diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index bd8a6a67c6..29fc2b6020 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -137,7 +137,23 @@ class RtpVideoSenderTestFixture {
 };
 }  // namespace
 
-TEST(RtpVideoSenderTest, SendOnOneModule) {
+class RtpVideoSenderTest : public ::testing::Test,
+                           public ::testing::WithParamInterface<std::string> {
+ public:
+  RtpVideoSenderTest() : field_trial_(GetParam()) {}
+
+ private:
+  test::ScopedFieldTrials field_trial_;
+};
+
+INSTANTIATE_TEST_CASE_P(Default, RtpVideoSenderTest, ::testing::Values(""));
+
+INSTANTIATE_TEST_CASE_P(
+    TaskQueueTrial,
+    RtpVideoSenderTest,
+    ::testing::Values("WebRTC-TaskQueueCongestionControl/Enabled/"));
+
+TEST_P(RtpVideoSenderTest, SendOnOneModule) {
   uint8_t payload = 'a';
   EncodedImage encoded_image;
   encoded_image.SetTimestamp(1);
@@ -167,7 +183,7 @@ TEST(RtpVideoSenderTest, SendOnOneModule) {
       test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
 }
 
-TEST(RtpVideoSenderTest, SendSimulcastSetActive) {
+TEST_P(RtpVideoSenderTest, SendSimulcastSetActive) {
   uint8_t payload = 'a';
   EncodedImage encoded_image_1;
   encoded_image_1.SetTimestamp(1);
@@ -211,7 +227,7 @@ TEST(RtpVideoSenderTest, SendSimulcastSetActive) {
 // behavior of the payload router. First sets one module to active and checks
 // that outgoing data can be sent on this module, and checks that no data can
 // be sent if both modules are inactive.
-TEST(RtpVideoSenderTest, SendSimulcastSetActiveModules) {
+TEST_P(RtpVideoSenderTest, SendSimulcastSetActiveModules) {
   uint8_t payload = 'a';
   EncodedImage encoded_image_1;
   encoded_image_1.SetTimestamp(1);
@@ -252,7 +268,7 @@ TEST(RtpVideoSenderTest, SendSimulcastSetActiveModules) {
                 .error);
 }
 
-TEST(RtpVideoSenderTest, CreateWithNoPreviousStates) {
+TEST_P(RtpVideoSenderTest, CreateWithNoPreviousStates) {
   RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, kPayloadType, {});
   test.router()->SetActive(true);
 
@@ -263,7 +279,7 @@ TEST(RtpVideoSenderTest, CreateWithNoPreviousStates) {
   EXPECT_NE(initial_states.find(kSsrc2), initial_states.end());
 }
 
-TEST(RtpVideoSenderTest, CreateWithPreviousStates) {
+TEST_P(RtpVideoSenderTest, CreateWithPreviousStates) {
   const int64_t kState1SharedFrameId = 123;
   const int64_t kState2SharedFrameId = 234;
   RtpPayloadState state1;
diff --git a/modules/congestion_controller/rtp/send_side_congestion_controller.cc b/modules/congestion_controller/rtp/send_side_congestion_controller.cc
index e0d8f145cb..70a8d496bb 100644
--- a/modules/congestion_controller/rtp/send_side_congestion_controller.cc
+++ b/modules/congestion_controller/rtp/send_side_congestion_controller.cc
@@ -368,6 +368,15 @@ SendSideCongestionController::SendSideCongestionController(
   initial_config_.constraints = ConvertConstraints(
       min_bitrate_bps, max_bitrate_bps, start_bitrate_bps, clock_);
   RTC_DCHECK(start_bitrate_bps > 0);
+  // To be fully compatible with legacy SendSideCongestionController, make sure
+  // pacer is initialized even if there are no registered streams. This should
+  // not happen under normal circumstances, but some tests rely on it and there
+  // are no checks detecting when the legacy SendSideCongestionController is
+  // used. This way of setting the value has the drawback that it might be wrong
+  // compared to what the actual value from the congestion controller will be.
+  // TODO(srte): Remove this when the legacy SendSideCongestionController is
+  // removed.
+  pacer_->SetEstimatedBitrate(start_bitrate_bps);
 }
 
 // There is no point in having a network controller for a network that is not