diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index c831b5132a..97469ca35e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -556,6 +556,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
   {
     rtc::CritScope lock(&send_critsect_);
+    if (!sending_media_)
+      return 0;
     if ((rtx_ & kRtxRedundantPayloads) == 0)
       return 0;
   }
@@ -618,6 +620,8 @@ size_t RTPSender::SendPadData(size_t bytes,
     bool over_rtx;
     {
       rtc::CritScope lock(&send_critsect_);
+      if (!sending_media_)
+        return bytes_sent;
       if (!timestamp_provided) {
         timestamp = timestamp_;
         capture_time_ms = capture_time_ms_;
@@ -1011,11 +1015,6 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
   if (audio_configured_ || bytes == 0)
     return 0;
-  {
-    rtc::CritScope lock(&send_critsect_);
-    if (!sending_media_)
-      return 0;
-  }
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 6de9e483d8..1c804098e8 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -3148,7 +3148,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
   DestroyStreams();
 }
 
-TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
+TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpState) {
   TestRtpStatePreservation(false);
 }