diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 942551b..5c88e01 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -284,7 +284,11 @@
       // send side congestion control, wich depends on feedback packets which
       // requires transport sequence numbers to be enabled.
       if (stream->rtp_transport_) {
-        stream->rtp_transport_->EnablePeriodicAlrProbing(true);
+        // Optionally request ALR probing but do not override any existing
+        // request from other streams.
+        if (stream->allocation_settings_.RequestAlrProbing()) {
+          stream->rtp_transport_->EnablePeriodicAlrProbing(true);
+        }
         bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver();
       }
     }
diff --git a/rtc_base/experiments/audio_allocation_settings.cc b/rtc_base/experiments/audio_allocation_settings.cc
index 66d9d47..62f0c8c 100644
--- a/rtc_base/experiments/audio_allocation_settings.cc
+++ b/rtc_base/experiments/audio_allocation_settings.cc
@@ -17,10 +17,11 @@
 constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
 }  // namespace
 AudioAllocationSettings::AudioAllocationSettings()
-    : audio_send_side_bwe_("Enabled"),
-      allocate_audio_without_feedback_("Enabled"),
-      force_no_audio_feedback_("Enabled"),
-      send_side_bwe_with_overhead_("Enabled"),
+    : audio_send_side_bwe_(/*Flag key*/ "Enabled"),
+      allocate_audio_without_feedback_(/*Flag key*/ "Enabled"),
+      force_no_audio_feedback_(/*Flag key*/ "Enabled"),
+      disable_audio_alr_request_(/*Flag key*/ "Disabled"),
+      send_side_bwe_with_overhead_(/*Flag key*/ "Enabled"),
       min_bitrate_("min"),
       max_bitrate_("max"),
       priority_bitrate_("prio_rate", DataRate::Zero()),
@@ -31,6 +32,8 @@
                   field_trial::FindFullName("WebRTC-Audio-ABWENoTWCC"));
   ParseFieldTrial({&force_no_audio_feedback_},
                   field_trial::FindFullName("WebRTC-Audio-ForceNoTWCC"));
+  ParseFieldTrial({&disable_audio_alr_request_},
+                  field_trial::FindFullName("WebRTC-Audio-AlrProbing"));
 
   ParseFieldTrial({&send_side_bwe_with_overhead_},
                   field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
@@ -68,6 +71,10 @@
          transport_seq_num_extension_header_id != 0;
 }
 
+bool AudioAllocationSettings::RequestAlrProbing() const {
+  return !disable_audio_alr_request_;
+}
+
 bool AudioAllocationSettings::IncludeAudioInAllocationOnStart(
     int min_bitrate_bps,
     int max_bitrate_bps,
diff --git a/rtc_base/experiments/audio_allocation_settings.h b/rtc_base/experiments/audio_allocation_settings.h
index d85b007..c908a57 100644
--- a/rtc_base/experiments/audio_allocation_settings.h
+++ b/rtc_base/experiments/audio_allocation_settings.h
@@ -34,6 +34,8 @@
   // configured.
   bool ShouldSendTransportSequenceNumber(
       int transport_seq_num_extension_header_id) const;
+  // Returns true if audio should request ALR probing from network controller.
+  bool RequestAlrProbing() const;
   // Returns true if audio should be added to rate allocation when the audio
   // stream is started.
   // |min_bitrate_bps| the configured min bitrate, set to -1 if unset.
@@ -80,6 +82,7 @@
   FieldTrialFlag audio_send_side_bwe_;
   FieldTrialFlag allocate_audio_without_feedback_;
   FieldTrialFlag force_no_audio_feedback_;
+  FieldTrialFlag disable_audio_alr_request_;
   FieldTrialFlag send_side_bwe_with_overhead_;
   int min_overhead_bps_ = 0;
   // Field Trial configured bitrates to use as overrides over default/user