_packets and audio_delay.
     auto received_stats = NewGetStats();
-    auto track_stats =
-        received_stats->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>()[0];
-    ASSERT_TRUE(track_stats->relative_packet_arrival_delay.is_defined());
     auto rtp_stats =
         received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0];
+    ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
     ASSERT_TRUE(rtp_stats->packets_received.is_defined());
     ASSERT_TRUE(rtp_stats->track_id.is_defined());
-    audio_track_stats_id_ = track_stats->id();
-    ASSERT_TRUE(received_stats->Get(audio_track_stats_id_));
     rtp_stats_id_ = rtp_stats->id();
-    ASSERT_EQ(audio_track_stats_id_, *rtp_stats->track_id);
     audio_packets_stat_ = *rtp_stats->packets_received;
-    audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
-    audio_samples_stat_ = *track_stats->total_samples_received;
-    audio_concealed_stat_ = *track_stats->concealed_samples;
+    audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
+    audio_samples_stat_ = *rtp_stats->total_samples_received;
+    audio_concealed_stat_ = *rtp_stats->concealed_samples;
   }
 
   void UpdateDelayStats(std::string tag, int desc_size) {
     auto report = NewGetStats();
-    auto track_stats =
-        report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id_);
-    ASSERT_TRUE(track_stats);
     auto rtp_stats =
         report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_);
     ASSERT_TRUE(rtp_stats);
     auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
     auto delta_rpad =
-        *track_stats->relative_packet_arrival_delay - audio_delay_stat_;
+        *rtp_stats->relative_packet_arrival_delay - audio_delay_stat_;
     auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
     // The purpose of these checks is to sound the alarm early if we introduce
     // serious regressions. The numbers are not acceptable for production, but
@@ -694,9 +688,9 @@
     EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size;
 #endif
     auto delta_samples =
-        *track_stats->total_samples_received - audio_samples_stat_;
+        *rtp_stats->total_samples_received - audio_samples_stat_;
     auto delta_concealed =
-        *track_stats->concealed_samples - audio_concealed_stat_;
+        *rtp_stats->concealed_samples - audio_concealed_stat_;
     // These limits should be adjusted down as we improve:
     //
     // Concealing more than 4000 samples during a renegotiation is unacceptable.
@@ -731,9 +725,9 @@
     }
     // Increment trailing counters
     audio_packets_stat_ = *rtp_stats->packets_received;
-    audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
-    audio_samples_stat_ = *track_stats->total_samples_received;
-    audio_concealed_stat_ = *track_stats->concealed_samples;
+    audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
+    audio_samples_stat_ = *rtp_stats->total_samples_received;
+    audio_concealed_stat_ = *rtp_stats->concealed_samples;
   }
 
   // Sets number of candidates expected
@@ -1240,7 +1234,6 @@
   uint64_t audio_samples_stat_ = 0;
   uint64_t audio_concealed_stat_ = 0;
   std::string rtp_stats_id_;
-  std::string audio_track_stats_id_;
 
   ScopedTaskSafety task_safety_;