diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 8822022103..560fb4c16a 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1990,31 +1990,42 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::SetCodec(
 
   parameters_.codec_settings = codec_settings;
 
-  // Settings for mixed-codec simulcast
+  // Settings for mixed-codec simulcast.
   if (!codec_settings_list.empty()) {
-    RTC_DCHECK_EQ(parameters_.config.rtp.ssrcs.size(),
-                  codec_settings_list.size());
-    parameters_.config.rtp.stream_configs.resize(
-        parameters_.config.rtp.ssrcs.size());
-    for (size_t i = 0; i < codec_settings_list.size(); i++) {
-      auto& stream_config = parameters_.config.rtp.stream_configs[i];
-      const auto& cs = codec_settings_list[i];
-      stream_config.ssrc = parameters_.config.rtp.ssrcs[i];
-      if (i < parameters_.config.rtp.rids.size()) {
-        stream_config.rid = parameters_.config.rtp.rids[i];
-      }
-      stream_config.payload_name = cs.codec.name;
-      stream_config.payload_type = cs.codec.id;
-      stream_config.raw_payload =
-          cs.codec.packetization == kPacketizationParamRaw;
-      if (i < parameters_.config.rtp.rtx.ssrcs.size()) {
-        auto& rtx = stream_config.rtx.emplace(
-            decltype(stream_config.rtx)::value_type());
-        rtx.ssrc = parameters_.config.rtp.rtx.ssrcs[i];
-        rtx.payload_type = cs.rtx_payload_type;
+    if (parameters_.config.rtp.ssrcs.size() == codec_settings_list.size()) {
+      parameters_.config.rtp.stream_configs.resize(
+          parameters_.config.rtp.ssrcs.size());
+      for (size_t i = 0; i < codec_settings_list.size(); i++) {
+        auto& stream_config = parameters_.config.rtp.stream_configs[i];
+        const auto& cs = codec_settings_list[i];
+        stream_config.ssrc = parameters_.config.rtp.ssrcs[i];
+        if (i < parameters_.config.rtp.rids.size()) {
+          stream_config.rid = parameters_.config.rtp.rids[i];
+        }
+        stream_config.payload_name = cs.codec.name;
+        stream_config.payload_type = cs.codec.id;
+        stream_config.raw_payload =
+            cs.codec.packetization == kPacketizationParamRaw;
+        if (i < parameters_.config.rtp.rtx.ssrcs.size()) {
+          auto& rtx = stream_config.rtx.emplace(
+              decltype(stream_config.rtx)::value_type());
+          rtx.ssrc = parameters_.config.rtp.rtx.ssrcs[i];
+          rtx.payload_type = cs.rtx_payload_type;
+        }
       }
+    } else {
+      // TODO(crbug.com/378724147): We need to investigate when it
+      // has mismatched sizes.
+      RTC_DCHECK_EQ(parameters_.config.rtp.ssrcs.size(),
+                    codec_settings_list.size());
+
+      RTC_LOG(LS_ERROR) << "Mismatched sizes between codec_settings_list:"
+                        << codec_settings_list.size()
+                        << ", parameters_.config.rtp.ssrcs:"
+                        << parameters_.config.rtp.ssrcs.size();
     }
   }
+
   parameters_.codec_settings_list = codec_settings_list;
 
   // TODO(bugs.webrtc.org/8830): Avoid recreation, it should be enough to call
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 91249da4bd..a76b209dd2 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -9225,6 +9225,33 @@ TEST_F(WebRtcVideoChannelTest, SetMixedCodecSimulcastStreamConfig) {
   EXPECT_EQ(config.rtp.stream_configs[2].payload_type, vp9.id);
 }
 
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(WebRtcVideoChannelTest,
+       SetMixedCodecSimulcastWithDifferentConfigSettingsSizes) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-MixedCodecSimulcast/Enabled/");
+  AddSendStream();
+
+  cricket::VideoSenderParameters parameters;
+  cricket::Codec vp8 = GetEngineCodec("VP8");
+  parameters.codecs.push_back(vp8);
+
+  // `codec_settings_list.size()` is 1 after this in the
+  EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
+
+  // It sets 2 sizes of config ssrc.
+  StreamParams sp = CreateSimStreamParams("cname", {123, 456});
+  std::vector<cricket::RidDescription> rid_descriptions2;
+  rid_descriptions2.emplace_back("f", cricket::RidDirection::kSend);
+  rid_descriptions2.emplace_back("h", cricket::RidDirection::kSend);
+  sp.set_rids(rid_descriptions2);
+
+  // `WebRtcVideoSendStream::SetCodec` test for different sizes
+  // between parameters_.config.rtp.ssrcs.size() and codec_settings_list.size().
+  EXPECT_DEATH(send_channel_->AddSendStream(sp), "");
+}
+#endif
+
 // Test that min and max bitrate values set via RtpParameters are correctly
 // propagated to the underlying encoder for a single stream.
 TEST_F(WebRtcVideoChannelTest, MinAndMaxBitratePropagatedToEncoder) {