fault and performance management for next generation ip ...€¦ · voip developers conference -...
TRANSCRIPT
Fault and Performance Management for
Next Generation IP Communication
Alan Clark, Telchemy
Fault and Performance Management for
Next Generation IP Communication
Alan Clark, Telchemy
VoIP Developers Conference - 2005 2
Outline
• Problems affecting VoIP performance• Tools for Measuring and Diagnosing Problems• Protocols for Reporting QoS• Performance Management Architecture• What to ask for/ integrate?
VoIP Developers Conference - 2005 3
Enterprise VoIP Deployment
Branch Office
IP Phone
IP VPN
IP Phone
Teleworker
IP Phones
Gateway
VoIP Developers Conference - 2005 4
VoIP Deployment - IssuesIP Phone
IP VPN
IP Phone
IP Phones
GatewayECHO
ACCESSLINKCONGESTION
LAN CONGESTION,DUPLEX MISMATCH,LONG CABLES….
ROUTEFLAPPING,LINK FAIL
CODECDISTORTION
VoIP Developers Conference - 2005 5
Call Quality Problems
• Packet Loss• Jitter (Packet Delay Variation)• Codecs and PLC• Delay (Latency)• Echo• Signal Level• Noise Level
VoIP Developers Conference - 2005 6
Packet Loss and Jitter
CodecIPNetwork
JitterBuffer
Packets lostin network
Packets discardeddue to jitter
DistortedSpeech
VoIP Developers Conference - 2005 7
Routers, Loss and Jitter
Arrivingpackets
Outputqueue
Prioritize/Route
Voice packet delayedby one or more datapackets
Queuing delay
Serialization delay
Packet loss due to bufferOverflow or RED
Inputqueue
Queuing delay
Processing delay
VoIP Developers Conference - 2005 8
Queuing Delays
0
25
50
75
100
125
150
175
200
0 500 1000 1500 2000
Transmission speed (kbits/s)
Ma
x d
ela
y (
mS
)
1 x 1500 byte MTU
2 x 1500 byte MTU
3 x 1500 byte MTU
Added delay due towait for data packetsto be sent = Jitter
VoIP Developers Conference - 2005 9
Jitter
50
75
100
125
150
0 0.5 1 1.5 2
Time (Seconds)
De
lay (
mS
)
Average jitter level (PPDV) = 4.5mSPeak jitter level = 60mS
VoIP Developers Conference - 2005 10
WiFi can also cause jitter
0
50
100
150
200
250
300
0 25 50 75 100
125
150
175
200
225
250
275
300
325
350
375
400
425
450
Time
Dela
y (
mS)
& R
SSI
RSSIDelay
VoIP Developers Conference - 2005 11
Effects of Jitter
• Low levels of jitter absorbed by jitter buffer• High levels of jitter
o lead to packets being discardedo 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
VoIP Developers Conference - 2005 12
Packet Loss
0
10
20
30
40
50
30 35 40 45 50 55 60 65 70
Time (seconds)
500m
S A
vge P
ack
et Loss
Rate Average packet loss rate = 2.1%
Peak packet loss = 30%
VoIP Developers Conference - 2005 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
• Terminologyo Consecutive bursto Sparse bursto Burst of Loss vs Loss/Discard
VoIP Developers Conference - 2005 14
0
50
100
150
200
0 100 200 300 400 500Burst length (packets)
Bu
rst
we
igh
t (p
ack
ets
)Example Packet Loss Distribution
20 percent burst density (sparse burst)
Con
secu
tive
loss
VoIP Developers Conference - 2005 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.
VoIP Developers Conference - 2005 16
Example Loss/Discard Distribution
0
50
100
150
200
0 100 200 300 400 500Burst length (packets)
Burs
t w
eig
ht
(pack
ets
)
VoIP Developers Conference - 2005 17
Leads To Time Varying Call Quality
12345
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18
Time
MO
S
0100200300400500
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18
Ba
nd
wid
th (
kb
it/
s)
Voice
Data
High jitter/ loss/ discard
VoIP Developers Conference - 2005 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
VoIP Developers Conference - 2005 19
Effectiveness of PLC
1
2
3
4
5
0 5 10 15 20
Packet Loss/Discard Rate
AC
R M
OS
G.711 no PLCG.711 PLCG.729A
Codecdistortion Impact of
loss/ discard and PLC
VoIP Developers Conference - 2005 20
Call Quality Problems
• Packet Loss• Jitter (Packet Delay Variation)• Codecs and PLC• Delay (Latency)• Echo• Signal Level• Noise Level
VoIP Developers Conference - 2005 21
Effect of Delay on Conversational Quality
1
2
3
4
5
0 100 200 300 400 500 600
Round trip delay (milliseconds)
MO
S S
core
55dB Echo Return Loss
35dB Echo Return Loss
VoIP Developers Conference - 2005 22
Causes of Delay
CODEC Echo Control
RTP
IPUDPTCP
CODEC Echo Control
RTP
IPUDPTCP
External delayAccumulate and encode
Network delay Jitter buffer, decode and playout
VoIP Developers Conference - 2005 23
Cause of Echo
IP
EchoCanceller
Gateway
LineEchoRound trip delay - typically 50mS+
Additional delay introduced by VoIP makes existing echo problems more obvious
Also - “convergence” echo
AcousticEcho
VoIP Developers Conference - 2005 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 o 55dB or above is goodo 25dB or below is bad
VoIP Developers Conference - 2005 25
Call Quality Problems
• Packet Loss• Jitter (Packet Delay Variation)• Codecs and PLC• Delay (Latency)• Echo• Signal Level• Noise Level
VoIP Developers Conference - 2005 26
Signal Level Problems
Temporal Clipping occurs with VAD or Echo Suppressors -- gaps in speech, start/end of words missing
Amplitude Clipping occurs -- speech sounds loud and “buzzy”
0 dBm0
-36 dBm0
VoIP Developers Conference - 2005 27
Noise
• Noise can be due too Low signal levelo Equipment/ encoding (e.g. quantization noise)o External local loopso Environmental (room) noise
• From a service provider perspective - how to distinguish between o Room noise (not my problem)o Network/equipment/circuit noise (is my problem)
VoIP Developers Conference - 2005 28
Measuring VoIP performance
VQmon
ITU G.107ITU P.862 (PESQ)
VQmon
ITU P.VTQITU P.563
Active Test- Measure test calls
Passive Test- Measure live calls
VoIP SpecificAnalog signal based
VoIP Developers Conference - 2005 29
“Gold Standard” - ACR Test
• Speech materialo Phonetically balanced speech samples 8-10 seconds in lengtho Test designed to eliminate bias (e.g. presentation order different for each
listener)o Known files included as anchors (e.g. MNRU)
• Listening conditionso Panel of listenerso Controlled conditions (quiet environment with known level of background
noise)
23 2
4
VoIP Developers Conference - 2005 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
0
10
20
30
40
50
Votes
1 2 3 4 5
Opinion Score
VoIP Developers Conference - 2005 31
Packet based approaches
VoIPTest
System
VoIPTest
SystemIP
VoIPEnd
System
VoIPEnd
SystemIP
PassiveTest
PassiveTest
Measurecall
Test Call
Live Call VQmon,G.107.P.VTQ
VoIP Developers Conference - 2005 32
Packet based approaches
• ITU G.107 R = Ro - Is - Ie - Id + Ao Really a network planning toolo Missing many essential monitoring features
• VQmono ITU G.107 + ETSI TS 101 329-5 Annex E +…….o Proprietary but widely used (Superset of G.107 & P.VTQ)
• ITU P.VTQ o Available late 2005, very limited functionality
VoIP Developers Conference - 2005 33
Extended E Model - VQmonArrivingpackets
Discarded
CODEC
Jitterbuffer
Loss/ Discardevents
MetricsCalculation
4 State Markov ModelGather detailedpacket loss infoin real time
Signal levelNoise levelEcho level
Call Quality ScoresDiagnostic Data
VoIP Developers Conference - 2005 34
Modeling transient effects
10 15 20 25 30 35Time (seconds)
MeasuredCall quality
User ReportedCall quality
Ie(gap)
Ie(burst)
Ie(VQmon)
VoIP Developers Conference - 2005 35
VQmon - computational modelBurst lossrate
Gap lossrate
Ie mapping
Perceptual model
CalculateR-LQMOS-LQ
CalculateRo, Is
Signal levelNoise level
CalculateId
EchoDelay
CalculateR-CQMOS-CQ
Recencymodel
ETSI TS101 329-5
ITU-T G.107
VoIP Developers Conference - 2005 36
Accuracy: Non-bursty conditions
Comparison of VQmon vs ACR MOS - ILBC 15.2k
1
1.5
2
2.5
3
3.5
4
4.5
5
0 5 10 15 20Packet Loss Rate (%)
MO
S S
co
re
ACR MOS
VQmon MOS-LQ
Comparison of VQmon vs PESQ - ILBC 15.2k
1
1.5
2
2.5
3
3.5
4
0 5 10 15 20 25 30Packet Loss Rate (%)
PE
SQ
Sco
re
PESQ
VQmon MOS-PQ
VoIP Developers Conference - 2005 37
1.5
2
2.5
3
3.5
4
1.5 2 2.5 3 3.5 4ACR MOS
Est
imate
d M
OS
Accuracy: Bursty conditions
• G.107o Well established model for
network planningo No way to represent jittero Few codec modelso Inaccurate for bursty losso Conversational Quality only
• VQmono Extended G.107o Transient impairment modelo Wide range of codec modelso Narrow & Widebando Jitter Buffer Emulatoro Listening and Conversational
Quality
VQmon
E Model
Comparison of VQmon and E Modelfor severely time varying conditions
VoIP Developers Conference - 2005 38
Signal based approaches
VoIPEnd
System
VoIPEnd
SystemIP
VoIPEnd
System
VoIPEnd
SystemIP
P.862TesterTest Call
P.563Tester
P.862 is an Active Test Approach
P.563 is a Passive Test Approach
VoIP Developers Conference - 2005 39
ITU P.862 - Active testing
IP
Timealign
Audiofiles
FFT…
FFT…
ComparePESQScore
Tested segment of connection
PESQ
VoIP Developers Conference - 2005 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
1
1.5
2
2.5
3
3.5
4
0 5 10 15 20 25 30 35 40Packet Loss Rate
PE
SQ
Sco
res
Results for G.729A codec for a set ofspeech files (i.e. for each packet lossrate the only thing changed is the speechsource file)
VoIP Developers Conference - 2005 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 call1.00
2.00
3.00
4.00
5.00
1 2 3 4 5
P563 Score
AC
R M
OS
Comparison of P.563 estimated MOS scores with actual ACR test scores.Each point is average per file ACR MOS with 16listeners compared to P.563 score
VoIP Developers Conference - 2005 42
Performance Monitoring - Passive Test
RTCP XR
SIP QoSReport
EmbeddedMonitoringFunction
VoIP Developers Conference - 2005 43
SLA Monitoring - Active Test
Active Test Functions
Test call
VoIP Developers Conference - 2005 44
Active or Passive Testing?
• Active testing o works for pre-deployment testing and on-demand troubleshooting
• But!!!!o IP problems are transient
• Passive monitoring o Monitors every call made - but needs a call to monitoro Captures information on transient problemso Provides data for post-analysis
• Therefore - you need both
VoIP Developers Conference - 2005 45
VoIP Performance Management Framework
Media Path Reporting(RTCP XR)
Call Server andCDR database
VoIPEndpoint
VoIPGateway
SNMPReporting
NetworkManagementSystem
Signaling Based QoS Reporting
Embedded Monitoring
Network Probe,Analyzer orRouter
VQVQ
Embedded Monitoring
VQ
RTP stream (possibly encrypted)
VoIP Developers Conference - 2005 46
VoIP Performance Management Framework
• Embedded monitoring function in IP phones, residential gateways….
o Close to the usero Least cost + widest coverage
• Protocol support developedo RTCP XR (RFC3611), SIP, MGCP, H.323, Megacoo Draft SNMP MIB
• Works in encrypted environments• Already being deployed by equipment vendors
VoIP Developers Conference - 2005 47
The role of RTCP XR
RTCP XR (RFC3611)
1. Provides a useful set of metrics for VoIP performance monitoringand diagnosis
2. Supports both real time monitoring and post-analysis
3. Extracts signal level, noise level and echo level from DSP software in the endpoint
4. Exchanges info on endpoint delay and echo to allow remote endpoint to assess echo impact
5. Provides midstream probes/ analyzers access to analog metrics ifsecure RTP is used
6. Goes through firewalls………
VoIP Developers Conference - 2005 48
RFC3611 - RTCP XR
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
VoIP Developers Conference - 2005 49
SIP Service Quality Reporting Event
PUBLISH sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP pc22.example.com;branch=z9hG4bK3343d7………
Content-Type: application/rtcpxrContent-Length: ...
VQSessionReportLocalMetrics:TimeStamps=START:10012004.18.23.43 STOP:10012004.18.26.02SessionDesc=PT:0 PD:G.711 SR:8000 FD:20 FPP:2 PLC:3 SSUP:[email protected]………
Signal=SL:2 NL:10 RERL:14QualityEst=RLQ:90 RCQ:85 EXTR:90 MOSLQ:3.4 MOSCQ:3.3
QoEEstAlg:VQMonv2.1DialogID:38419823470834;to-tag=8472761;from-tag=9123dh311
VoIP Developers Conference - 2005 50
RTCP XR MIB
Session table
Basic parameters
Call quality metrics
History table
Alerting
VoIP Developers Conference - 2005 51
Passive Monitoring Framework
Branch Office
IP Phone
IP VPN
IP Phone
Teleworker
VQ
IP Phones
Gateway
NMS
VQ
VQ
VQ
VQ
VQ
VQ
VQ
VQ
VQ
VQ
VQ
RTCP XR
SIP QoS ReportSNMP
VoIP Developers Conference - 2005 52
What to Implement/ Ask For
• Embedded monitoring functionality in IP Phones and Gateways (e.g. VQmon)
• RTCP XR for mid-call data exchange between endpoints
• SIP Service Quality Events for reporting end of call quality
• RTCP XR MIB for SNMP support
VoIP Developers Conference - 2005 53
Summary
• Problems affecting VoIP performance• Tools for Measuring and Diagnosing Problems• Protocols for Reporting QoS• Performance Management Architecture• What to ask for/ integrate?