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Performance Analysis for VoIP System

Performance Analysis for VoIP System. Members R94922009 周宜穎 R94922020 吳鴻鑫 R94922064 張嘉輔. Outline. What is Performance Performance Bound How to analyze Performance Some Performance Analysis Exmaple. What is Performance ? [10].

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Performance Analysis for VoIP System

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  1. Performance Analysis for VoIP System Members R94922009 周宜穎 R94922020 吳鴻鑫 R94922064 張嘉輔

  2. Outline • What is Performance • Performance Bound • How to analyze Performance • Some Performance Analysis Exmaple

  3. What is Performance ? [10] • There are numerous factors that affect the performance assessments. • Human factors • Device factors • Network factors

  4. 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 ) <=> MOS rankings and User Satisfaction

  5. Human factors Voice Quality Classes

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

  7. Network factors • Network congestion • Link failures • Routing instabilities • Competing traffic • General Measruement : • Delay • Jitter • Packet loss

  8. The Performance Standard • Delay • Good (0ms-150ms) • Acceptable (150ms-300ms) • Poor (> 300ms) • Jitter • Good (0ms-20ms) • Acceptable (20ms-50ms) • Poor (> 50ms). • Loss • Good (0%-0.5%) • Acceptable (0.5%-1.5%) • Poor (> 1:5%)

  9. 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

  10. 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.

  11. 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

  12. 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

  13. 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

  14. Curve Diagram

  15. 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

  16. MOS Vs Delay

  17. MOS Vs Jitter

  18. MOS Vs Loss

  19. Normalized Each unit in the normalized scale corresponds to delay : 150ms jitter : 20ms loss : 0.5%.

  20. 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

  21. 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

  22. 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.

  23. 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 host’s UDP module to generate an ICMP port unreachable error when the datagram arrives

  24. ICMP • TCP/UDP/IP 協定若有錯誤情形發生時,會利用 Internet Control Message Protocol(ICMP)協定來送錯誤訊息 。 • 在 ICMP 的 type 中,目前約有 15 種 • The ICMP echo mechanism should be installed in host in the measurement

  25. RTT of one sent packet

  26. How to Compute? • Ti = Bi / v + Di /v + CL + C • Ti − CL =(Bi + Di) / v + C.

  27. 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,i+1 = (T i+1 − T i ) • Packet loss = packetslost / packetssent

  28. How about more complicated? • Precision timestamping • Queuing Model • Special Model for Protocol or device • Seem to Traffic Analysis!?

  29. Ex: SIP Traffic Model [11] • A model for SIP Traffic • Two Sub Model • IP Path Model • SIP Finite State Machine

  30. 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

  31. Sample Computation • Call Dropping rate pcd

  32. 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

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

  34. Jitter in Ethernet Network • IPSec produced the highest jitter values for both H.323 and SIP communications.

  35. Jitter in Wireless-LAN • IPSec-based VoIP communications generally incurred the highest jitter values.

  36. Packet Loss Rates • IPSec and PPTP increased the packet loss rate in both Ethernet and WLAN. SIP H.323

  37. Performance in Satellite Network [1] • Also provides IP-base data services • For remote region • As backup links

  38. 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

  39. Test bed configuration

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

  41. Bandwidth Efficiency 5 5

  42. Bandwidth • A single channel

  43. Link Errors Tests • Random Error Tests and burst Error Tests • BERs (bit error rates) = BD/(B+GC) • Burst length (B) • Burst density (D) • Gap length (G) • Link capacity kb/s (C)

  44. Random Error Tests

  45. Burst Error Tests

  46. Link Loading Tests • Environment • With different link loading levels • Link errors or not • Packet loss • Packet delay

  47. Tests with an Error-Free Link

  48. Tests with an Error-Free Link

  49. Tests with Errors • Combine effect of both link loading and link errors. • Error ↑,background traffic↓ link loading level↓  link loading level can’t be pre- determined

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