diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index e7c2be600e..a2e6c1714e 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -211,7 +211,9 @@ bool IsEnabled(const webrtc::WebRtcKeyValueConfig& config,
 struct AdaptivePtimeConfig {
   bool enabled = false;
   webrtc::DataRate min_payload_bitrate = webrtc::DataRate::KilobitsPerSec(16);
-  webrtc::DataRate min_encoder_bitrate = webrtc::DataRate::KilobitsPerSec(12);
+  // Value is chosen to ensure FEC can be encoded, see LBRR_WB_MIN_RATE_BPS in
+  // libopus.
+  webrtc::DataRate min_encoder_bitrate = webrtc::DataRate::KilobitsPerSec(16);
   bool use_slow_adaptation = true;
 
   absl::optional<std::string> audio_network_adaptor_config;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 7fd138736a..3286837d81 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1218,7 +1218,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) {
   parameters.encodings[0].adaptive_ptime = true;
   EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
   EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX));
-  EXPECT_EQ(12000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
+  EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
 
   parameters.encodings[0].adaptive_ptime = false;
   EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());