rtc_tools/rtc_event_log_visualizer/alerts.cc - src - Git at Google

 /*
  *  Copyright (c) 2020 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 "rtc_tools/rtc_event_log_visualizer/alerts.h"

 #include <stdio.h>

 #include <algorithm>
 #include <limits>
 #include <map>
 #include <string>

 #include "logging/rtc_event_log/rtc_event_processor.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/strings/string_builder.h"

 namespace webrtc {

 void TriageHelper::Print(FILE* file) {
   fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
   for (const auto& alert : triage_alerts_) {
     fprintf(file, "%d %s. First occurrence at %3.3lf\n", alert.second.count,
             alert.second.explanation.c_str(), alert.second.first_occurrence);
   }
   fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
 }

 void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
                                      PacketDirection direction) {
   // With 100 packets/s (~800kbps), false positives would require 10 s without
   // data.
   constexpr int64_t kMaxSeqNumJump = 1000;
   // With a 90 kHz clock, false positives would require 10 s without data.
   constexpr int64_t kTicksPerMillisec = 90;
   constexpr int64_t kCaptureTimeGraceMs = 10000;

   std::string seq_num_explanation =
       direction == kIncomingPacket
           ? "Incoming RTP sequence number jumps more than 1000. Counter may "
             "have been reset or rewritten incorrectly in a group call."
           : "Outgoing RTP sequence number jumps more than 1000. Counter may "
             "have been reset.";
   std::string capture_time_explanation =
       direction == kIncomingPacket ? "Incoming capture time jumps more than "
                                      "10s. Clock might have been reset."
                                    : "Outgoing capture time jumps more than "
                                      "10s. Clock might have been reset.";
   TriageAlertType seq_num_alert = direction == kIncomingPacket
                                       ? TriageAlertType::kIncomingSeqNumJump
                                       : TriageAlertType::kOutgoingSeqNumJump;
   TriageAlertType capture_time_alert =
       direction == kIncomingPacket ? TriageAlertType::kIncomingCaptureTimeJump
                                    : TriageAlertType::kOutgoingCaptureTimeJump;

   const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

   // Check for gaps in sequence numbers and capture timestamps.
   for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
     if (IsRtxSsrc(parsed_log, direction, stream.ssrc)) {
       continue;
     }
     auto packets = stream.packet_view;
     if (packets.empty()) {
       continue;
     }
     SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
     int64_t last_seq_num =
         seq_num_unwrapper.Unwrap(packets[0].header.sequenceNumber);
     SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
     int64_t last_capture_time =
         capture_time_unwrapper.Unwrap(packets[0].header.timestamp);
     int64_t last_log_time_ms = packets[0].log_time_ms();
     for (const auto& packet : packets) {
       if (packet.log_time_us() > segment_end_us) {
         // Only process the first (LOG_START, LOG_END) segment.
         break;
       }

       int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
       if (std::abs(seq_num - last_seq_num) > kMaxSeqNumJump) {
         Alert(seq_num_alert, config_.GetCallTimeSec(packet.log_time_us()),
               seq_num_explanation);
       }
       last_seq_num = seq_num;

       int64_t capture_time =
           capture_time_unwrapper.Unwrap(packet.header.timestamp);
       if (std::abs(capture_time - last_capture_time) >
           kTicksPerMillisec *
               (kCaptureTimeGraceMs + packet.log_time_ms() - last_log_time_ms)) {
         Alert(capture_time_alert, config_.GetCallTimeSec(packet.log_time_us()),
               capture_time_explanation);
       }
       last_capture_time = capture_time;
     }
   }
 }

 void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
                                            PacketDirection direction) {
   constexpr int64_t kMaxRtpTransmissionGap = 500000;
   constexpr int64_t kMaxRtcpTransmissionGap = 3000000;
   std::string rtp_explanation =
       direction == kIncomingPacket
           ? "No RTP packets received for more than 500ms. This indicates a "
             "network problem. Temporary video freezes and choppy or robotic "
             "audio is unavoidable. Unnecessary BWE drops is a known issue."
           : "No RTP packets sent for more than 500 ms. This might be an issue "
             "with the pacer.";
   std::string rtcp_explanation =
       direction == kIncomingPacket
           ? "No RTCP packets received for more than 3 s. Could be a longer "
             "connection outage"
           : "No RTCP packets sent for more than 3 s. This is most likely a "
             "bug.";
   TriageAlertType rtp_alert = direction == kIncomingPacket
                                   ? TriageAlertType::kIncomingRtpGap
                                   : TriageAlertType::kOutgoingRtpGap;
   TriageAlertType rtcp_alert = direction == kIncomingPacket
                                    ? TriageAlertType::kIncomingRtcpGap
                                    : TriageAlertType::kOutgoingRtcpGap;

   const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

   // TODO(terelius): The parser could provide a list of all packets, ordered
   // by time, for each direction.
   std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
   for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
     for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
       rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
   }
   absl::optional<int64_t> last_rtp_time;
   for (const auto& kv : rtp_in_direction) {
     int64_t timestamp = kv.first;
     if (timestamp > segment_end_us) {
       // Only process the first (LOG_START, LOG_END) segment.
       break;
     }
     int64_t duration = timestamp - last_rtp_time.value_or(0);
     if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
       // No packet sent/received for more than 500 ms.
       Alert(rtp_alert, config_.GetCallTimeSec(timestamp), rtp_explanation);
     }
     last_rtp_time.emplace(timestamp);
   }

   absl::optional<int64_t> last_rtcp_time;
   if (direction == kIncomingPacket) {
     for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) {
       if (rtcp.log_time_us() > segment_end_us) {
         // Only process the first (LOG_START, LOG_END) segment.
         break;
       }
       int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
       if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
         // No feedback sent/received for more than 2000 ms.
         Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
               rtcp_explanation);
       }
       last_rtcp_time.emplace(rtcp.log_time_us());
     }
   } else {
     for (const auto& rtcp : parsed_log.outgoing_rtcp_packets()) {
       if (rtcp.log_time_us() > segment_end_us) {
         // Only process the first (LOG_START, LOG_END) segment.
         break;
       }
       int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
       if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
         // No feedback sent/received for more than 2000 ms.
         Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
               rtcp_explanation);
       }
       last_rtcp_time.emplace(rtcp.log_time_us());
     }
   }
 }

 // TODO(terelius): Notifications could possibly be generated by the same code
 // that produces the graphs. There is some code duplication that could be
 // avoided, but that might be solved anyway when we move functionality from the
 // analyzer to the parser.
 void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) {
   AnalyzeStreamGaps(parsed_log, kIncomingPacket);
   AnalyzeStreamGaps(parsed_log, kOutgoingPacket);
   AnalyzeTransmissionGaps(parsed_log, kIncomingPacket);
   AnalyzeTransmissionGaps(parsed_log, kOutgoingPacket);

   const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

   int64_t first_occurrence = parsed_log.last_timestamp();
   constexpr double kMaxLossFraction = 0.05;
   // Loss feedback
   int64_t total_lost_packets = 0;
   int64_t total_expected_packets = 0;
   for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
     if (bwe_update.log_time_us() > segment_end_us) {
       // Only process the first (LOG_START, LOG_END) segment.
       break;
     }
     int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
                            bwe_update.expected_packets;
     total_lost_packets += lost_packets;
     total_expected_packets += bwe_update.expected_packets;
     if (bwe_update.fraction_lost >= 255 * kMaxLossFraction) {
       first_occurrence = std::min(first_occurrence, bwe_update.log_time_us());
     }
   }
   double avg_outgoing_loss =
       static_cast<double>(total_lost_packets) / total_expected_packets;
   if (avg_outgoing_loss > kMaxLossFraction) {
     Alert(TriageAlertType::kOutgoingHighLoss, first_occurrence,
           "More than 5% of outgoing packets lost.");
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2020 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 "rtc_tools/rtc_event_log_visualizer/alerts.h"

	#include <stdio.h>

	#include <algorithm>
	#include <limits>
	#include <map>
	#include <string>

	#include "logging/rtc_event_log/rtc_event_processor.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/format_macros.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/numerics/sequence_number_util.h"
	#include "rtc_base/strings/string_builder.h"

	namespace webrtc {

	void TriageHelper::Print(FILE* file) {
	fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
	for (const auto& alert : triage_alerts_) {
	fprintf(file, "%d %s. First occurrence at %3.3lf\n", alert.second.count,
	alert.second.explanation.c_str(), alert.second.first_occurrence);
	}
	fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
	}

	void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
	PacketDirection direction) {
	// With 100 packets/s (~800kbps), false positives would require 10 s without
	// data.
	constexpr int64_t kMaxSeqNumJump = 1000;
	// With a 90 kHz clock, false positives would require 10 s without data.
	constexpr int64_t kTicksPerMillisec = 90;
	constexpr int64_t kCaptureTimeGraceMs = 10000;

	std::string seq_num_explanation =
	direction == kIncomingPacket
	? "Incoming RTP sequence number jumps more than 1000. Counter may "
	"have been reset or rewritten incorrectly in a group call."
	: "Outgoing RTP sequence number jumps more than 1000. Counter may "
	"have been reset.";
	std::string capture_time_explanation =
	direction == kIncomingPacket ? "Incoming capture time jumps more than "
	"10s. Clock might have been reset."
	: "Outgoing capture time jumps more than "
	"10s. Clock might have been reset.";
	TriageAlertType seq_num_alert = direction == kIncomingPacket
	? TriageAlertType::kIncomingSeqNumJump
	: TriageAlertType::kOutgoingSeqNumJump;
	TriageAlertType capture_time_alert =
	direction == kIncomingPacket ? TriageAlertType::kIncomingCaptureTimeJump
	: TriageAlertType::kOutgoingCaptureTimeJump;

	const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

	// Check for gaps in sequence numbers and capture timestamps.
	for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
	if (IsRtxSsrc(parsed_log, direction, stream.ssrc)) {
	continue;
	}
	auto packets = stream.packet_view;
	if (packets.empty()) {
	continue;
	}
	SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
	int64_t last_seq_num =
	seq_num_unwrapper.Unwrap(packets[0].header.sequenceNumber);
	SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
	int64_t last_capture_time =
	capture_time_unwrapper.Unwrap(packets[0].header.timestamp);
	int64_t last_log_time_ms = packets[0].log_time_ms();
	for (const auto& packet : packets) {
	if (packet.log_time_us() > segment_end_us) {
	// Only process the first (LOG_START, LOG_END) segment.
	break;
	}

	int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
	if (std::abs(seq_num - last_seq_num) > kMaxSeqNumJump) {
	Alert(seq_num_alert, config_.GetCallTimeSec(packet.log_time_us()),
	seq_num_explanation);
	}
	last_seq_num = seq_num;

	int64_t capture_time =
	capture_time_unwrapper.Unwrap(packet.header.timestamp);
	if (std::abs(capture_time - last_capture_time) >
	kTicksPerMillisec *
	(kCaptureTimeGraceMs + packet.log_time_ms() - last_log_time_ms)) {
	Alert(capture_time_alert, config_.GetCallTimeSec(packet.log_time_us()),
	capture_time_explanation);
	}
	last_capture_time = capture_time;
	}
	}
	}

	void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
	PacketDirection direction) {
	constexpr int64_t kMaxRtpTransmissionGap = 500000;
	constexpr int64_t kMaxRtcpTransmissionGap = 3000000;
	std::string rtp_explanation =
	direction == kIncomingPacket
	? "No RTP packets received for more than 500ms. This indicates a "
	"network problem. Temporary video freezes and choppy or robotic "
	"audio is unavoidable. Unnecessary BWE drops is a known issue."
	: "No RTP packets sent for more than 500 ms. This might be an issue "
	"with the pacer.";
	std::string rtcp_explanation =
	direction == kIncomingPacket
	? "No RTCP packets received for more than 3 s. Could be a longer "
	"connection outage"
	: "No RTCP packets sent for more than 3 s. This is most likely a "
	"bug.";
	TriageAlertType rtp_alert = direction == kIncomingPacket
	? TriageAlertType::kIncomingRtpGap
	: TriageAlertType::kOutgoingRtpGap;
	TriageAlertType rtcp_alert = direction == kIncomingPacket
	? TriageAlertType::kIncomingRtcpGap
	: TriageAlertType::kOutgoingRtcpGap;

	const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

	// TODO(terelius): The parser could provide a list of all packets, ordered
	// by time, for each direction.
	std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
	for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
	for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
	rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
	}
	absl::optional<int64_t> last_rtp_time;
	for (const auto& kv : rtp_in_direction) {
	int64_t timestamp = kv.first;
	if (timestamp > segment_end_us) {
	// Only process the first (LOG_START, LOG_END) segment.
	break;
	}
	int64_t duration = timestamp - last_rtp_time.value_or(0);
	if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
	// No packet sent/received for more than 500 ms.
	Alert(rtp_alert, config_.GetCallTimeSec(timestamp), rtp_explanation);
	}
	last_rtp_time.emplace(timestamp);
	}

	absl::optional<int64_t> last_rtcp_time;
	if (direction == kIncomingPacket) {
	for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) {
	if (rtcp.log_time_us() > segment_end_us) {
	// Only process the first (LOG_START, LOG_END) segment.
	break;
	}
	int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
	if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
	// No feedback sent/received for more than 2000 ms.
	Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
	rtcp_explanation);
	}
	last_rtcp_time.emplace(rtcp.log_time_us());
	}
	} else {
	for (const auto& rtcp : parsed_log.outgoing_rtcp_packets()) {
	if (rtcp.log_time_us() > segment_end_us) {
	// Only process the first (LOG_START, LOG_END) segment.
	break;
	}
	int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
	if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
	// No feedback sent/received for more than 2000 ms.
	Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
	rtcp_explanation);
	}
	last_rtcp_time.emplace(rtcp.log_time_us());
	}
	}
	}

	// TODO(terelius): Notifications could possibly be generated by the same code
	// that produces the graphs. There is some code duplication that could be
	// avoided, but that might be solved anyway when we move functionality from the
	// analyzer to the parser.
	void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) {
	AnalyzeStreamGaps(parsed_log, kIncomingPacket);
	AnalyzeStreamGaps(parsed_log, kOutgoingPacket);
	AnalyzeTransmissionGaps(parsed_log, kIncomingPacket);
	AnalyzeTransmissionGaps(parsed_log, kOutgoingPacket);

	const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();

	int64_t first_occurrence = parsed_log.last_timestamp();
	constexpr double kMaxLossFraction = 0.05;
	// Loss feedback
	int64_t total_lost_packets = 0;
	int64_t total_expected_packets = 0;
	for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
	if (bwe_update.log_time_us() > segment_end_us) {
	// Only process the first (LOG_START, LOG_END) segment.
	break;
	}
	int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
	bwe_update.expected_packets;
	total_lost_packets += lost_packets;
	total_expected_packets += bwe_update.expected_packets;
	if (bwe_update.fraction_lost >= 255 * kMaxLossFraction) {
	first_occurrence = std::min(first_occurrence, bwe_update.log_time_us());
	}
	}
	double avg_outgoing_loss =
	static_cast<double>(total_lost_packets) / total_expected_packets;
	if (avg_outgoing_loss > kMaxLossFraction) {
	Alert(TriageAlertType::kOutgoingHighLoss, first_occurrence,
	"More than 5% of outgoing packets lost.");
	}
	}

	} // namespace webrtc