diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 4c399a107e..26c8cdd8c7 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -24,9 +24,14 @@
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
+namespace {
+constexpr int64_t kStatisticsTimeoutMs = 8000;
+constexpr int64_t kStatisticsProcessIntervalMs = 1000;
 
-const int64_t kStatisticsTimeoutMs = 8000;
-const int64_t kStatisticsProcessIntervalMs = 1000;
+// Number of seconds since 1900 January 1 00:00 GMT (see
+// https://tools.ietf.org/html/rfc868).
+constexpr int64_t kNtpJan1970Millisecs = 2'208'988'800'000;
+}  // namespace
 
 StreamStatistician::~StreamStatistician() {}
 
@@ -35,6 +40,9 @@ StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc,
                                                int max_reordering_threshold)
     : ssrc_(ssrc),
       clock_(clock),
+      delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() -
+                                    clock_->TimeInMilliseconds() -
+                                    kNtpJan1970Millisecs),
       incoming_bitrate_(kStatisticsProcessIntervalMs,
                         RateStatistics::kBpsScale),
       max_reordering_threshold_(max_reordering_threshold),
@@ -172,8 +180,11 @@ RtpReceiveStats StreamStatisticianImpl::GetStats() const {
   // TODO(nisse): Can we return a float instead?
   // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
   stats.jitter = jitter_q4_ >> 4;
-  stats.last_packet_received_timestamp_ms =
-      receive_counters_.last_packet_received_timestamp_ms;
+  if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
+    stats.last_packet_received_timestamp_ms =
+        *receive_counters_.last_packet_received_timestamp_ms +
+        delta_internal_unix_epoch_ms_;
+  }
   stats.packet_counter = receive_counters_.transmitted;
   return stats;
 }
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.h b/modules/rtp_rtcp/source/receive_statistics_impl.h
index 2456f93e9a..be56f4ba5a 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -73,6 +73,8 @@ class StreamStatisticianImpl : public StreamStatisticianImplInterface {
 
   const uint32_t ssrc_;
   Clock* const clock_;
+  // Delta used to map internal timestamps to Unix epoch ones.
+  const int64_t delta_internal_unix_epoch_ms_;
   RateStatistics incoming_bitrate_;
   // In number of packets or sequence numbers.
   int max_reordering_threshold_;
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 36ee5425b1..4c9dfa489e 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -372,6 +372,7 @@ std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
         *voice_receiver_info.last_packet_received_timestamp_ms);
   }
   if (voice_receiver_info.estimated_playout_ntp_timestamp_ms) {
+    // TODO(bugs.webrtc.org/10529): Fix time origin.
     inbound_audio->estimated_playout_timestamp = static_cast<double>(
         *voice_receiver_info.estimated_playout_ntp_timestamp_ms);
   }
@@ -471,17 +472,16 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
   inbound_video->total_squared_inter_frame_delay =
       video_receiver_info.total_squared_inter_frame_delay;
   if (video_receiver_info.last_packet_received_timestamp_ms) {
-    inbound_video->last_packet_received_timestamp =
-        static_cast<double>(
-            *video_receiver_info.last_packet_received_timestamp_ms) /
-        rtc::kNumMillisecsPerSec;
+    inbound_video->last_packet_received_timestamp = static_cast<double>(
+        *video_receiver_info.last_packet_received_timestamp_ms);
   }
   if (video_receiver_info.estimated_playout_ntp_timestamp_ms) {
+    // TODO(bugs.webrtc.org/10529): Fix time origin if needed.
     inbound_video->estimated_playout_timestamp = static_cast<double>(
         *video_receiver_info.estimated_playout_ntp_timestamp_ms);
   }
-  // TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
-  // optional, support the "unspecified" value.
+  // TODO(bugs.webrtc.org/10529): When info's |content_info| is optional
+  // support the "unspecified" value.
   if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)
     inbound_video->content_type = RTCContentType::kScreenshare;
   if (!video_receiver_info.decoder_implementation_name.empty()) {
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 897226d25e..655f7e6315 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2107,7 +2107,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   video_media_info.receivers[0].qp_sum = 9;
   expected_video.qp_sum = 9;
   video_media_info.receivers[0].last_packet_received_timestamp_ms = 1000;
-  expected_video.last_packet_received_timestamp = 1.0;
+  expected_video.last_packet_received_timestamp = 1000.0;
   video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
   expected_video.content_type = "screenshare";
   video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 1234;