Performance Analysis for VoIP System

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Performance Analysis for VoIP System

• Members
• R94922009 ???
• R94922020 ???
• R94922064 ???

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Outline

• What is Performance
• Performance Bound
• How to analyze Performance
• Some Performance Analysis Exmaple

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What is Performance ? 10

• There are numerous factors that affect the
  performance assessments.
• Human factors
• Device factors
• Network factors

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Human factors -Audiovisual Quality Assessment
Metrics

• Subjective quality assessment - MOS
• Objective quality assessment
• Signal-to-Noise Ratio (SNR)
• Mean Square Error (MSE)
• Perceptual Analysis Measurement System (PAMS)
• Perceptual Evaluation of Speech Quality (PESQ)
• E -model
• E-model
• R-scale ( 0 to 100 ) ltgt MOS rankings and User
  Satisfaction

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Human factors
Voice Quality Classes
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Device factors

• Essential devices such as
• VoIP endpoints
• Gateways
• MCUs (Multipoint Control Units )
• Routers
• Firewalls
• NATs (Network Address Translators )
• Modems
• Operating System
• Processor
• memory

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Network factors

• Network congestion
• Link failures
• Routing instabilities
• Competing traffic
• General Measruement
• Delay
• Jitter
• Packet loss

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The Performance Standard

• Delay
• Good (0ms-150ms)
• Acceptable (150ms-300ms)
• Poor (gt 300ms)
• Jitter
• Good (0ms-20ms)
• Acceptable (20ms-50ms)
• Poor (gt 50ms).
• Loss
• Good (0-0.5)
• Acceptable (0.5-1.5)
• Poor (gt 15)

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E-model

• ITU-T recommendation
• Well established computational model
• Using Transmission parameters to predict the
  quality
• We can get the basic Performance Standard by
  through the model

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Basic formula for the E-model

• R-value Ro - Is - Id - Ie A
• Ro
• the basic signal-to-noise ratio based on sender
  and receiver loudness ratings and the circuit and
  room noise
• Is
• the sum of real-time or simultaneous speech
  transmission impairments,e.g. loudness levels,
  sidetone and PCM quantizing distortion
• Id
• the sum of delay impairments relative to the
  speech signal, e.g., talker echo, listener echo
  and absolute delay
• Ie
• the equipment impairment factor for special
  equipment, e.g., low bit-rate coding (determined
  subjectively for each codec and for each packet
  loss and documented in ITU-T Recommendation
  G.113)
• A
• the advantage factor adds to the total and
  improves the R-value for new services.

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Estimating the R

• R (Ro - Is) - Id - Ie A
• Ro , Is
• do not depend on network environment
• Id
• This the Argument of Delay
• Ie
• It mostly affect by codec and packet loss
• A
• Additional adjust argument ,not considered in
  general

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Estimating Id and Ie

• Id Idte Idle Idd
• Idte -Talker echo delay
• Idle - Listener echo delay
• Idd - Long delay
• Ie
• It base on codec, but packet loss affect can be
  emulated as a function

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Estimating Id and Ie

• The distortion as a function of packet loss also
  depends on whether or not PLC (Packet Loss
  Concealment)
• increases 4 units for codecs with PLC (in the R
  scale per 1 packet loss)
• 10 units for codecs without PLC

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Curve Diagram
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Test Setup

• Using 9 scenarios to test 27 possibilities
• Using NISTnet network emulator
• (http//snad.ncsl.nist.gov/itg/nistnet/)
• create the various network health scenarios

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MOS Vs Delay
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MOS Vs Jitter
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MOS Vs Loss
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Normalized
Each unit in the normalized scale corresponds to
delay 150ms jitter 20ms loss 0.5.
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The Conclusion about Performance bounds

• We show that end-user perception of audiovisual
  quality is more sensitive to the variations in
  end-to-end jitter than to variations in delay or
  loss
• We get a simple standard about the Performance to
  estimate Performance

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How to Analyze Performance

• Thinking about two topic
• Measurement
• Network Condition
• Measurement mean the analysis model that estimate
  key parameters
• Of course, it is the way to compute delay, jitter
  ,packet loss

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Two Measurement 7,8,9

• There are two methods in performance measurement
• passive measurement
• records and analyzes existing traffic.
• active measurement
• Inject sample packets into the network.

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Introduce a simple Measure

• Measurement Method in LAN
• sends sequences of UDP packets to unlikely values
  of destination port numbers (larger than 30,000)
• This causes the destination hosts UDP module to
  generate an ICMP port unreachable error when the
  datagram arrives

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ICMP

• TCP/UDP/IP ???????????,??? Internet Control
  Message Protocol(ICMP)???????? ?
• ? ICMP ? type ?,???? 15 ?
• The ICMP echo mechanism should be installed in
  host in the measurement

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RTT of one sent packet
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How to Compute?

• Ti Bi / v
• Di /v
• CL
• C
• Ti - CL
• (Bi Di) / v
• C.

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Keep estimating

• one-way delay (T i )
• T i (Ri - Si) -Di / v -CL/2 - C/2
• This calculation assumes that all delay happens
  on the sending path.
• J i,i1 (T i1 - T i )
• Packet loss packetslost / packetssent

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How about more complicated?

• Precision timestamping
• Queuing Model
• Special Model for Protocol or device
• Seem to Traffic Analysis!?

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Ex SIP Traffic Model 11

• A model for SIP Traffic
• Two Sub Model
• IP Path Model
• SIP Finite State Machine

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FSH Notation

• Q State set
• M fixed number of sessions
• C the bottleneck transmission rate( bit/s)
• R total capacity of IP Path measure in packets
  of D bits
• rtt round trip time measured in seconds
• p probability of 3xx Response
• ps successful probability of packet transmission

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Sample Computation

• Call Dropping rate pcd

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Enviroment condition for VoIP performance 4 ,
5

• The aspects about VoIP Performance Analysis
• Protocols
• H.323 v.s. SIP
• Network
• Ethernet network v.s. wireless LAN (WLAN) network
• Security for VoIP Communication
• VPN protocols PPTP v.s. IPSec

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Delay in Ethernet Network

• Both SIP and H.323 incurred higher delays in
  secure network-to-network environment.

SIP
H.323
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Jitter in Ethernet Network

• IPSec produced the highest jitter values for both
  H.323 and SIP communications.

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Jitter in Wireless-LAN

• IPSec-based VoIP communications generally
  incurred the highest jitter values.

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Packet Loss Rates

• IPSec and PPTP increased the packet loss rate in
  both Ethernet and WLAN.

SIP
H.323
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Performance in Satellite Network 1

• Also provides IP-base data services
• For remote region
• As backup links

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The purpose

• The performance under
• Delay
• Random errors , burst errors
• Link loading
• Two codecs
• 8 kb/s G.729
• 6.3/5.3 kb/s G.723.1

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Test bed configuration
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Baseline Tests

• Bandwidth and bandwidth efficiency
• Environment
• No background traffic
• No error
• Link delay set 270ms
• Run 15min with all 24 channel

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Bandwidth Efficiency

• 
  

5
5
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Bandwidth

• A single channel

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Link Errors Tests

• Random Error Tests and burst Error Tests
• BERs (bit error rates) BD/(BGC)
• Burst length (B)
• Burst density (D)
• Gap length (G)
• Link capacity kb/s (C)

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Random Error Tests
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Burst Error Tests
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Link Loading Tests

• Environment
• With different link loading levels
• Link errors or not
• Packet loss
• Packet delay

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Tests with an Error-Free Link
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Tests with an Error-Free Link
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Tests with Errors

• Combine effect of both link loading and link
  errors.
• Error ?,background traffic?
• link loading level?
• ? link loading level cant be pre-
• determined

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Impact of link failures on VoIP performance 3

• Three major causes of performance degradation
• network congestion
• link failures
• routing instabilities
• Congestion is always negligible.
• Link failures may be followed by long periods of
  routing instability.
• The goal is to study the impact of link failures
  on VoIP performance.

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Portion of the network topology

• Solid arrows primary path
• Dashed arrows alternative path used after the
  failure

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Impact of failures on data traffic

• 0634 R1, R2, R5 link to R4 is down
• 0635 R1, R2, R5 adjacency with R4 recovered
• 06360647 R4 is instable

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Impact of failures on data traffic

• 0648 R4 finally reboots
• 0659 R4 builds its first routing table
• 0717 R1, R2, R5 link to R4 is definitely up
• 0736 an alternative path is chosen

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Impact of failures on data traffic

• the failure we observed in four phases
• 0634 link is down, delay?, few packet loss
• 06360647 router is instable, same delay,
  packet loss?
• 06480704 router reboots, no delay, packet
  loss?
• 07050717 router builds routing table, delay?,
  packet loss?

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Referencs

• 1Voice over IP Service and Performance in
  Satellite Network, IEEE Communications Magazine
• 2Technique for Performance Improvement of VoIP
  Applications, IEEE MELECON 2002
• 3Impact of link failures on VoIP performance,
  ACM 1-58113-512-2/02/0005
• 4VoIP Performance Measure Using Qos Parameters
  , The Second International Conference
• 5VoIP Performance Management, Internet
  Telephony Fall 2005
• 6Comparative Analysis of Traditional Telephone
  and VoIP System
• 7VoIP Performance on differenriate service
• 8Measuring Voice Readiness of Local Area
  Networks
• 9Experimental Investigation of the Relationship
  between IP Network Performances and Speech
  Quality of VoIP
• 10Performance Measurement and Analysis of H.323
  Traffic
• 11 A Technique to Analyse Session Initiation
  Protocol Traffic

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Appendix 1

• Delay within the E-model
• Id Idte Idle Idd
• Idte -Talker echo delay
• Idle - Listener echo delay
• Idd - Long delay

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The E-model delay measures

• T mean one-way delay
• Ta absolute delay
• Tr round-trip delay
• Id can be computed by three argument

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VoIP delay estimate

• Drtcp delay estimate from RTCP packets.
• De coding and packetization delay (at least as
  large as packet size)
• Dj delay introduced by jitter buffer and
  decoder
• Ds send sides access delay
• Dr receive sides access delay

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Delay measures transform

• T Drtcp Dj De Dr
• Tr 2 Drtcp Dj De
• Ta Drtcp Dj De Dr Ds
• So the following importance is .
• How to estimate the Delay?

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Estimation of Delays

• Estimation of Ds and Dr defaulted to zero
• Estimation of De
• the length of a coded fram
• the codec lookahead
• the number of frames in the packet
• the efficiency of the coder.
• choosing best-case 20 of the frame size would
  be a reasonable estimate of encoding delay

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Estimation of Delays

• Estimation of Drtcp
• Drtcp is the round-trip delay estimate divided by
  2.
• Estimation of Dj
• This is dependent on the VoIP gateways jitter
  buffer and decoder.
• A possible equation for Dj is
• Dj min ( codec_frame_size 0.9 RTP_jitter ,
  300 )

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Appendix 2

• Qos

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Parameters of VoIP performance and improvement
techniques 2 , 6 , 7

• End-to-End Delay
• Jitter
• Frame erasure
• Out-of-order packet delay

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End-to-End Delay

• The delay from the mouth of speaker to the ear of
  listener
• Network delay
• packet processing in both end system
• packet processing in network device
• propagation delay
• Others (but leave out here)
• speech processing
• speech compression
• speech packetization

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Network delay

• fixed part
• In every network note (router) IP packets are
  delayed
• propagation delay
• transmission delay
• variable part
• the time spent in queues of the network nodes on
  the transmission path

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Reduce network delay

• fixed part
• If the network and the transmission path are
  fixed
• ?shorter IP packets
• variable part
• Some advanced queue-scheduling mechanisms
• e.g. the IETF document RFC 2598
• Using fragments time of long packets to send

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Reduce jitter

• employ a playout buffer
• playout
  time 1
• playout
  time 2

packet loss
Trade-off
additional delay
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jitter absorption

• Three main technique
• fixed playout time ?static playout time
• Adaptive adjusting of the playout time during
  silence periods
• Constantly adapting the playoit time for each
  individual packet

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Frame erasure

• the packet does not arrive in time
• is corrupted during the transmission through the
  network
• is dropped because of the network congestion
• is lost because of a network malfunction
• just arrives too late

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Reduce frame erasure

• FEC (Forward Error Correction)
• need additional bandwidth and increases delays
  because additional processing.
• Loss concealment
• be used independently or in the combination with
  FEC
• is effective only at low loss rate of a single
  frame

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Out-of-order packet delay

• occurs in the network with a complex topology
• Done in the jitter buffer
• reordering (using RTP header)
• elimination of jitter