diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index ceefb6c555..521bcb7fbb 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -106,20 +106,19 @@ SendAudioStream::SendAudioStream(
       max_rate = *config.encoder.max_rate;
     }
     if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
-      TimeDelta min_frame_length = config.encoder.initial_frame_length;
-      TimeDelta max_frame_length = config.encoder.initial_frame_length;
-      if (field_trial::IsEnabled("WebRTC-Audio-FrameLengthAdaptation") &&
-          !config.adapt.frame.min_rate_for_20_ms.IsZero()) {
-        if (!config.adapt.frame.min_rate_for_60_ms.IsZero()) {
-          max_frame_length = TimeDelta::ms(120);
-        } else {
-          max_frame_length = TimeDelta::ms(60);
-        }
-      }
+      TimeDelta min_frame_length = TimeDelta::ms(20);
+      // Note, depends on WEBRTC_OPUS_SUPPORT_120MS_PTIME being set, which is
+      // the default.
+      TimeDelta max_frame_length = TimeDelta::ms(120);
       DataSize rtp_overhead = DataSize::bytes(12);
+      // Note that this does not include rtp extension overhead and will not
+      // follow updates in the transport overhead over time.
       DataSize total_overhead =
           sender_->transport_.packet_overhead() + rtp_overhead;
+
       min_rate += total_overhead / max_frame_length;
+      // In WebRTCVoiceEngine the max rate is also based on the max frame
+      // length.
       max_rate += total_overhead / min_frame_length;
     }
     send_config.min_bitrate_bps = min_rate.bps();