diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
index 3798e96..98bb17a 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
@@ -241,7 +241,9 @@
 class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
  public:
   DelayBasedBweTestWithBackoffTimeoutExperiment()
-      : DelayBasedBweTest("WebRTC-BweInitialBackOffInterval/Enabled-200/") {}
+      : DelayBasedBweTest(
+            "WebRTC-BweAimdRateControlConfig/initial_backoff_interval:200ms/") {
+  }
 };
 
 // This test subsumes and improves DelayBasedBweTest.TestInitialOveruse above.
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index e15f375..055ae68 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -51,6 +51,7 @@
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:safe_minmax",
+    "../../rtc_base/experiments:field_trial_parser",
     "../../system_wrappers",
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 1dd87d5..7cd145b 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -17,9 +17,11 @@
 #include <cstdio>
 #include <string>
 
+#include "api/units/data_rate.h"
 #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "modules/remote_bitrate_estimator/overuse_detector.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "system_wrappers/include/field_trial.h"
@@ -27,11 +29,8 @@
 namespace webrtc {
 constexpr TimeDelta kDefaultRtt = TimeDelta::Millis<200>();
 constexpr double kDefaultBackoffFactor = 0.85;
-constexpr TimeDelta kDefaultInitialBackOffInterval = TimeDelta::Millis<200>();
 
 const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
-const char kBweInitialBackOffIntervalExperiment[] =
-    "WebRTC-BweInitialBackOffInterval";
 
 double ReadBackoffFactor() {
   std::string experiment_string =
@@ -53,25 +52,6 @@
   return kDefaultBackoffFactor;
 }
 
-TimeDelta ReadInitialBackoffInterval() {
-  std::string experiment_string =
-      webrtc::field_trial::FindFullName(kBweInitialBackOffIntervalExperiment);
-  int64_t backoff_interval;
-  int parsed_values =
-      sscanf(experiment_string.c_str(), "Enabled-%" SCNd64, &backoff_interval);
-  if (parsed_values == 1) {
-    if (10 <= backoff_interval && backoff_interval <= 200) {
-      return TimeDelta::ms(backoff_interval);
-    }
-    RTC_LOG(WARNING)
-        << "Initial back-off interval must be between 10 and 200 ms.";
-  }
-  RTC_LOG(LS_WARNING) << "Failed to parse parameters for "
-                      << kBweInitialBackOffIntervalExperiment
-                      << " experiment. Using default.";
-  return kDefaultInitialBackOffInterval;
-}
-
 AimdRateControl::AimdRateControl()
     : min_configured_bitrate_(congestion_controller::GetMinBitrate()),
       max_configured_bitrate_(DataRate::kbps(30000)),
@@ -90,13 +70,16 @@
       in_experiment_(!AdaptiveThresholdExperimentIsDisabled()),
       smoothing_experiment_(
           webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")),
-      in_initial_backoff_interval_experiment_(
-          webrtc::field_trial::IsEnabled(kBweInitialBackOffIntervalExperiment)),
-      initial_backoff_interval_(kDefaultInitialBackOffInterval) {
-  if (in_initial_backoff_interval_experiment_) {
-    initial_backoff_interval_ = ReadInitialBackoffInterval();
+      initial_backoff_interval_("initial_backoff_interval"),
+      low_throughput_threshold_("low_throughput", DataRate::Zero()) {
+  // E.g
+  // WebRTC-BweAimdRateControlConfig/initial_backoff_interval:100ms,
+  // critical_low_bitrate:100kbps/
+  ParseFieldTrial({&initial_backoff_interval_, &low_throughput_threshold_},
+                  field_trial::FindFullName("WebRTC-BweAimdRateControlConfig"));
+  if (initial_backoff_interval_) {
     RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
-                     << " " << ToString(initial_backoff_interval_) << ".";
+                     << " " << ToString(*initial_backoff_interval_) << ".";
   }
   RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
 }
@@ -146,7 +129,7 @@
 }
 
 bool AimdRateControl::InitialTimeToReduceFurther(Timestamp at_time) const {
-  if (!in_initial_backoff_interval_experiment_) {
+  if (!initial_backoff_interval_) {
     return ValidEstimate() &&
            TimeToReduceFurther(at_time,
                                LatestEstimate() / 2 - DataRate::bps(1));
@@ -154,7 +137,7 @@
   // TODO(terelius): We could use the RTT (clamped to suitable limits) instead
   // of a fixed bitrate_reduction_interval.
   if (time_last_bitrate_decrease_.IsInfinite() ||
-      at_time - time_last_bitrate_decrease_ >= initial_backoff_interval_) {
+      at_time - time_last_bitrate_decrease_ >= *initial_backoff_interval_) {
     return true;
   }
   return false;
@@ -280,16 +263,24 @@
       break;
 
     case kRcDecrease:
-      // Set bit rate to something slightly lower than the measured throughput
-      // to get rid of any self-induced delay.
-      new_bitrate = estimated_throughput * beta_;
-      if (new_bitrate > current_bitrate_) {
-        // Avoid increasing the rate when over-using.
-        if (link_capacity_.has_estimate()) {
-          new_bitrate = beta_ * link_capacity_.estimate();
+      if (estimated_throughput > low_throughput_threshold_) {
+        // Set bit rate to something slightly lower than the measured throughput
+        // to get rid of any self-induced delay.
+        new_bitrate = estimated_throughput * beta_;
+        if (new_bitrate > current_bitrate_) {
+          // Avoid increasing the rate when over-using.
+          if (link_capacity_.has_estimate()) {
+            new_bitrate = beta_ * link_capacity_.estimate();
+          }
         }
-        new_bitrate = std::min(new_bitrate, current_bitrate_);
+      } else {
+        new_bitrate = estimated_throughput;
+        if (link_capacity_.has_estimate()) {
+          new_bitrate = std::max(new_bitrate, link_capacity_.estimate());
+        }
+        new_bitrate = std::min(new_bitrate, low_throughput_threshold_.Get());
       }
+      new_bitrate = std::min(new_bitrate, current_bitrate_);
 
       if (bitrate_is_initialized_ && estimated_throughput < current_bitrate_) {
         constexpr double kDegradationFactor = 0.9;
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.h b/modules/remote_bitrate_estimator/aimd_rate_control.h
index 7a4fd02..4d7bc7e 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.h
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -18,6 +18,7 @@
 #include "api/units/timestamp.h"
 #include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
 #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/experiments/field_trial_parser.h"
 
 namespace webrtc {
 // A rate control implementation based on additive increases of
@@ -95,9 +96,9 @@
   TimeDelta rtt_;
   const bool in_experiment_;
   const bool smoothing_experiment_;
-  const bool in_initial_backoff_interval_experiment_;
-  TimeDelta initial_backoff_interval_;
   absl::optional<DataRate> last_decrease_;
+  FieldTrialOptional<TimeDelta> initial_backoff_interval_;
+  FieldTrialParameter<DataRate> low_throughput_threshold_;
 };
 }  // namespace webrtc