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Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy

Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy. Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy. Outline. Problems affecting VoIP performance Tools for Measuring and Diagnosing Problems

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Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy

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  1. Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy Fault and Performance Management for Next Generation IP Communication Alan Clark, Telchemy

  2. Outline • Problems affecting VoIP performance • Tools for Measuring and Diagnosing Problems • Protocols for Reporting QoS • Performance Management Architecture • What to ask for/ integrate?

  3. Enterprise VoIP Deployment IP Phone IP Phones IP VPN Branch Office Teleworker Gateway IP Phone

  4. VoIP Deployment - Issues ROUTE FLAPPING, LINK FAIL IP Phone IP Phones IP VPN CODEC DISTORTION ECHO Gateway LAN CONGESTION, DUPLEX MISMATCH, LONG CABLES…. ACCESS LINK CONGESTION IP Phone

  5. Call Quality Problems • Packet Loss • Jitter (Packet Delay Variation) • Codecs and PLC • Delay (Latency) • Echo • Signal Level • Noise Level

  6. Packet Loss and Jitter Jitter Buffer Codec IP Network Distorted Speech Packets lost in network Packets discarded due to jitter

  7. Routers, Loss and Jitter Processing delay Queuing delay Serialization delay Queuing delay Input queue Output queue Arriving packets Prioritize/ Route Packet loss due to buffer Overflow or RED Voice packet delayed by one or more data packets

  8. Queuing Delays Added delay due to wait for data packets to be sent = Jitter

  9. Jitter Average jitter level (PPDV) = 4.5mS Peak jitter level = 60mS

  10. WiFi can also cause jitter

  11. Effects of Jitter • Low levels of jitter absorbed by jitter buffer • High levels of jitter • lead to packets being discarded • cause adaptive jitter buffer to grow - increasing delay but reducing discards • If packets are discarded by the jitter buffer as they arrive too late they are regarded as “discarded” • If packets arrive extremely late they are regarded as “lost” hence sometimes “lost” packets actually did arrive

  12. Packet Loss Average packet loss rate = 2.1% Peak packet loss = 30%

  13. Packet Loss is bursty • Packet loss (and packet discard) tends to occur in sparse bursts - say 20-30% in density and one second or so in length • Terminology • Consecutive burst • Sparse burst • Burst of Loss vs Loss/Discard

  14. Example Packet Loss Distribution Consecutive loss 20 percent burst density (sparse burst)

  15. Loss and Discard • Loss is often associated with periods of high congestion • Jitter is due to congestion (usually) and leads to packet discard • Hence Loss and Discard often coincide • Other factors can apply - e.g. duplex mismatch, link failures etc.

  16. Example Loss/Discard Distribution

  17. Leads To Time Varying Call Quality High jitter/ loss/ discard

  18. Packet Loss Concealment • Mitigates impact of packet loss/ discard by replacing lost speech segments • Very effective for isolated lost packets, less effective for bursty loss/discard • But isn’t loss/discard bursty? • Need to be able to deal with 10-20-30% loss!!! Estimated by PLC

  19. Effectiveness of PLC Codec distortion Impact of loss/ discard and PLC

  20. Call Quality Problems • Packet Loss • Jitter (Packet Delay Variation) • Codecs and PLC • Delay (Latency) • Echo • Signal Level • Noise Level

  21. Effect of Delay on Conversational Quality

  22. Causes of Delay Accumulate and encode External delay IP UDP TCP RTP CODEC Echo Control Network delay Jitter buffer, decode and playout IP UDP TCP RTP CODEC Echo Control

  23. Cause of Echo Gateway IP Echo Canceller Acoustic Echo Line Echo Round trip delay - typically 50mS+ Additional delay introduced by VoIP makes existing echo problems more obvious Also - “convergence” echo

  24. Echo problems • Echo with very low delay sounds like “sidetone” • Echo with some delay makes the line sound hollow • Echo with over 50mS delay sounds like…. Echo • Echo Return Loss • 55dB or above is good • 25dB or below is bad

  25. Call Quality Problems • Packet Loss • Jitter (Packet Delay Variation) • Codecs and PLC • Delay (Latency) • Echo • Signal Level • Noise Level

  26. Signal Level Problems Amplitude Clipping occurs -- speech sounds loud and “buzzy” 0 dBm0 -36 dBm0 Temporal Clipping occurs with VAD or Echo Suppressors -- gaps in speech, start/end of words missing

  27. Noise • Noise can be due to • Low signal level • Equipment/ encoding (e.g. quantization noise) • External local loops • Environmental (room) noise • From a service provider perspective - how to distinguish between • room noise (not my problem) • Network/equipment/circuit noise (is my problem)

  28. Measuring VoIP performance Analog signal based VoIP Specific VQmon ITU G.107 ITU P.862 (PESQ) Active Test - Measure test calls VQmon ITU P.VTQ ITU P.563 Passive Test - Measure live calls

  29. 4 2 3 2 “Gold Standard” - ACR Test • Speech material • Phonetically balanced speech samples 8-10 seconds in length • Test designed to eliminate bias (e.g. presentation order different for each listener) • Known files included as anchors (e.g. MNRU) • Listening conditions • Panel of listeners • Controlled conditions (quiet environment with known level of background noise)

  30. Example ACR test results • Extract from an ITU subjective test • Mean Opinion Score (MOS) was 2.4 • 1=Unacceptable • 2=Poor • 3=Fair • 4=Good • 5=Excellent

  31. Packet based approaches Test Call VoIP Test System VoIP Test System IP Measure call Live Call VQmon, G.107. P.VTQ VoIP End System VoIP End System IP Passive Test Passive Test

  32. Packet based approaches • ITU G.107 R = Ro - Is - Ie - Id + A • Really a network planning tool • Missing many essential monitoring features • VQmon • ITU G.107 + ETSI TS 101 329-5 Annex E +……. • Proprietary but widely used (Superset of G.107 & P.VTQ) • ITU P.VTQ • Available late 2005, very limited functionality

  33. Extended E Model - VQmon 4 State Markov Model Gather detailed packet loss info in real time Arriving packets Loss/ Discard events Discarded Jitter buffer Signal level Noise level Echo level CODEC Call Quality Scores Diagnostic Data Metrics Calculation

  34. Modeling transient effects Ie(burst) Measured Call quality User Reported Call quality Ie(VQmon) Ie(gap) 10 15 20 25 30 35 Time (seconds)

  35. VQmon - computational model Burst loss rate Perceptual model Calculate R-LQ MOS-LQ Ie mapping Gap loss rate ETSI TS101 329-5 Recency model Calculate Ro, Is Signal level Noise level ITU-T G.107 Calculate R-CQ MOS-CQ Calculate Id Echo Delay

  36. Accuracy: Non-bursty conditions

  37. Accuracy: Bursty conditions • G.107 • Well established model for network planning • No way to represent jitter • Few codec models • Inaccurate for bursty loss • Conversational Quality only • VQmon • Extended G.107 • Transient impairment model • Wide range of codec models • Narrow & Wideband • Jitter Buffer Emulator • Listening and Conversational Quality VQmon E Model Comparison of VQmon and E Model for severely time varying conditions

  38. Signal based approaches P.862 Tester Test Call VoIP End System VoIP End System IP P.862 is an Active Test Approach VoIP End System VoIP End System IP P.563 Tester P.563 is a Passive Test Approach

  39. ITU P.862 - Active testing Tested segment of connection IP PESQ Audio files Time align FFT… Compare PESQ Score FFT…

  40. ITU P.862 - Active testing • Send speech file • Compare received file with original using FFT • Takes typically 50-100 MIPS per call • MOS-like score in the range -0.5 to 4.5 • Widely used within the industry Results for G.729A codec for a set of speech files (i.e. for each packet loss rate the only thing changed is the speech source file)

  41. ITU P.563 - Passive monitoring • Analyses received speech file (single ended) • Produces a MOS score • Correlates well with MOS when averaged over many calls • Requires 100MIPS per call Comparison of P.563 estimated MOS scores with actual ACR test scores. Each point is average per file ACR MOS with 16 listeners compared to P.563 score

  42. Performance Monitoring - Passive Test Embedded Monitoring Function RTCP XR SIP QoS Report

  43. SLA Monitoring - Active Test Test call Active Test Functions

  44. Active or Passive Testing? • Active testing • works for pre-deployment testing and on-demand troubleshooting • But!!!! • IP problems are transient • Passive monitoring • Monitors every call made - but needs a call to monitor • Captures information on transient problems • Provides data for post-analysis • Therefore - you need both

  45. VoIP Performance Management Framework Network Management System Call Server and CDR database Signaling Based QoS Reporting SNMP Reporting Network Probe, Analyzer or Router VQ VoIP Gateway VQ VoIP Endpoint VQ RTP stream (possibly encrypted) Embedded Monitoring Embedded Monitoring Media Path Reporting (RTCP XR)

  46. VoIP Performance Management Framework • Embedded monitoring function in IP phones, residential gateways…. • Close to the user • Least cost + widest coverage • Protocol support developed • RTCP XR (RFC3611), SIP, MGCP, H.323, Megaco • Draft SNMP MIB • Works in encrypted environments • Already being deployed by equipment vendors

  47. The role of RTCP XR RTCP XR (RFC3611) • Provides a useful set of metrics for VoIP performance monitoring and diagnosis • Supports both real time monitoring and post-analysis • Extracts signal level, noise level and echo level from DSP software in the endpoint • Exchanges info on endpoint delay and echo to allow remote endpoint to assess echo impact • Provides midstream probes/ analyzers access to analog metrics if secure RTP is used • Goes through firewalls………

  48. Loss Rate Discard Rate Burst Density Gap Density Burst Duration (mS) Gap Duration (mS) Round Trip Delay (mS) End System Delay (mS) Signal level RERL Noise Level Gmin R Factor Ext R MOS-LQ MOS-CQ Rx Config - Jitter Buffer Nominal Jitter Buffer Max Jitter Buffer Abs Max RFC3611 - RTCP XR

  49. SIP Service Quality Reporting Event PUBLISH sip:collector@example.com SIP/2.0 Via: SIP/2.0/UDP pc22.example.com;branch=z9hG4bK3343d7 ……… Content-Type: application/rtcpxr Content-Length: ... VQSessionReport LocalMetrics: TimeStamps=START:10012004.18.23.43 STOP:10012004.18.26.02 SessionDesc=PT:0 PD:G.711 SR:8000 FD:20 FPP:2 PLC:3 SSUP:on CallID=1890463548@alice.uac.chicago.com ……… Signal=SL:2 NL:10 RERL:14 QualityEst=RLQ:90 RCQ:85 EXTR:90 MOSLQ:3.4 MOSCQ:3.3 QoEEstAlg:VQMonv2.1 DialogID:38419823470834;to-tag=8472761;from-tag=9123dh311

  50. RTCP XR MIB History table Session table Basic parameters Alerting Call quality metrics

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