| /* |
| * 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/logging.h" |
| #include "rtc_base/numerics/sequence_number_util.h" |
| #include "rtc_base/strings/string_builder.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); |
| } |
| 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::Micros(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()), |
| 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 |