diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 9fc4ba1..c09aefd 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -91,7 +91,7 @@
   return sb.Release();
 }
 
-RtpDemuxer::RtpDemuxer() = default;
+RtpDemuxer::RtpDemuxer(bool use_mid /* = true*/) : use_mid_(use_mid) {}
 
 RtpDemuxer::~RtpDemuxer() {
   RTC_DCHECK(sink_by_mid_.empty());
diff --git a/call/rtp_demuxer.h b/call/rtp_demuxer.h
index 3aa7e9d..b89f154 100644
--- a/call/rtp_demuxer.h
+++ b/call/rtp_demuxer.h
@@ -94,7 +94,7 @@
   // relevant for demuxing.
   static std::string DescribePacket(const RtpPacketReceived& packet);
 
-  RtpDemuxer();
+  explicit RtpDemuxer(bool use_mid = true);
   ~RtpDemuxer();
 
   RtpDemuxer(const RtpDemuxer&) = delete;
@@ -132,10 +132,6 @@
   // if the packet was forwarded and false if the packet was dropped.
   bool OnRtpPacket(const RtpPacketReceived& packet);
 
-  // Configure whether to look at the MID header extension when demuxing
-  // incoming RTP packets. By default this is enabled.
-  void set_use_mid(bool use_mid) { use_mid_ = use_mid; }
-
  private:
   // Returns true if adding a sink with the given criteria would cause conflicts
   // with the existing criteria and should be rejected.
@@ -191,7 +187,7 @@
   // Adds a binding from the SSRC to the given sink.
   void AddSsrcSinkBinding(uint32_t ssrc, RtpPacketSinkInterface* sink);
 
-  bool use_mid_ = true;
+  const bool use_mid_;
 };
 
 }  // namespace webrtc
diff --git a/call/rtp_stream_receiver_controller.cc b/call/rtp_stream_receiver_controller.cc
index f440b42..7150b34 100644
--- a/call/rtp_stream_receiver_controller.cc
+++ b/call/rtp_stream_receiver_controller.cc
@@ -37,11 +37,7 @@
   controller_->RemoveSink(sink_);
 }
 
-RtpStreamReceiverController::RtpStreamReceiverController() {
-  // At this level the demuxer is only configured to demux by SSRC, so don't
-  // worry about MIDs (MIDs are handled by upper layers).
-  demuxer_.set_use_mid(false);
-}
+RtpStreamReceiverController::RtpStreamReceiverController() {}
 
 RtpStreamReceiverController::~RtpStreamReceiverController() = default;
 
@@ -52,19 +48,19 @@
 }
 
 bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
-  rtc::CritScope cs(&lock_);
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
   return demuxer_.OnRtpPacket(packet);
 }
 
 bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
                                           RtpPacketSinkInterface* sink) {
-  rtc::CritScope cs(&lock_);
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
   return demuxer_.AddSink(ssrc, sink);
 }
 
 size_t RtpStreamReceiverController::RemoveSink(
     const RtpPacketSinkInterface* sink) {
-  rtc::CritScope cs(&lock_);
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
   return demuxer_.RemoveSink(sink);
 }
 
diff --git a/call/rtp_stream_receiver_controller.h b/call/rtp_stream_receiver_controller.h
index 62447aa..2611561 100644
--- a/call/rtp_stream_receiver_controller.h
+++ b/call/rtp_stream_receiver_controller.h
@@ -14,7 +14,7 @@
 
 #include "call/rtp_demuxer.h"
 #include "call/rtp_stream_receiver_controller_interface.h"
-#include "rtc_base/deprecated/recursive_critical_section.h"
+#include "rtc_base/synchronization/sequence_checker.h"
 
 namespace webrtc {
 
@@ -58,13 +58,18 @@
     RtpPacketSinkInterface* const sink_;
   };
 
-  // TODO(nisse): Move to a TaskQueue for synchronization. When used
-  // by Call, we expect construction and all methods but OnRtpPacket
-  // to be called on the same thread, and OnRtpPacket to be called
-  // by a single, but possibly distinct, thread. But applications not
-  // using Call may have use threads differently.
-  rtc::RecursiveCriticalSection lock_;
-  RtpDemuxer demuxer_ RTC_GUARDED_BY(&lock_);
+  // TODO(bugs.webrtc.org/11993): We expect construction and all methods to be
+  // called on the same thread/tq. Currently this is the worker thread
+  // (including OnRtpPacket) but a more natural fit would be the network thread.
+  // Using a sequence checker to ensure that usage is correct but at the same
+  // time not require a specific thread/tq, an instance of this class + the
+  // associated functionality should be easily moved from one execution context
+  // to another (i.e. when network packets don't hop to the worker thread inside
+  // of Call).
+  SequenceChecker demuxer_sequence_;
+  // At this level the demuxer is only configured to demux by SSRC, so don't
+  // worry about MIDs (MIDs are handled by upper layers).
+  RtpDemuxer demuxer_ RTC_GUARDED_BY(&demuxer_sequence_){false /*use_mid*/};
 };
 
 }  // namespace webrtc