diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 841d7e3b94..525acf7b99 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -40,25 +40,14 @@ size_t RemoveFromMapByValue(Map* map, const Value& value) {
   return EraseIf(*map, [&](const auto& elem) { return elem.second == value; });
 }
 
-// Temp fix: MID in SDP is allowed to be slightly longer than what's allowed
-// in the RTP demuxer. Truncate if needed; this won't match, but it only
-// makes sense in places that wouldn't use this for matching anyway.
-// TODO(bugs.webrtc.org/12517): remove when length 16 is policed by parser.
-std::string CheckMidLength(absl::string_view mid) {
-  std::string new_mid(mid);
-  if (new_mid.length() > BaseRtpStringExtension::kMaxValueSizeBytes) {
-    RTC_LOG(LS_WARNING) << "`mid` attribute too long. Truncating.";
-    new_mid.resize(BaseRtpStringExtension::kMaxValueSizeBytes);
-  }
-  return new_mid;
-}
-
 }  // namespace
 
 RtpDemuxerCriteria::RtpDemuxerCriteria(
     absl::string_view mid,
     absl::string_view rsid /*= absl::string_view()*/)
-    : mid_(CheckMidLength(mid)), rsid_(rsid) {}
+    : mid_(mid), rsid_(rsid) {
+  RTC_DCHECK(mid.length() <= BaseRtpStringExtension::kMaxValueSizeBytes);
+}
 
 RtpDemuxerCriteria::RtpDemuxerCriteria() = default;
 RtpDemuxerCriteria::~RtpDemuxerCriteria() = default;
diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc
index e85052810a..e460af8536 100644
--- a/call/rtp_demuxer_unittest.cc
+++ b/call/rtp_demuxer_unittest.cc
@@ -1238,26 +1238,14 @@ TEST_F(RtpDemuxerTest, PacketWithMidAndUnknownRsidIsNotRoutedByPayloadType) {
   EXPECT_FALSE(demuxer_.OnRtpPacket(*packet));
 }
 
-TEST_F(RtpDemuxerTest, MidMustNotExceedMaximumLength) {
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST_F(RtpDemuxerDeathTest, MidMustNotExceedMaximumLength) {
   MockRtpPacketSink sink1;
   std::string mid1(BaseRtpStringExtension::kMaxValueSizeBytes + 1, 'a');
-  // Adding the sink should pass even though the supplied mid is too long.
-  // The mid will be truncated though.
-  EXPECT_TRUE(AddSinkOnlyMid(mid1, &sink1));
-
-  // Adding a second sink with a mid that matches the truncated mid that was
-  // just added, should fail.
-  MockRtpPacketSink sink2;
-  std::string mid2(mid1.substr(0, BaseRtpStringExtension::kMaxValueSizeBytes));
-  EXPECT_FALSE(AddSinkOnlyMid(mid2, &sink2));
-  EXPECT_FALSE(RemoveSink(&sink2));
-
-  // Remove the original sink.
-  EXPECT_TRUE(RemoveSink(&sink1));
+  EXPECT_DEATH(AddSinkOnlyMid(mid1, &sink1), "");
 }
 
-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-
 TEST_F(RtpDemuxerDeathTest, CriteriaMustBeNonEmpty) {
   MockRtpPacketSink sink;
   RtpDemuxerCriteria criteria;