| /* | |
| * 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 <algorithm> | |
| #include <cstdint> | |
| #include <cstdio> | |
| #include <cstdlib> | |
| #include <functional> | |
| #include <map> | |
| #include <optional> | |
| #include <string> | |
| #include "api/units/timestamp.h" | |
| #include "logging/rtc_event_log/events/logged_rtp_rtcp.h" | |
| #include "logging/rtc_event_log/rtc_event_log_parser.h" | |
| #include "rtc_base/numerics/sequence_number_unwrapper.h" | |
| #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h" | |
| namespace webrtc { | |
| 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()), | |
| 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()), | |
| 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); | |
| } | |
| std::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::Micros(timestamp)), | |
| rtp_explanation); | |
| } | |
| last_rtp_time.emplace(timestamp); | |
| } | |
| std::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()), | |
| 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()), | |
| 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(); | |
| Timestamp 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()); | |
| } | |
| } | |
| double avg_outgoing_loss = | |
| static_cast<double>(total_lost_packets) / total_expected_packets; | |
| if (avg_outgoing_loss > kMaxLossFraction) { | |
| Alert(TriageAlertType::kOutgoingHighLoss, | |
| config_.GetCallTimeSec(first_occurrence), | |
| "More than 5% of outgoing packets lost."); | |
| } | |
| } | |
| 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::ProcessAlerts( | |
| std::function<void(int, float, std::string)> f) { | |
| for (const auto& alert : triage_alerts_) { | |
| f(alert.second.count, alert.second.first_occurrence, | |
| alert.second.explanation); | |
| } | |
| } | |
| } // namespace webrtc |