diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index d140eb04fe..f6bcddb5c8 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -240,7 +240,10 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics(
   // Since cumulative loss is carried in a signed 24-bit field in RTCP, we may
   // need to clamp it.
   statistics.packets_lost = std::min(statistics.packets_lost, 0x7fffff);
-  statistics.packets_lost = std::max(statistics.packets_lost, -0x800000);
+  // TODO(bugs.webrtc.org/9598): This packets_lost should be signed according to
+  // RFC3550. However, old WebRTC implementations reads it as unsigned.
+  // Therefore we limit this to 0.
+  statistics.packets_lost = std::max(statistics.packets_lost, 0);
   statistics.extended_highest_sequence_number = extended_seq_max;
   // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
   statistics.jitter = jitter_q4_ >> 4;
diff --git a/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 3c7fb6f83c..eedcf46df7 100644
--- a/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -479,7 +479,9 @@ TEST_F(ReceiveStatisticsTest, NegativeLoss) {
   receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
       &statistics, /*update_fraction_lost=*/true);
   EXPECT_EQ(0u, statistics.fraction_lost);
-  EXPECT_EQ(-1, statistics.packets_lost);
+  // TODO(bugs.webrtc.org/9598): Since old WebRTC implementations reads this
+  // value as unsigned we currently limit it to 0.
+  EXPECT_EQ(0, statistics.packets_lost);
 
   // Lose 2 packets; now cumulative loss should become positive again.
   header1_.sequenceNumber = 7;
@@ -521,7 +523,9 @@ TEST_F(ReceiveStatisticsTest, NegativeCumulativeLossClamped) {
   RtcpStatistics statistics;
   receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
       &statistics, /*update_fraction_lost=*/false);
-  EXPECT_EQ(-0x800000, statistics.packets_lost);
+  // TODO(bugs.webrtc.org/9598): Since old WebRTC implementations reads this
+  // value as unsigned we currently limit it to 0.
+  EXPECT_EQ(0, statistics.packets_lost);
 }
 
 // Test that the extended highest sequence number is computed correctly when