diff --git a/call/call.cc b/call/call.cc
index 437f4df2fb..9defa67b91 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1363,6 +1363,19 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
+
+  // TODO(bugs.webrtc.org/11993): This DCHECK is here just to maintain the
+  // invariant that currently the only call path to this function is via
+  // `PeerConnection::InitializeRtcpCallback()`. DeliverRtp on the other hand
+  // gets called via the channel classes and
+  // WebRtc[Audio|Video]Channel's `OnPacketReceived`. We'll remove the
+  // PeerConnection involvement as well as
+  // `JsepTransportController::OnRtcpPacketReceived_n` and `rtcp_handler`
+  // and make sure that the flow of packets is consistent from the
+  // `RtpTransport` class, via the *Channel and *Engine classes and into Call.
+  // This way we'll also know more about the context of the packet.
+  RTC_DCHECK_EQ(media_type, MediaType::ANY);
+
   // TODO(pbos): Make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
@@ -1409,6 +1422,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 rtc::CopyOnWriteBuffer packet,
                                                 int64_t packet_time_us) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
+  RTC_DCHECK_NE(media_type, MediaType::ANY);
 
   RtpPacketReceived parsed_packet;
   if (!parsed_packet.Parse(std::move(packet)))