diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 18cfe4d165..407767b236 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -126,4 +126,24 @@ if (rtc_enable_protobuf) {
       }
     }
   }
+  if (rtc_include_tests) {
+    rtc_executable("rtc_event_log2text") {
+      testonly = true
+      sources = [
+        "rtc_event_log/rtc_event_log2text.cc",
+      ]
+      deps = [
+        ":rtc_event_log_api",
+        ":rtc_event_log_impl",
+        ":rtc_event_log_parser",
+        "../base:rtc_base_approved",
+        "../modules/rtp_rtcp:rtp_rtcp",
+        "//third_party/gflags",
+      ]
+      if (!build_with_chromium && is_clang) {
+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+      }
+    }
+  }
 }
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
new file mode 100644
index 0000000000..32d5ae5f7a
--- /dev/null
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
@@ -0,0 +1,425 @@
+/*
+ *  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 <iostream>
+#include <sstream>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/call/call.h"
+#include "webrtc/common_types.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace {
+
+DEFINE_bool(noincoming, false, "Excludes incoming packets.");
+DEFINE_bool(nooutgoing, false, "Excludes outgoing packets.");
+// TODO(terelius): Note that the media type doesn't work with outgoing packets.
+DEFINE_bool(noaudio, false, "Excludes audio packets.");
+// TODO(terelius): Note that the media type doesn't work with outgoing packets.
+DEFINE_bool(novideo, false, "Excludes video packets.");
+// TODO(terelius): Note that the media type doesn't work with outgoing packets.
+DEFINE_bool(nodata, false, "Excludes data packets.");
+DEFINE_bool(nortp, false, "Excludes RTP packets.");
+DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
+// TODO(terelius): Allow a list of SSRCs.
+DEFINE_string(ssrc,
+              "",
+              "Print only packets with this SSRC (decimal or hex, the latter "
+              "starting with 0x).");
+
+static uint32_t filtered_ssrc = 0;
+
+// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
+// written to the static global variable |filtered_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) {
+  // 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 >> filtered_ssrc;
+  return str.empty() || (!ss.fail() && ss.eof());
+}
+
+bool ExcludePacket(webrtc::PacketDirection direction,
+                   webrtc::MediaType media_type,
+                   uint32_t packet_ssrc) {
+  if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
+    return true;
+  if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
+    return true;
+  if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+    return true;
+  if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+    return true;
+  if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+    return true;
+  if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
+    return true;
+  return false;
+}
+
+const char* StreamInfo(webrtc::PacketDirection direction,
+                       webrtc::MediaType media_type) {
+  if (direction == webrtc::kOutgoingPacket) {
+    if (media_type == webrtc::MediaType::AUDIO)
+      return "(out,audio)";
+    else if (media_type == webrtc::MediaType::VIDEO)
+      return "(out,video)";
+    else if (media_type == webrtc::MediaType::DATA)
+      return "(out,data)";
+    else
+      return "(out)";
+  }
+  if (direction == webrtc::kIncomingPacket) {
+    if (media_type == webrtc::MediaType::AUDIO)
+      return "(in,audio)";
+    else if (media_type == webrtc::MediaType::VIDEO)
+      return "(in,video)";
+    else if (media_type == webrtc::MediaType::DATA)
+      return "(in,data)";
+    else
+      return "(in)";
+  }
+  return "(unknown)";
+}
+
+void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
+                       uint64_t log_timestamp,
+                       webrtc::PacketDirection direction,
+                       webrtc::MediaType media_type) {
+  webrtc::rtcp::SenderReport sr;
+  if (!sr.Parse(rtcp_block))
+    return;
+  if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
+    return;
+  std::cout << log_timestamp << "\t"
+            << "RTCP_SR" << StreamInfo(direction, media_type)
+            << "\tSSRC=" << sr.sender_ssrc()
+            << "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
+}
+
+void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
+                         uint64_t log_timestamp,
+                         webrtc::PacketDirection direction,
+                         webrtc::MediaType media_type) {
+  webrtc::rtcp::ReceiverReport rr;
+  if (!rr.Parse(rtcp_block))
+    return;
+  if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
+    return;
+  std::cout << log_timestamp << "\t"
+            << "RTCP_RR" << StreamInfo(direction, media_type)
+            << "\tSSRC=" << rr.sender_ssrc() << std::endl;
+}
+
+void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
+             uint64_t log_timestamp,
+             webrtc::PacketDirection direction,
+             webrtc::MediaType media_type) {
+  webrtc::rtcp::ExtendedReports xr;
+  if (!xr.Parse(rtcp_block))
+    return;
+  if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
+    return;
+  std::cout << log_timestamp << "\t"
+            << "RTCP_XR" << StreamInfo(direction, media_type)
+            << "\tSSRC=" << xr.sender_ssrc() << std::endl;
+}
+
+void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
+               uint64_t log_timestamp,
+               webrtc::PacketDirection direction,
+               webrtc::MediaType media_type) {
+  std::cout << log_timestamp << "\t"
+            << "RTCP_SDES" << StreamInfo(direction, media_type) << std::endl;
+  RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
+}
+
+void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
+              uint64_t log_timestamp,
+              webrtc::PacketDirection direction,
+              webrtc::MediaType media_type) {
+  webrtc::rtcp::Bye bye;
+  if (!bye.Parse(rtcp_block))
+    return;
+  if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
+    return;
+  std::cout << log_timestamp << "\t"
+            << "RTCP_BYE" << StreamInfo(direction, media_type)
+            << "\tSSRC=" << bye.sender_ssrc() << std::endl;
+}
+
+void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
+                      uint64_t log_timestamp,
+                      webrtc::PacketDirection direction,
+                      webrtc::MediaType media_type) {
+  std::cout << "Rtp feedback found";
+  switch (rtcp_block.fmt()) {
+    case webrtc::rtcp::Nack::kFeedbackMessageType: {
+      webrtc::rtcp::Nack nack;
+      if (!nack.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_NACK" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << nack.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
+      webrtc::rtcp::Tmmbr tmmbr;
+      if (!tmmbr.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_TMMBR" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << tmmbr.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
+      webrtc::rtcp::Tmmbn tmmbn;
+      if (!tmmbn.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_TMMBN" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << tmmbn.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
+      webrtc::rtcp::RapidResyncRequest sr_req;
+      if (!sr_req.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_SRREQ" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << sr_req.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
+      webrtc::rtcp::TransportFeedback transport_feedback;
+      if (!transport_feedback.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type,
+                        transport_feedback.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_NEWFB" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << transport_feedback.sender_ssrc() << std::endl;
+      break;
+    }
+    default:
+      RTC_DCHECK(false);
+      break;
+  }
+}
+
+void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
+                     uint64_t log_timestamp,
+                     webrtc::PacketDirection direction,
+                     webrtc::MediaType media_type) {
+  switch (rtcp_block.fmt()) {
+    case webrtc::rtcp::Pli::kFeedbackMessageType: {
+      webrtc::rtcp::Pli pli;
+      if (!pli.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_PLI" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << pli.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Sli::kFeedbackMessageType: {
+      webrtc::rtcp::Sli sli;
+      if (!sli.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, sli.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_SLI" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << sli.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Rpsi::kFeedbackMessageType: {
+      webrtc::rtcp::Rpsi rpsi;
+      if (!rpsi.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, rpsi.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_RPSI" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << rpsi.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Fir::kFeedbackMessageType: {
+      webrtc::rtcp::Fir fir;
+      if (!fir.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_FIR" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << fir.sender_ssrc() << std::endl;
+      break;
+    }
+    case webrtc::rtcp::Remb::kFeedbackMessageType: {
+      webrtc::rtcp::Remb remb;
+      if (!remb.Parse(rtcp_block))
+        return;
+      if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
+        return;
+      std::cout << log_timestamp << "\t"
+                << "RTCP_REMB" << StreamInfo(direction, media_type)
+                << "\tSSRC=" << remb.sender_ssrc() << std::endl;
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+}  // namespace
+
+// This utility will print basic information about each packet to stdout.
+// Note that parser will assert if the protobuf event is missing some required
+// fields and we attempt to access them. We don't handle this at the moment.
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "Tool for printing packet information from an RtcEventLog as text.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name + " input.rel\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 2) {
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+  std::string input_file = argv[1];
+
+  if (!FLAGS_ssrc.empty())
+    RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed.";
+
+  webrtc::ParsedRtcEventLog parsed_stream;
+  if (!parsed_stream.ParseFile(input_file)) {
+    std::cerr << "Error while parsing input file: " << input_file << std::endl;
+    return -1;
+  }
+
+  for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
+    if (!FLAGS_nortp &&
+        parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
+      size_t header_length;
+      size_t total_length;
+      uint8_t header[IP_PACKET_SIZE];
+      webrtc::PacketDirection direction;
+      webrtc::MediaType media_type;
+      parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
+                                 &header_length, &total_length);
+
+      // Parse header to get SSRC and RTP time.
+      webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
+      webrtc::RTPHeader parsed_header;
+      rtp_parser.Parse(&parsed_header);
+
+      if (ExcludePacket(direction, media_type, parsed_header.ssrc))
+        continue;
+
+      std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
+                << StreamInfo(direction, media_type)
+                << "\tSSRC=" << parsed_header.ssrc
+                << "\ttimestamp=" << parsed_header.timestamp << std::endl;
+    }
+    if (!FLAGS_nortcp &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+      size_t length;
+      uint8_t packet[IP_PACKET_SIZE];
+      webrtc::PacketDirection direction;
+      webrtc::MediaType media_type;
+      parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
+
+      webrtc::rtcp::CommonHeader rtcp_block;
+      const uint8_t* packet_end = packet + length;
+      for (const uint8_t* next_block = packet; next_block != packet_end;
+           next_block = rtcp_block.NextPacket()) {
+        ptrdiff_t remaining_blocks_size = packet_end - next_block;
+        RTC_DCHECK_GT(remaining_blocks_size, 0);
+        if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
+          break;
+        }
+
+        uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
+        switch (rtcp_block.type()) {
+          case webrtc::rtcp::SenderReport::kPacketType:
+            PrintSenderReport(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          case webrtc::rtcp::ReceiverReport::kPacketType:
+            PrintReceiverReport(rtcp_block, log_timestamp, direction,
+                                media_type);
+            break;
+          case webrtc::rtcp::Sdes::kPacketType:
+            PrintSdes(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          case webrtc::rtcp::ExtendedReports::kPacketType:
+            PrintXr(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          case webrtc::rtcp::Bye::kPacketType:
+            PrintBye(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          case webrtc::rtcp::Rtpfb::kPacketType:
+            PrintRtpFeedback(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          case webrtc::rtcp::Psfb::kPacketType:
+            PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
+            break;
+          default:
+            break;
+        }
+      }
+    }
+  }
+  return 0;
+}