diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index ba318ca040..b7560f1b34 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -602,7 +602,9 @@ struct RtpParameters {
   // abstraction on which RTCP parameters are set.
   RtcpParameters rtcp;
 
-  // TODO(deadbeef): Not implemented.
+  // When bandwidth is constrained and the RtpSender needs to choose between
+  // degrading resolution or degrading framerate, degradationPreference
+  // indicates which is preferred. Only for video tracks.
   DegradationPreference degradation_preference =
       DegradationPreference::BALANCED;
 
diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc
index 70353fe45e..84623f17a2 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -288,7 +288,6 @@ void FakeVideoSendStream::Stop() {
 void FakeVideoSendStream::SetSource(
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
     const webrtc::DegradationPreference& degradation_preference) {
-  RTC_DCHECK(source != source_);
   if (source_)
     source_->RemoveSink(this);
   source_ = source;
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 134fa40489..d75b48b00c 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1693,19 +1693,23 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const {
   // |this| acts like a VideoSource to make sure SinkWants are handled on the
   // correct thread.
   webrtc::DegradationPreference degradation_preference;
-  if (!enable_cpu_overuse_detection_) {
+  if (rtp_parameters_.degradation_preference !=
+      webrtc::DegradationPreference::BALANCED) {
+    // If the degradationPreference is different from the default value, assume
+    // it is what we want, regardless of trials or other internal settings.
+    degradation_preference = rtp_parameters_.degradation_preference;
+  } else if (!enable_cpu_overuse_detection_) {
     degradation_preference = webrtc::DegradationPreference::DISABLED;
+  } else if (parameters_.options.is_screencast.value_or(false)) {
+    degradation_preference = webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
+  } else if (webrtc::field_trial::IsEnabled(
+                 "WebRTC-Video-BalancedDegradation")) {
+    degradation_preference = webrtc::DegradationPreference::BALANCED;
   } else {
-    if (parameters_.options.is_screencast.value_or(false)) {
-      degradation_preference =
-          webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
-    } else if (webrtc::field_trial::IsEnabled(
-                   "WebRTC-Video-BalancedDegradation")) {
-      degradation_preference = webrtc::DegradationPreference::BALANCED;
-    } else {
-      degradation_preference =
-          webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
-    }
+    // TODO(orphis): The default should be BALANCED as the standard mandates.
+    // Right now, there is no way to set it to BALANCED as it would change
+    // the behavior for any project expecting MAINTAIN_FRAMERATE by default.
+    degradation_preference = webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
   }
   return degradation_preference;
 }
@@ -1812,6 +1816,12 @@ webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
     }
   }
 
+  bool new_degradation_preference = false;
+  if (new_parameters.degradation_preference !=
+      rtp_parameters_.degradation_preference) {
+    new_degradation_preference = true;
+  }
+
   // TODO(bugs.webrtc.org/8807): The bitrate priority really doesn't require an
   // entire encoder reconfiguration, it just needs to update the bitrate
   // allocator.
@@ -1838,6 +1848,9 @@ webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
   if (new_send_state) {
     UpdateSendState();
   }
+  if (new_degradation_preference) {
+    stream_->SetSource(this, GetDegradationPreference());
+  }
   return webrtc::RTCError::OK();
 }
 
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index 1d56bffa27..e42e3a5f19 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -6035,6 +6035,30 @@ TEST_F(WebRtcVideoChannelTest, DetectRtpSendParameterHeaderExtensionsChange) {
   EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
 }
 
+TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersDegradationPreference) {
+  AddSendStream();
+
+  FakeVideoCapturerWithTaskQueue capturer;
+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
+
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpSendParameters(last_ssrc_);
+  EXPECT_EQ(rtp_parameters.degradation_preference,
+            webrtc::DegradationPreference::BALANCED);
+  rtp_parameters.degradation_preference =
+      webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
+
+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
+
+  webrtc::RtpParameters updated_rtp_parameters =
+      channel_->GetRtpSendParameters(last_ssrc_);
+  EXPECT_EQ(updated_rtp_parameters.degradation_preference,
+            webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
+
+  // Remove the source since it will be destroyed before the channel
+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
+}
+
 // Test that if we set/get parameters multiple times, we get the same results.
 TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) {
   AddSendStream();
diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index 927379d588..91708fa225 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -67,8 +67,7 @@ bool PerSenderRtpEncodingParameterHasValue(
 // Returns true if any RtpParameters member that isn't implemented contains a
 // value.
 bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
-  if (!parameters.mid.empty() ||
-      parameters.degradation_preference != DegradationPreference::BALANCED) {
+  if (!parameters.mid.empty()) {
     return true;
   }
   for (size_t i = 0; i < parameters.encodings.size(); ++i) {
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 3e0488dcb4..4014df54cb 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -675,18 +675,12 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
 
-  // Unimplemented RtpParameters: mid, header_extensions,
-  // degredation_preference.
+  // Unimplemented RtpParameters: mid
   params.mid = "dummy_mid";
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
             audio_rtp_sender_->SetParameters(params).type());
   params = audio_rtp_sender_->GetParameters();
 
-  ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
-  params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
-            audio_rtp_sender_->SetParameters(params).type());
-
   DestroyAudioRtpSender();
 }
 
@@ -868,18 +862,12 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
 
-  // Unimplemented RtpParameters: mid, header_extensions,
-  // degredation_preference.
+  // Unimplemented RtpParameters: mid
   params.mid = "dummy_mid";
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
             video_rtp_sender_->SetParameters(params).type());
   params = video_rtp_sender_->GetParameters();
 
-  ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
-  params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
-            video_rtp_sender_->SetParameters(params).type());
-
   DestroyVideoRtpSender();
 }