diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 5c88e01c13..24f6fe1abb 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -434,6 +434,8 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   stats.apm_statistics =
       audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
 
+  stats.report_block_datas = std::move(call_stats.report_block_datas);
+
   return stats;
 }
 
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 38e89d8e3d..793a7dd263 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -1079,6 +1079,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
   stats.packetsSent =
       rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
   stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
+  stats.report_block_datas = _rtpRtcpModule->GetLatestReportBlockData();
 
   return stats;
 }
diff --git a/audio/channel_send.h b/audio/channel_send.h
index fb98be3c55..2762f5360b 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -22,6 +22,7 @@
 #include "api/media_transport_config.h"
 #include "api/media_transport_interface.h"
 #include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtp_sender_audio.h"
 
@@ -41,6 +42,11 @@ struct CallSendStatistics {
   int packetsSent;
   // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
   uint64_t retransmitted_packets_sent;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  std::vector<ReportBlockData> report_block_datas;
 };
 
 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 64cba5079b..1696478ded 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -44,6 +44,7 @@ rtc_source_set("call_interfaces") {
     "../modules/audio_processing",
     "../modules/audio_processing:api",
     "../modules/audio_processing:audio_processing_statistics",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
     "../modules/utility",
     "../rtc_base",
     "../rtc_base:audio_format_to_string",
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index b21b2ef342..d8fdddb764 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -29,6 +29,7 @@
 #include "api/scoped_refptr.h"
 #include "call/rtp_config.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
 
 namespace webrtc {
 
@@ -66,6 +67,11 @@ class AudioSendStream {
     AudioProcessingStats apm_statistics;
 
     int64_t target_bitrate_bps = 0;
+    // A snapshot of Report Blocks with additional data of interest to
+    // statistics. Within this list, the sender-source SSRC pair is unique and
+    // per-pair the ReportBlockData represents the latest Report Block that was
+    // received for that pair.
+    std::vector<ReportBlockData> report_block_datas;
   };
 
   struct Config {
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 435b3abad0..55a2826ca9 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2215,6 +2215,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
     sinfo.ana_statistics = stats.ana_statistics;
     sinfo.apm_statistics = stats.apm_statistics;
+    sinfo.report_block_datas = std::move(stats.report_block_datas);
     info->senders.push_back(sinfo);
   }
 
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 216f758d63..22de74ef57 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -350,8 +350,15 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
 
   // Returns received RTCP report block.
   // Returns -1 on failure else 0.
+  // TODO(https://crbug.com/webrtc/10678): Remove this in favor of
+  // GetLatestReportBlockData().
   virtual int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const = 0;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
 
   // (APP) Sets application specific data.
   // Returns -1 on failure else 0.
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index eaad03d9a6..b1ec2ca728 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -126,6 +126,7 @@ class MockRtpRtcp : public RtpRtcp {
                      void(bool, uint32_t, struct RtpPacketLossStats*));
   MOCK_CONST_METHOD1(RemoteRTCPStat,
                      int32_t(std::vector<RTCPReportBlock>* receive_blocks));
+  MOCK_CONST_METHOD0(GetLatestReportBlockData, std::vector<ReportBlockData>());
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
                int32_t(uint8_t sub_type,
                        uint32_t name,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 1d7f048e4b..4855775beb 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -589,6 +589,11 @@ int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
   return rtcp_receiver_.StatisticsReceived(receive_blocks);
 }
 
+std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
+    const {
+  return rtcp_receiver_.GetLatestReportBlockData();
+}
+
 // (REMB) Receiver Estimated Max Bitrate.
 void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps,
                                 std::vector<uint32_t> ssrcs) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 71a856cfe5..286a8af5fd 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -200,6 +200,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
   // Get received RTCP report, report block.
   int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const override;
+  // A snapshot of the most recent Report Block with additional data of
+  // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
+  // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
+  // which is the SSRC of the corresponding outbound RTP stream, is unique.
+  std::vector<ReportBlockData> GetLatestReportBlockData() const override;
 
   // (REMB) Receiver Estimated Max Bitrate.
   void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;