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PART2: VOIP AND CRITICAL PARAMETERS FOR A VOIP DEPLOYMENT Voice Performance Measurement and related technologies 1

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Page 1: PART2: VOIP AND CRITICAL PARAMETERS FOR A VOIP DEPLOYMENT Voice Performance Measurement and related technologies 1

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PART2: VOIP AND CRITICAL PARAMETERS FOR A VOIP DEPLOYMENT

Voice Performance Measurement and related

technologies

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VoIPMean Opinion Scores (MOS)Impairment/Calculated Planning Impairment

Factor (ICPIF)Network Elements in the Voice Path

Passive Voice Performance Measurement Active Voice Performance Measurement

Cisco CallManager (CCM)Calculating voice jitter

Outline

Page 3: PART2: VOIP AND CRITICAL PARAMETERS FOR A VOIP DEPLOYMENT Voice Performance Measurement and related technologies 1

https://www.fcc.gov/encyclopedia/voice-over-internet-protocol-voip

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VoIP

Voice over Internet Protocol (VoIP), is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line.

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VoIP

Some VoIP services may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers..

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VoIP

Also, while some VoIP services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter

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VoIP connection

VoIP connects:Phone to PhoneComputer to ComputerPhone to computer

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http://www.ontelecomuk.com/voip.html

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VoIP transport protocol

VoIP uses RTP (real-time transport protocol) which runs on top of the User Datagram Protocol (UDP)

Because VoIP does not require reliability.So the transmitted packet may suffer from

different Impairment such as: Delay Jitter Packet lost

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Mean Opinion Scores (MOS)

The dilemma of measuring the quality of transmitted speech is that it is subjective to the listener. In addition, each VoIP transmission codec delivers a different level of quality. A common benchmark to determine voice quality is MOS.

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With MOS, a wide range of listeners have judged the quality of voice samples on a scale of 1 (bad quality) to 5 (excellent quality).

Score Quality Description of Quality Impairment

5 Excellent Imperceptible

4 Good Just perceptible, but not annoying

3 Fair Perceptible and slightly annoying

2 Poor Annoying but not objectionable

1 Bad Very annoying and objectionable

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MOS-CQE

As the MOS ratings for codecs and other transmission impairments are known, an estimated MOS can be computed and displayed based on measured impairments. The ITU-T calls this estimated value Mean Opinion Score–Conversational Quality, Estimated (MOS-CQE) to distinguish it from subjective MOS values.

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Calculating MOS

Originally, the MOS was meant to represent the arithmetic mean average of all the individual quality assessments given by people who listened to a test phone call and ranked the quality of that cal

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Calculating MOS artificially

Today, human participation is no longer required to determine the quality of the audio stream. Modern VoIP quality assessment tools employ artificial software models to calculate the MOS.

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MOS limitation

The MOS is highly subjective. One should not make decisions on a VoIP system based on the MOS alone. Other measurable parameters should be analyzed such as network delay, packet loss, jitter, and so on. As an alternative to the MOS, a different, less subjective rating has been introduced

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R-Factor

R-Factor is an alternative method of assessing call quality. Scaling from 0 to 120 as opposed to the limited scale of 1 to 5 makes R-Factor a somewhat more precise tool for measuring voice quality.

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R-Factor is calculated by evaluating user perceptions as well as the objective factors that affect the overall quality of a VoIP system, accounting for the Network R-factor and the User R-factor separately.

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The following table demonstrates the effect of the MOS and R-Factor on the perceived call quality.

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Some users believe R-Factor to be a more objective measure of the quality of a VoIP system than MOS. Still, a network analyzer should be able to calculate both scores and produce the two assessments for better judgment of the call quality.

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Impairment/Calculated Planning Impairment Factor (ICPIF)

ICPIF attempts to quantify the impairments to voice quality that are encountered in the network.

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Impairment/Calculated Planning Impairment Factor (ICPIF)

ICPIF is calculated by the following formula:ICPIF = Io + Iq + Idte + Idd + Ie – Awhere:• Io— Impairment caused by nonoptimal loudness

rating• Iq— PCM quantizing distortion impairment• Idte— Talker echo impairment• Idd— One-way delay impairment• Ie— Equipment impairment• A— An Advantage or expectation factor that

compensates for the fact that users may accept quality degradation, such as with mobile services

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Upper Limit for ICPIF Speech Communication Quality

5 Very good

10 Good

20 Adequate

30 Limiting case

45 Exceptional limiting case

55 Customers likely to react strongly (complaints, change of network

operator)

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Passive Voice Performance Measurement

Active Voice Performance Measurement

Network Elements in the Voice Path

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Passive Voice Performance Measurement

Cisco voice gateways calculate the ICPIF factor

If this value exceeds a predefined ICPIF threshold, an SNMP notification is generated.

The call durations must be at least 10 seconds for the gateway to calculate the ICPIF value for the call.

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Active Voice Performance Measurement

Cisco IOS IP SLA uses synthetic traffic to measure performance between multiple network locations or across multiple network paths.

It simulates VoIP codecs and collects network performance information, including response time, one-way latency, jitter, packet loss, and voice quality scoring.

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Cisco CallManager (CCM)

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/30266-ts-ccm-301.html

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Cisco CallManager (CCM)

Cisco CallManager is an IP-based PBX that controls the call processing of a VoIP network. CCM is a central component in a Cisco Communication Network (CCN) system

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CCM distribution

A CCN comprises multiple regions, with each region consisting of several CallManager groups with multiple CallManagers.

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CCM main function

CCM establishes voice calls and gathers call detail information in a VoIP environment. It generates records for each call placed to and from IP phones, conferences bridges, and PSTN gateways.

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http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/30266-ts-ccm-301.html

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Call records types

Two different types of call records are produced:

Call Detail Records (CDR)Call Management Records (CMR)

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CDR

Call Detail Records (CDR) store call connection information, such as the called number, the date and time the call was initiated, the time it connected, and the termination time. In addition, CDRs include call control and routing information.

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CMR

Call Management Records (CMR) store information about the call's audio quality, such as bytes and packets sent or dropped, jitter, and latency. CMRs are also called diagnostic records.

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Generating CDR

CCM generates a CDR when: A call is initiated or terminated orIf significant changes occur to an active call,

such as transferring, redirecting, splitting, or joining a call.

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Generating CMR

When diagnostics are enabled at the CCM, a CMR is stored for each call, separately for each IP phone involved or each MGCP gateway

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Discovering voice quality

Voice quality trends can be discovered by inspecting the CDR's corresponding CMRs. The two records are linked by the GlobalCallID_callManagerId and GlobalCallID_Called fields in the CDR and CMR

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http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/30266-ts-ccm-301.html

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Calculating voice jitter

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Calculating voice jitter

To measure Jitter, we take the difference between samples, then divide by the number of samples (minus 1).

Jitter=difference between samples/(the number of samples-1)

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Example

Here's an example. We have collected 5 samples with the following latencies: 136, 184, 115, 148, 125 (in that order). The average latency is 142 - (add them, divide by 5). The 'Jitter' is calculated by taking the difference between samples.136 to 184, diff = 48184 to 115, diff = 69115 to 148, diff = 33148 to 125, diff = 23 (Notice how we have only 4

differences for 5 samples). The total difference is 173 - so the jitter is 173 / 4, or 43.25.

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Abbreviations

Meaning

MOS-CQE Mean Opinion Score–Conversational Quality, Estimated

RTP real-time transport protocol

CCN Cisco Communication Network

CCM Cisco CallManager

CDR Call Detail Records

CMR Call Management Records

PSTN public switched telephone network

PBX private branch exchange