diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index b953705..e90a9a6 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -394,6 +394,7 @@
       encoder_target_rate_bps_(0),
       frame_counts_(rtp_config.ssrcs.size()),
       frame_count_observer_(observers.frame_count_observer) {
+  transport_checker_.Detach();
   RTC_DCHECK_EQ(rtp_config_.ssrcs.size(), rtp_streams_.size());
   if (send_side_bwe_with_overhead_ && has_packet_feedback_)
     transport_->IncludeOverheadInPacedSender();
@@ -460,6 +461,10 @@
 }
 
 RtpVideoSender::~RtpVideoSender() {
+  // TODO(bugs.webrtc.org/13517): Remove once RtpVideoSender gets deleted on the
+  // transport task queue.
+  transport_checker_.Detach();
+
   SetActiveModulesLocked(
       std::vector<bool>(rtp_streams_.size(), /*active=*/false));
   transport_->GetStreamFeedbackProvider()->DeRegisterStreamFeedbackObserver(
@@ -467,6 +472,7 @@
 }
 
 void RtpVideoSender::SetActive(bool active) {
+  RTC_DCHECK_RUN_ON(&transport_checker_);
   MutexLock lock(&mutex_);
   if (active_ == active)
     return;
@@ -475,12 +481,14 @@
 }
 
 void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
+  RTC_DCHECK_RUN_ON(&transport_checker_);
   MutexLock lock(&mutex_);
   return SetActiveModulesLocked(active_modules);
 }
 
 void RtpVideoSender::SetActiveModulesLocked(
     const std::vector<bool> active_modules) {
+  RTC_DCHECK_RUN_ON(&transport_checker_);
   RTC_DCHECK_EQ(rtp_streams_.size(), active_modules.size());
   active_ = false;
   for (size_t i = 0; i < active_modules.size(); ++i) {
@@ -514,6 +522,7 @@
 }
 
 bool RtpVideoSender::IsActive() {
+  RTC_DCHECK_RUN_ON(&transport_checker_);
   MutexLock lock(&mutex_);
   return IsActiveLocked();
 }
@@ -622,6 +631,7 @@
 
 void RtpVideoSender::OnBitrateAllocationUpdated(
     const VideoBitrateAllocation& bitrate) {
+  RTC_DCHECK_RUN_ON(&transport_checker_);
   MutexLock lock(&mutex_);
   if (IsActiveLocked()) {
     if (rtp_streams_.size() == 1) {
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 7e5de98..25ef20f 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -170,7 +170,11 @@
   const bool has_packet_feedback_;
   const bool simulate_generic_structure_;
 
-  // TODO(holmer): Remove mutex_ once RtpVideoSender runs on the
+  // Semantically equivalent to checking for `transport_->GetWorkerQueue()`
+  // but some tests need to be updated to call from the correct context.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker transport_checker_;
+
+  // TODO(bugs.webrtc.org/13517): Remove mutex_ once RtpVideoSender runs on the
   // transport task queue.
   mutable Mutex mutex_;
   bool active_ RTC_GUARDED_BY(mutex_);