diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index acd0a3b3a4..8bf6f62ac4 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -65,6 +65,15 @@ webrtc_fuzzer_test("h264_depacketizer_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("generic_depacketizer_fuzzer") {
+  sources = [
+    "generic_depacketizer_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp",
+  ]
+}
+
 webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
   sources = [
     "vp8_depacketizer_fuzzer.cc",
diff --git a/test/fuzzers/generic_depacketizer_fuzzer.cc b/test/fuzzers/generic_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..4775501b36
--- /dev/null
+++ b/test/fuzzers/generic_depacketizer_fuzzer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RtpDepacketizerGeneric depacketizer;
+  RtpDepacketizer::ParsedPayload parsed_payload;
+  depacketizer.Parse(&parsed_payload, data, size);
+}
+
+}  // namespace webrtc