diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index d7dc9ea1fa..4679c6980b 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -191,12 +191,11 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
   ]
 }
 
-webrtc_fuzzer_test("packet_buffer_fuzzer") {
-  sources = [ "packet_buffer_fuzzer.cc" ]
+webrtc_fuzzer_test("rtp_video_frame_assembler_fuzzer") {
+  sources = [ "rtp_video_frame_assembler_fuzzer.cc" ]
   deps = [
-    "../../modules/video_coding:packet_buffer",
-    "../../modules/video_coding/",
-    "../../system_wrappers",
+    "../../api/video:rtp_video_frame_assembler",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
   ]
 }
 
diff --git a/test/fuzzers/packet_buffer_fuzzer.cc b/test/fuzzers/packet_buffer_fuzzer.cc
deleted file mode 100644
index 3c4e9688ab..0000000000
--- a/test/fuzzers/packet_buffer_fuzzer.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2017 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 <memory>
-#include <utility>
-
-#include "modules/video_coding/frame_object.h"
-#include "modules/video_coding/packet_buffer.h"
-#include "test/fuzzers/fuzz_data_helper.h"
-
-namespace webrtc {
-
-void IgnoreResult(video_coding::PacketBuffer::InsertResult result) {}
-
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 200000) {
-    return;
-  }
-  video_coding::PacketBuffer packet_buffer(8, 1024);
-  test::FuzzDataHelper helper(rtc::ArrayView<const uint8_t>(data, size));
-
-  while (helper.BytesLeft()) {
-    auto packet = std::make_unique<video_coding::PacketBuffer::Packet>();
-    // Fuzz POD members of the packet.
-    helper.CopyTo(&packet->marker_bit);
-    helper.CopyTo(&packet->payload_type);
-    helper.CopyTo(&packet->seq_num);
-    helper.CopyTo(&packet->timestamp);
-    helper.CopyTo(&packet->times_nacked);
-
-    // Fuzz non-POD member of the packet.
-    packet->video_payload.SetSize(helper.ReadOrDefaultValue<uint8_t>(0));
-    // TODO(danilchap): Fuzz other non-POD members of the `packet`.
-
-    IgnoreResult(packet_buffer.InsertPacket(std::move(packet)));
-  }
-}
-
-}  // namespace webrtc
diff --git a/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc b/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
new file mode 100644
index 0000000000..6ab6b9a905
--- /dev/null
+++ b/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2023 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 <algorithm>
+#include <cstddef>
+#include <cstdint>
+
+#include "api/video/rtp_video_frame_assembler.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size == 0) {
+    return;
+  }
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<RtpDependencyDescriptorExtension>(1);
+  extensions.Register<RtpGenericFrameDescriptorExtension00>(2);
+  RtpPacketReceived rtp_packet(&extensions);
+
+  RtpVideoFrameAssembler assembler(
+      static_cast<RtpVideoFrameAssembler::PayloadFormat>(data[0] % 6));
+
+  for (size_t i = 1; i < size;) {
+    size_t packet_size = std::min<size_t>(size - i, 300);
+    if (rtp_packet.Parse(data + i, packet_size)) {
+      assembler.InsertPacket(rtp_packet);
+    }
+    i += packet_size;
+  }
+}
+
+}  // namespace webrtc