diff --git a/test/rtp_file_writer.cc b/test/rtp_file_writer.cc
index 793e51a..d9e0586 100644
--- a/test/rtp_file_writer.cc
+++ b/test/rtp_file_writer.cc
@@ -40,7 +40,6 @@
 
   bool WritePacket(const RtpPacket* packet) override {
     uint16_t len = static_cast<uint16_t>(packet->length + kPacketHeaderSize);
-    RTC_CHECK_GE(packet->original_length, packet->length);
     uint16_t plen = static_cast<uint16_t>(packet->original_length);
     uint32_t offset = packet->time_ms;
     RTC_CHECK(WriteUint16(len));
diff --git a/video/rtc_event_log2rtp_dump.cc b/video/rtc_event_log2rtp_dump.cc
new file mode 100644
index 0000000..4f1d93b
--- /dev/null
+++ b/video/rtc_event_log2rtp_dump.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2015 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 <iostream>
+#include <sstream>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtp_file_writer.h"
+#include "webrtc/video/rtc_event_log.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
+#else
+#include "webrtc/video/rtc_event_log.pb.h"
+#endif
+
+namespace {
+
+DEFINE_bool(noaudio,
+            false,
+            "Excludes audio packets from the converted RTPdump file.");
+DEFINE_bool(novideo,
+            false,
+            "Excludes video packets from the converted RTPdump file.");
+DEFINE_bool(nodata,
+            false,
+            "Excludes data packets from the converted RTPdump file.");
+DEFINE_bool(nortp,
+            false,
+            "Excludes RTP packets from the converted RTPdump file.");
+DEFINE_bool(nortcp,
+            false,
+            "Excludes RTCP packets from the converted RTPdump file.");
+DEFINE_string(ssrc,
+              "",
+              "Store only packets with this SSRC (decimal or hex, the latter "
+              "starting with 0x).");
+
+// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
+// written to the output variable |ssrc|, and true is returned. Otherwise,
+// false is returned.
+// The empty string must be validated as true, because it is the default value
+// of the command-line flag. In this case, no value is written to the output
+// variable.
+bool ParseSsrc(std::string str, uint32_t* ssrc) {
+  // If the input string starts with 0x or 0X it indicates a hexadecimal number.
+  auto read_mode = std::dec;
+  if (str.size() > 2 &&
+      (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
+    read_mode = std::hex;
+    str = str.substr(2);
+  }
+  std::stringstream ss(str);
+  ss >> read_mode >> *ssrc;
+  return str.empty() || (!ss.fail() && ss.eof());
+}
+
+}  // namespace
+
+// This utility will convert a stored event log to the rtpdump format.
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "Tool for converting an RtcEventLog file to an RTP dump file.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name + " input.rel output.rtp\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 3) {
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+  std::string input_file = argv[1];
+  std::string output_file = argv[2];
+
+  uint32_t ssrc_filter = 0;
+  if (!FLAGS_ssrc.empty())
+    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
+        << "Flag verification has failed.";
+
+  webrtc::rtclog::EventStream event_stream;
+  if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
+    std::cerr << "Error while parsing input file: " << input_file << std::endl;
+    return -1;
+  }
+
+  rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer(
+      webrtc::test::RtpFileWriter::Create(
+          webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
+
+  if (!rtp_writer.get()) {
+    std::cerr << "Error while opening output file: " << output_file
+              << std::endl;
+    return -1;
+  }
+
+  std::cout << "Found " << event_stream.stream_size()
+            << " events in the input file." << std::endl;
+  int rtp_counter = 0, rtcp_counter = 0;
+  bool header_only = false;
+  // TODO(ivoc): This can be refactored once the packet interpretation
+  //             functions are finished.
+  for (int i = 0; i < event_stream.stream_size(); i++) {
+    const webrtc::rtclog::Event& event = event_stream.stream(i);
+    if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
+      if (event.has_timestamp_us() && event.has_rtp_packet() &&
+          event.rtp_packet().has_header() &&
+          event.rtp_packet().header().size() >= 12 &&
+          event.rtp_packet().has_packet_length() &&
+          event.rtp_packet().has_type()) {
+        const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+        if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
+          continue;
+        if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
+          continue;
+        if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
+          continue;
+        if (!FLAGS_ssrc.empty()) {
+          const uint32_t packet_ssrc =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                  reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
+                                                   8));
+          if (packet_ssrc != ssrc_filter)
+            continue;
+        }
+
+        webrtc::test::RtpPacket packet;
+        packet.length = rtp_packet.header().size();
+        if (packet.length > packet.kMaxPacketBufferSize) {
+          std::cout << "Skipping packet with size " << packet.length
+                    << ", the maximum supported size is "
+                    << packet.kMaxPacketBufferSize << std::endl;
+          continue;
+        }
+        packet.original_length = rtp_packet.packet_length();
+        if (packet.original_length > packet.length)
+          header_only = true;
+        packet.time_ms = event.timestamp_us() / 1000;
+        memcpy(packet.data, rtp_packet.header().data(), packet.length);
+        rtp_writer->WritePacket(&packet);
+        rtp_counter++;
+      } else {
+        std::cout << "Skipping malformed event." << std::endl;
+      }
+    }
+    if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
+      if (event.has_timestamp_us() && event.has_rtcp_packet() &&
+          event.rtcp_packet().has_type() &&
+          event.rtcp_packet().has_packet_data() &&
+          event.rtcp_packet().packet_data().size() > 0) {
+        const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
+        if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
+          continue;
+        if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
+          continue;
+        if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
+          continue;
+        if (!FLAGS_ssrc.empty()) {
+          const uint32_t packet_ssrc =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                  reinterpret_cast<const uint8_t*>(
+                      rtcp_packet.packet_data().data() + 4));
+          if (packet_ssrc != ssrc_filter)
+            continue;
+        }
+
+        webrtc::test::RtpPacket packet;
+        packet.length = rtcp_packet.packet_data().size();
+        if (packet.length > packet.kMaxPacketBufferSize) {
+          std::cout << "Skipping packet with size " << packet.length
+                    << ", the maximum supported size is "
+                    << packet.kMaxPacketBufferSize << std::endl;
+          continue;
+        }
+        // For RTCP packets the original_length should be set to 0 in the
+        // RTPdump format.
+        packet.original_length = 0;
+        packet.time_ms = event.timestamp_us() / 1000;
+        memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
+        rtp_writer->WritePacket(&packet);
+        rtcp_counter++;
+      } else {
+        std::cout << "Skipping malformed event." << std::endl;
+      }
+    }
+  }
+  std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
+            << " RTP packets and " << rtcp_counter << " RTCP packets to the "
+            << "output file." << std::endl;
+  return 0;
+}
diff --git a/webrtc.gyp b/webrtc.gyp
index 12b14ee..4f4f100 100644
--- a/webrtc.gyp
+++ b/webrtc.gyp
@@ -29,6 +29,17 @@
           },
         'includes': ['build/protoc.gypi'],
         },
+        {
+          'target_name': 'rtc_event_log2rtp_dump',
+          'type': 'executable',
+          'sources': ['video/rtc_event_log2rtp_dump.cc',],
+          'dependencies': [
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            'rtc_event_log',
+            'rtc_event_log_proto',
+            'test/test.gyp:rtp_test_utils'
+          ],
+        }
       ],
     }],
   ],