diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index fc7e12ef38..d3f2d87c69 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -198,6 +198,7 @@ if (rtc_include_tests) {
       "../modules/rtp_rtcp:rtp_rtcp_format",
       "../modules/utility:utility",
       "../rtc_base:checks",
+      "../rtc_base:gunit_helpers",
       "../rtc_base:macromagic",
       "../rtc_base:refcount",
       "../rtc_base:rtc_base_tests_utils",
diff --git a/audio/test/non_sender_rtt_test.cc b/audio/test/non_sender_rtt_test.cc
index 0c7dc6cbee..278193e335 100644
--- a/audio/test/non_sender_rtt_test.cc
+++ b/audio/test/non_sender_rtt_test.cc
@@ -9,6 +9,8 @@
  */
 
 #include "audio/test/audio_end_to_end_test.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/task_queue_for_test.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/gtest.h"
 
@@ -20,9 +22,12 @@ using NonSenderRttTest = CallTest;
 TEST_F(NonSenderRttTest, NonSenderRttStats) {
   class NonSenderRttTest : public AudioEndToEndTest {
    public:
-    const int kTestDurationMs = 10000;
+    const int kLongTimeoutMs = 20000;
     const int64_t kRttMs = 30;
 
+    explicit NonSenderRttTest(TaskQueueBase* task_queue)
+        : task_queue_(task_queue) {}
+
     BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
       BuiltInNetworkBehaviorConfig pipe_config;
       pipe_config.queue_delay_ms = kRttMs / 2;
@@ -38,7 +43,12 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) {
       send_config->send_codec_spec->enable_non_sender_rtt = true;
     }
 
-    void PerformTest() override { SleepMs(kTestDurationMs); }
+    void PerformTest() override {
+      // Wait until we have an RTT measurement, but no longer than
+      // `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare
+      // cases it can take more than 10 seconds.
+      EXPECT_TRUE_WAIT(HasRoundTripTimeMeasurement(), kLongTimeoutMs);
+    }
 
     void OnStreamsStopped() override {
       AudioReceiveStreamInterface::Stats recv_stats =
@@ -49,7 +59,23 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) {
       EXPECT_GE(recv_stats.total_round_trip_time.ms(),
                 recv_stats.round_trip_time->ms());
     }
-  } test;
+
+   protected:
+    bool HasRoundTripTimeMeasurement() {
+      bool has_rtt = false;
+      // GetStats() can only be called on `task_queue_`, block while we check.
+      SendTask(task_queue_, [this, &has_rtt]() {
+        if (receive_stream() &&
+            receive_stream()->GetStats(true).round_trip_time_measurements > 0) {
+          has_rtt = true;
+        }
+      });
+      return has_rtt;
+    }
+
+   private:
+    TaskQueueBase* task_queue_;
+  } test(task_queue());
 
   RunBaseTest(&test);
 }
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 1228861c42..1d33bf2e0b 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -99,7 +99,7 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
     uint64_t sender_reports_reports_count = 0;
     absl::optional<TimeDelta> round_trip_time;
     TimeDelta total_round_trip_time = TimeDelta::Zero();
-    int round_trip_time_measurements;
+    int round_trip_time_measurements = 0;
   };
 
   struct Config {