diff --git a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
index 07f3d63b8b..b448dc218e 100644
--- a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
@@ -45,10 +45,24 @@ void DefaultAudioQualityAnalyzer::OnStatsReports(
     if (strcmp(media_type->static_string_val(), kStatsAudioMediaType) != 0) {
       continue;
     }
+
     if (stats_report->FindValue(
             webrtc::StatsReport::kStatsValueNameBytesSent)) {
-      // If kStatsValueNameBytesSent is present, it means it's a send stream,
-      // but we need audio metrics for receive stream, so skip it.
+      // If kStatsValueNameBytesSent is present, it means it's a send stream.
+      // All we need from a send stream is bytes sent.
+      const webrtc::StatsReport::Value* bytes_sent = stats_report->FindValue(
+          StatsReport::StatsValueName::kStatsValueNameBytesSent);
+      const webrtc::StatsReport::Value* report_track_id =
+          stats_report->FindValue(
+              StatsReport::StatsValueName::kStatsValueNameTrackId);
+
+      rtc::CritScope crit(&lock_);
+      // Note: outgoing streams have their "stream label" directly in the
+      // report's track id field.  There is no need to look it up using
+      // GetStreamLabelFromStatsReport(), and in fact doing so will crash.
+      AudioStreamStats& audio_stream_stats =
+          streams_stats_[report_track_id->string_val()];
+      audio_stream_stats.bytes_sent = bytes_sent->int64_val();
       continue;
     }
 
@@ -112,6 +126,9 @@ void DefaultAudioQualityAnalyzer::Stop() {
                  item.second.speech_expand_rate, "unitless");
     ReportResult("preferred_buffer_size_ms", item.first,
                  item.second.preferred_buffer_size_ms, "ms");
+    test::PrintResult("bytes_sent", "", GetTestCaseName(item.first),
+                      item.second.bytes_sent, "sizeInBytes",
+                      /*important=*/false);
   }
 }
 
diff --git a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
index ee34ed345b..824da607d9 100644
--- a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
@@ -29,6 +29,7 @@ struct AudioStreamStats {
   SamplesStatsCounter preemptive_rate;
   SamplesStatsCounter speech_expand_rate;
   SamplesStatsCounter preferred_buffer_size_ms;
+  int64_t bytes_sent;
 };
 
 // TODO(bugs.webrtc.org/10430): Migrate to the new GetStats as soon as
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index 499d04e2d5..45516d2ea4 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -27,6 +27,7 @@ constexpr int kMaxActiveComparisons = 10;
 constexpr int kFreezeThresholdMs = 150;
 constexpr int kMicrosPerSecond = 1000000;
 constexpr int kBitsInByte = 8;
+constexpr char kStatsVideoMediaType[] = "video";
 
 void LogFrameCounters(const std::string& name, const FrameCounters& counters) {
   RTC_LOG(INFO) << "[" << name << "] Captured    : " << counters.captured;
@@ -373,7 +374,28 @@ void DefaultVideoQualityAnalyzer::OnStatsReports(
     const std::string& pc_label,
     const StatsReports& stats_reports) {
   for (const StatsReport* stats_report : stats_reports) {
-    // The only stats collected by this analyzer are present in
+    // Record the number of video bytes sent from outgoing SSRC reports.
+    if (stats_report->type() == StatsReport::StatsType::kStatsReportTypeSsrc &&
+        strcmp(stats_report
+                   ->FindValue(
+                       StatsReport::StatsValueName::kStatsValueNameMediaType)
+                   ->static_string_val(),
+               kStatsVideoMediaType) == 0 &&
+        stats_report->FindValue(StatsReport::kStatsValueNameBytesSent)) {
+      const webrtc::StatsReport::Value* bytes_sent = stats_report->FindValue(
+          StatsReport::StatsValueName::kStatsValueNameBytesSent);
+      const webrtc::StatsReport::Value* track_id = stats_report->FindValue(
+          StatsReport::StatsValueName::kStatsValueNameTrackId);
+
+      rtc::CritScope crit(&comparison_lock_);
+      // Note: outgoing streams have their "stream label" directly in the
+      // report's track id field.  There is no need to look it up using
+      // GetStreamLabelFromStatsReport(), and in fact doing so will crash.
+      StreamStats& stream_stats = stream_stats_[track_id->string_val()];
+      stream_stats.bytes_sent = bytes_sent->int64_val();
+    }
+
+    // The only other stats collected by this analyzer are present in
     // kStatsReportTypeBwe reports, so all other reports are just ignored.
     if (stats_report->type() != StatsReport::StatsType::kStatsReportTypeBwe) {
       continue;
@@ -629,6 +651,8 @@ void DefaultVideoQualityAnalyzer::ReportResults(
                     /*important=*/false);
   ReportResult("max_skipped", test_case_name, stats.skipped_between_rendered,
                "unitless");
+  test::PrintResult("bytes_sent", "", test_case_name, stats.bytes_sent,
+                    "sizeInBytes", /*important=*/false);
 }
 
 void DefaultVideoQualityAnalyzer::ReportResult(
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
index e7be2b5af9..ef10d27c54 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -96,6 +96,8 @@ struct StreamStats {
 
   int64_t dropped_by_encoder = 0;
   int64_t dropped_before_encoder = 0;
+
+  int64_t bytes_sent = 0;
 };
 
 struct AnalyzerStats {