IP Telephony: Your First Course

Presented byGary AudinDelphi, Inc.

delphi-inc@att.net

Tutorial Outline1. What Are IPT and VoIP?2. Justifying the Move to VoIP/IPT3. Standards for VoIP/IPT4. Constructing a VoIP Network, the

Piece Parts5. How Does a VoIP Call Work?6. IP Trunking Concepts and Operation7. Carrying VoIP over Broadband8. Preparing the LAN and WAN for VoIP9. Measuring Voice Quality (MOS)10. The Future of VoIP/IPT

Further Education1. “VoIP and IP Telephony” - 2 day seminar(Material from this seminar is used in this tutorial)2. “Deploying VOIP and IP Telephony in the Enterprise” - 2 day seminar (This seminar covers advanced material and is the follow on to the previous seminar)

“Ask the Experts” is being held at the BCR Training exhibit booth on Monday and Tuesday from 4 to 6 PM

Information Resources

www.voiploop.com - weekly BLOG on communications subjects

www.webtorials.com and www.bcr.com -15 articles on VoIP and IP Telephony

Section 1

What Are IP Telephony and VoIP?

What is VoIP/IP Telephony?Wired and Wireless

Analog Voice Digital Voice (64 Kbps)CONVERSION

Compression to 16 Kbps or less – Optional

Silence Suppression (Voice Activity Detection) – Optional

Transfer through routers and LAN NIC usingtheir protocols

Compression Concepts

CODEC Only

AnalogVoice

Digital64,000 bps PCMor 32,000 bps

ADPCMstandards

CODEC and compressorusing software or DSP

Digital 4,000 to16,000 bps

various standards(G.7xx) andproprietarymethods

AnalogVoice

DSP = Digital Signal Processor

CODer

DECoder

Talk

Silence

Send packets

No packetsreceived

ANALOG DIGITAL

Conversations are usually half duplex, i.e., alternating speakers.

Silence simulation required (comfort noise).

There are silence gaps between:— Words— Sentences

Total silence is about 70%.

Silence Suppression(Voice Activity Detection - VAD)

Voice over IP Placement (1)

-Voice Quality,-Delays

Tie Line, FX, OPX, Trunk, Local Loop

IP TrunkingSIP Trunking

+ Low Risk, Fast Return +

Transmission Substitution

Voice over IP Placement (2)

Switching Alternatives

PBX, ACD, CO,Centrex, Call Center

IP PBX

Reliability ?,+Features-

+ Small Size, Integration +

Voice over IP Placement (3)

-New Cabling,-Limited Distance,Power to Phone ?

+ Integration +

Telephone Replacement

IP Phone, PC Phone (SoftPhone)

Converged Networks (part 1)

Analog

Network

Voice / Video

CODEC G.711

Compress G.723/28/29

Create voice datagram (RTP)

Add headers (TCP/UDP/IP, etc.)

Digital Encounter delay, jitter,errors, loss, out-of-sequence

Data LAN Fax ENTRY

Converged Networks (part 2)

Analog

Digital Network Encounter delay, jitter, errors,loss, out-of-sequence

CODEC G.711

Decompress, Buffer delays

Process headers

Resequence packets

Voice / Video

Loss compensation

Data LAN Fax EXIT

Network Characteristics

Voice / Video--------Short delayConstant delayNo loss allowedNo retransmissionDirect pass-throughNo overhead

Data / FaxLow error rateReasonable delayVariable delayLoss due to congestionRetransmissionUses protocolHas overhead

Traditional OSI Layered Approach

APPSSERVER

TCP / UDP

IPNET

MANAGERS

OSILAYER DATA VOICE

PBXAS

MANAGER

CODEC

SS#7

ISUP

Q.931

Q.921

PHYSICAL

APISERVER

NETWORK

7

6

5

4

3

2

1

New Model from Data View(Voice Is An Application)

OSILayer

API

ServerManager

VoIPManager

IPNetworkManager

APPSand

Server

TCP / UDP

IPLink

Physical

7

6

5

4

3

2

1

Voice CODECSignaling

H.323SIP

MGCP

Voice over IP Networks

THEINTERNET

ANINTRANET

VPNMIP

INHERITED ENVIRONMENT

CONTROLLEDENVIRONMENT

OUTSOURCE

— No Goals

— No Guarantees

— Performance Objectives

— Designed for Performance

MIP = Managed IP NetworkVPN = Virtual Private Network

CHEAP MODERATE EXPENSIVE

Section 2

Justifying the Move to VoIP/IPT

Why Do It?Reduce transmission cost (10% to 30%)Fill unused data and ISP bandwidthCombine voice call center and web access into a single serviceSupport voice portals over IPProvide a multi-media / mail access deviceReduce moves, adds, changes, labor and costsSupport new applicationsSupport new applications

Coverage Where?Campus (as part of LAN based PBX)

Local calls (intra LATA)

Toll calls (intra LATA)

Long distance calls (inter LATA) (harder as prices decrease)

International calls (definitely)

Tie line (private line; easy to do)

FX (foreign exchange)

OPX (off-premise extension)

Hoot-and-Holler / Ringdown

Junkyard circuits

Cannot be justified alone

CTI / Web Integration

Gatekeeper

PBX / ACD

Gateway

CustomerDatabase

DataFirewallCaller Content

Switch

Agents

Data

Voice

Data

DigitalVoice

Gateway

The Gatekeeper can replace the PBX/ACD.

CTIVoice

Voice= API= Hard Link

VoiceFirewall

Economic Value of Compression

Conclusions:Little money is saved by heavy compressionPerformance degrades with compression

2.5¢

1¢

2¢

64K

Circuit-Switched 64Kbps

8K Packet Voice64K Packet Voice

8K 16K 32KVoice Digitization in bps

Percent of Respondents Reporting the Benefit

0 10 20 30 40 50 60 70 80

Easier move, add or change process allows employees to move workstations more often

Faster moves, adds or changes

New office opening completed quicker

IT staff time saved as end users can use telephony features without help

Reduced need for IT staff to travel

Less telephone tag for all employees

Improved remote office employee productivity

Improved telecommuter productivity

Source: BCR

Application Residence

N e t w o r k

A P I ?

IP Phone

Apps Server Call Server

SoftPhone

Acceptance of VoIPCarrier Acceptance

Trunking (Yes)Switching (Some Class 4)VoCable and VoDSL (Limited)

Enterprise CautionIP Telephony (Small sites and pilots)Trunking (Yes)Call/Web Center Integration (Early use)

Section 3

Standards for VoIP/IPT

StandardsH.323 signaling (ITU-T) SIP and MGCP signaling (IETF)Non-guaranteed bandwidthUses packets to carry voiceRequires call server for switching controlIndependent of compression standardG.7xx compression standards (ITU)RTP for voice transmissionRTCP for QoS monitoring

VoIP Signaling StandardsSIP

SGCP (Telcordia)

MGCP

MEGACOH.248

IPDC(Level 3)

H.323 H.GCP

MEGACO (H.248) = ITU + IETF Standard

H.323 v.1 through v.5Network – Non-guaranteed Bandwidth Packet Network

Audio – G.711, G.722, G.728, G.723, G.729Video – H.261, H.263

Call Control / Security – H.225.0Open / Close Channels – H.245

Telephony Features – H.450Multipoint (Conferencing) – H.323

Data – T.120Interface – TCP/IP, Frame Relay, Ethernet

H.323 Protocol StackCall Establishment and Control

Presentation

Addressing Audio CODECG.711 or G.729

DTMF Addressing

RAS(H.225)

DNS RTP / RTCP H.245 Q.931(H.225)

DNS

Unreliable Transport(UDP)

(Speech)

Reliable Transport(TCP)

(Signaling)

Network (IP)

Link

PhysicalSource: IMTC VoIP Forum

SIP Signaling Stack

RTCPSIP RTP DNS DHCP

I P

N e t w o r k T e c h n o l o g i e s

UDPTCP

SDP G.7xx

H.323 vs. SIPFACTOR H.323 SIPDesign Complex

(736-page spec) Simplex

(128-page spec)Number of Elements 100's 37

Messages Based on ASN.1 HTTP and RTSP

Call Setup Multiple Requests Single Request

Extensibility Non Standard Use SessionDescription Protocol

Large-NumberDomains

Designed for LAN Designed for IPNetworks

Server Processing "Hold" state for allcalls

Pass Through

Conferencing Limited Open to all sizes

Feedback H.245 does not workin Multicast

RTCP

Firewall Support Difficult EasierIn teroperab i l i t y Not Common Bec oming Common

Complex(736-page spec)

Simplex(128-page spec)

Use SessionDescription Protocol

Large-NumberDomains

Designed for IPNetworks

"Hold" state for allcalls

H.245 does not workin Multicast

Protocol Usage

H.323SIP Call Server

MGCP

Proprietary

C a ll S e rve r

H.323SIP-T

MGCPH.248

Gateway

SIP

Streaming vs. Conferencing

* Try Real Player Radio vs. Broadcast Radio

STREAMING VOICE / VIDEO — RTSP

Constant Rate

5- to 10-second buffer

Unpredictable Rate

CONFERENCING VOICE / VIDEO — RTP

NearConstant Rate

1/20- to 1/5-secondBuffer

1/20- to 1/5-secondBuffer

NearConstant Rate

Reference DocumentsREFERENCE DOCUMENT DESCRIPTIONITU G.711 (Universal) Pulse Code Modulation at 64KbpsITU G.723.1 (IP) Dual Rate Speech Coder for Multimedia Communications

Transmitting at 5.3 and 6.3 Kbit/sITU G.723.1, Annex A (IP) Silence Compression SchemeITU G.723.1, Annex B (IP) Alternative Specification Based on Floating Point ArithmeticITU G.723.1, Annex C (IP) Scaleable Channel Coding Scheme for Wireless ApplicationsITU G.726 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation

(ADPCM)ITU G.727 (AAL2) 5-, 4-, 3-, and 2-bits Sample Embedded Adaptive Differential Pulse

Code ModulationITU G.728 (VIDEO) Coding of Speech at 16 kbit/s Using Low Delay Code Excited

Linear PredictionITU G.729/ITU G.729, Annex A(FRAME RELAY)

Coding of Speech at 8 kbit/s using Conjugate Structure-AlgebraicCode Excited Linear Predictive (CS-ACEP) Coding

ITU G.764 Voice Packetization – Packetized Voice Protocols

REFERENCE DOCUMENT DESCRIPTIONITU G.711 (Universal) Pulse Code Modulation at 64KbpsITU G.723.1 (IP) Dual Rate Speech Coder for Multimedia Communications

Transmitting at 5.3 and 6.3 Kbit/sITU G.723.1, Annex A (IP) Silence Compression SchemeITU G.723.1, Annex B (IP) Alternative Specification Based on Floating Point ArithmeticITU G.723.1, Annex C (IP) Scaleable Channel Coding Scheme for Wireless ApplicationsITU G.726 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation

(ADPCM)ITU G.727 (AAL2) 5-, 4-, 3-, and 2-bits Sample Embedded Adaptive Differential Pulse

Code ModulationITU G.728 (VIDEO) Coding of Speech at 16 kbit/s Using Low Delay Code Excited

Linear PredictionITU G.729/ITU G.729, Annex A(FRAME RELAY)

Coding of Speech at 8 kbit/s using Conjugate Structure-AlgebraicCode Excited Linear Predictive (CS-ACEP) Coding

ITU G.764 Voice Packetization – Packetized Voice Protocols

Pulse Code Modulation(Not Compressed)

8000 samples per secondSample rate = 2 x bandwidth of 4000 HzSample = 8 bitsTotal = 8 x 8000 + 64,000 bpsSupports modems and faxes up through 56 Kbps without compression

G.711 — Today’s CODEC

Code Excited Linear Prediction (CELP)

10 ms

80 PCM Samples

G.729 and G.729/ASamples adjusted to normal (standardized) shape.Shape is compared to shape table with values.Selected shape and loudness level transmitted.10ms block = 160 bitsCompressed to 20 bits used in VoIP frame relay.

CELP Shape Code Book (example)

1

2

4

3

5

6

7

8

Va lues / Shapes

VoIP Packet

Voice24-240

IP20+

UDP8

RTP12

Ethernet, Frame Relay, PPP

Trailer Headers

Headers + Trailer = Overhead

Bandwidth Tax

G.711

G.729

Overhead Costs

VoIP T1 Line

Bandwidth ConsumptionG.711 (64Kbps) uses 80 to

110Kbps

G.729 (8Kbps) uses 24 to 28 Kbps

Section 4

Constructing the VoIP Network, the Piece Parts

IP PBX Components

TrunkGateway

Access / MediaGateway

IP LAN / WAN

IP PhoneSoftphone

Call Server/DNS

Register VoIP Device

Server

DHCP

Get IP Address

What about VLANs, scope options

Network Identity Services

DHCPDynamic Host

Configuration Protocol

ADDRESSING

Software and configurations for IP phones can change

Provides firmware and configs to IP devices

Trivial File Transfer Protocol

FILE DELIVERY

TFTP

Essential for TIME!!!

Time clock

Network Time Protocol

TIME

NTP

Automatically assigns IP addresses to phones

Needed for screen phones, SIP

The network “Phone Book”(yahoo.com = 66.94.234.13)

DNSDomain Name System

NAMING

IP / Ethernet PhonesApplication support over LAN with APPs serverPower over LAN

OptionalProprietary (Cisco)Standard (802.3af)

Power sourcesAC outletLAN SwitchPower Bar

IP SoftphonePC

Sound Quality dependson Data Apps running

SeparateProcessor

DSPCODECor MICRO

PROCESSOR (UDP / IP)DATAGRAM

LAN

LANNIC

ROUTER

Windows XP issuperior to earlier

OS by up to 60 ms

Gateway or IP Phone Elements

Voice

CODEC Signaling

G.7xx

RTP

TCPUDP

IP

Compressor

DatagramAssembler

H.323

SIP

T h e IS P / In te rn e t

Gateway Connections

ETHERNET

IP

UDP

TCP

RTP

Voice Codec Comp.64K 8K

Fax

Demodulator

9.6/14.4KTelemetry

Alarms

TDD

IVR

Modem

64K

IP-enabled PBXAnalog/Digital Phones

Analog/Digital Trunks

DigitalPBX

IP Trunks

IPPhones Gateway Gateway

Circuit switchingUses IP peripheralsGateway functions can be IP line / trunk cards65% of PBXs installed can be IP enabledLimited solution

Gatekeeper

IP PBX Configuration

Ethernet LAN hub/switch

Call Server and MCU

IP phonesEther phones

TelephoneGateway(access)

Carrier Access Gateway(Trunk)

LEC

IXCTrunks

100m

100m

1000mServer = Virtual PBXGateway = Legacy access

Converged IP PBX (TDM / LAN)

LAN

IP Phones

Proprietary controlCentralized / distributed controlTDM and IP signalingIntegrated gatewayCentralized / distributed gateways

Analog / DigitalPhones

DigitalCircuit Switch

Call ServerIP Network

Analog / DigitalTrunks

Section 5

How Does a VoIP Call Work?

The Telephone Network

AnalogLocal Loop

PBX

Digital

CODECAnalog (Digital)Central Office

Local Loop

LEC 64,000 bps Digi tal

POP

POP

CODEC

Central Office

64,000 bps Digi tal

64,000 bps Digi tal

H.323 and SIP Signaling Paths

TrunkGateway

PSTN

Analog

T1/E1

PRI

Access / MediaGateway

IP LAN / WANFAX

Phone

Modem

Sof tphone

Call Server/DNS/DHCP

IP Phone

SIP Protocol SessionSIP Phone SIP Phone

Cal l Server Cal l Server

Media Sess ion

O K O K O K

ByeByeBye

ACK ACK ACK

InviteInvite

Invite

TryingTrying

O K

O K

O K

RingingRinging Ringing

RTP Speech Paths

Access / MediaGateway

IP LAN / WANFAX

Phone

Modem

Call Server/DNS/DHCP

SoftphoneIP Phone

T runkGateway

PSTN

Analog

T1/E1

PRI

N + X Server Reliability

GatewaysIP Phones

Gatekeeper(Call Server)

X1+

NETWORK

Gatekeeper(Call Server)

N2N1 X2

Section 6

IP Trunking Concepts and Operation

TDM Trunking Operation

PBX

A A A AB B B BC C C C

Constant / Short Delay

A

B

C

A

B

C

Pre-assigned Capacity

Guaranteed BandwidthTDM/Digital Telephony

PBX

Router Packet Relay

DestinationOriginator

P1P2P3P4P5P6

Message += + + + +P6 P1P2P3P4P5

Switch = path setup Router = no path setup

IP Trunking Operation

1

2

3

1 1 12 23

Packets1

2

3

Variable / Longer Delay

IP Statistical Packet Multiplexing

UnassignedCapacity

Router Router

Switch vs. Router

A switch selects a path per call and uses a connection number in each packet to identify the destinationA router selects a path per packet and includes the source and destination addresses in each packetVoIP is routed, not switched

SIP Trunking

IP trunking to a carrierSupports VoIP traffic Can support IP trafficLess costly connection when compared to traditional trunking

Section 7

VoIP over Broadband Access

VoIP Service is Coming to a Desk near You

Consumer VoIP service is competing for your communications dollars.There are more than 1,100 vendors out there.Skype has about 46% of the US usersVonage is spending millions to expand their market.

Internet Telephony Service Providers

Inter-carrier (digital access)Inter-/Intra-corporate (digital access)Domestic (phone-to-phone-to-PC)International (phone-to-phone-to-PC)

Traditional ISP as ITSP (AOL)

Long distance carriers as ITSP (AT&T, MCI, Sprint)

Internet Telephony Providers

(Vonage, VoicePulse, Broadvox, Skype )

Subscribing to VoIP ServiceVoIP service providers use all or part of the Internet or ISP to carry the calls.The service can be peer-to-peer or server based. Is this another period like the dot coms?Should consumers look at these services?Will the VoIP service providers deliver business quality services?

Would you want to talk to a customer overthese services?…

User ComplaintsCall control does not work with fax and modem useProblems with firewall configurationPoor customer serviceCan make outgoing calls, no incoming callsFAXs going to voice mailCan not disable voice mail WEB interface could be betterTelephone adapters are not all the same in performanceSlow connection times and dropped calls

Business ConsiderationsThird party application/feature vendors offer service through the VoIP provider. Who is responsible when they do not work or are incorrectly charged?Will the security and privacy of the calls meet the business requirements?Will the provider comply with regulations such as CALEA, SOX, HIPAA…?The VoIP provider may use proprietary protocols that could pose problems to the enterprises firewalls and intranet.

Network NeutralityVoIP service is unregulated. ISPs are not required to carry this traffic.The FCC has a consent decree against Madison River Communications opening their broadband service to VoIP traffic.Broadband vendors do not want any government regulation on this issue. They want voluntary network neutrality. Deutsche Telecom, Saudi Arabia and wireless carrier Clearwire block VoIP trafficChina blocks Skype calls

Good for the Teleworker?Broadband service must be at worker locationWho pays for the VoIP service?Home firewall and router may block callsSoftware firewalls may add latency to callTelephone adapter is portable but requires some setup each time it is used away from the primary location, especially for 911Softphone software is not secure

Section 8

Preparing the LAN and WAN for VoIP

Ethernet IP-PBX ConfigurationCall

ProcessingServer

100 mTrunks

100 m

100 m

100 mGateways

Router

Cables and ClosetsComponent Legacy Phone IP Phone

Legacy Phone Closet — MDF Use As Is Use for Trunking Only — IDF Use As Is AbandonLAN Closet Not Used ExpandedCabling 1 Pair Voice Grade 2 to 4 Pair

Category 3 to 5Power From Switch From LAN Switch,

Gateway, Power Bar110/120 volts

Air Conditioning Switch Room,not MDF or IDF

LAN Closet

Distances 1000 to 2000 meters 100 metersClosets Same as Before Multiple per Floor

IP Phone to LAN Switch

1

2

3

LANSwitch

Questions:• Power• QoS• Failure• Cost

Power over Ethernet (PoE)Uses two pairs unassigned, pins 4,5,7,8May use data pairs, pins 1,2,3,6802.3af uses positive polarity powerCisco uses negative polarity powerMany products support all methods delivering 12.6 watts at end point, non standard delivers 39 wattsBIG QUESTION:

Are your cables ready for PoE?

PoE: A Power UtilityEnvironmental

ControlsIP

Phones

WLANAccessPoints

BluetoothDevices

Security DevicesVideo

Servers

StageLighting

WebCam/Security

PoE

Gateway / Data Network Problems

SoftPhone (XP) • Delay• Jitter• No errors• Loss

Entry Gateway • Delay• No jitter?• No errors• No loss?

IP Network Creates• Extra Delay• Jitter• Errors• Loss• Out-of-sequence

Packets

IPNetwork Gateway

Entry ExitGateway

ExitGateway/Phone• Adds delay• Corrects jitter

but adds delay• Corrects loss• Corrects out-of-

sequence butadds delay

(Does NOT include gateways or phones)

FACTOR PSTN VoIPTOLERANCE

VPNor

INTRANETINTERNET

ERRORS Very low andignored

IgnoredNo retransmission

LowCorrected byretransmission

LowCorrected byretransmission

ONE-WAYDELAY

1 - 30ms 50 - 100ms 20 - 200ms 50 - 2000ms

DELAYVARIANCE(JITTER)

0 - 1ms 10 - 20ms 10 - 100ms 10 - 300ms

LOSS 0% 1 - 2% 1 - 5% 1 - 30%

OUT OFSEQUENCEPACKETS

Does not occur Correction required butadds to delay

Corrected Corrected

Voice vs. Data Networks

IP Network Changes

QoS for voice (router and LAN switches)

Reduce delay

Increase bandwidth

VLANs for voice

Reduce router hop count

Change routing protocol (RIP, OSPF)

IP Network Performance Solutions

TECHNIQUE STANDARD PROPRIETARY CoS QoSIMPACT

GWPhone

Router

DiffServ Yes — Yes No Yes Yes

MPLS Yes — Yes Yes None Yes

RSVP Yes — Yes Yes Yes Yes

IPv6 Yes — Yes ? Yes Yes

L4Switching — Yes Yes No None Yes

CB WFQ — Yes Yes No None Yes

The Five 9s (99.999%)Hardware availability calculated by parts count method (two years to get field experience)Software availability

No standard for calculationSoftware stability is an issue

Power availabilityRequires UPS with auto restartGenerator backupUPS monitoringFour-hour UPS response service

Network availabilityRedundant componentsAutomatic switchover

Design / Test / Tune

Measure delay and variance in ms Keep modem access speeds > 28.8 KpbsUse private line access to ISPKeep utilization < 50% for access lineLimit router hops to five or lessAssign 12 to 16 Kbps for each voice transmission

1. One-Way Delay: Acceptable = ________________ msMaximum = ________________ msImperceptible < ________________ ms

2. Round-Trip Delay: Acceptable = ________________ msMaximum = ________________ msImperceptible < ________________ ms

3. Delay Variance: Acceptable = ________________ msMaximum = ________________ msImperceptible < ________________ ms

4. Male / Female RecognitionG Required G Not Required

5. Speaker RecognitionG Required G Not Required

6. Packet Dropout (Loss): Worst Case = ________________ %Imperceptible < ________________ %

Quality Worksheet

Section 9

Measuring Voice Quality (MOS)

Voice Conversation Quality— Loudness (volume)

— Distortion

— Noise

— Fading

— Crosstalk

— Echo

— End-to-end delay

— Silence suppression performance

— Echo canceller performance

Same asSound Quality

Voice Quality RelationshipsIncreasing Echo

Acceptable VoiceQuality

Decreasing Clarity (Due to compression

and loss)Increasing Delay

(Due to compression andcongestion)

Measuring Voice QualityObjective measurement

E-ModelPerceptual models• PSQM• PESQ

Subjective measurementMean Opinion Score (MOS)MOS is scoring by human listenersEnd point locations satisfying standards for ambient noise (ITU-T P.800 and P.830)

Mean Opinion Score (MOS)A numeric measure of the voice quality5 = perfect; 4.4 = toll quality 3.5 = marginally acceptable30+ people listening to sounds score the MOSIndustry moving to device measurement of MOS

Standard Speed MOS Delay + Processing G.711 64 Kbps 4.4 0.75 ms (5 ms)

G.726 32, 24, 16 Kbps 4.2 @ 32 Kbps 1 ms (10 ms)

G.728 16 Kbps 4.2 3 to 5 ms (10 ms)

G.729/A 8 Kbps 4.2 10 ms (14 ms)

G.723.1 6.3, 5.3 Kbps 4 @ 6.3 Kbps 30 ms (37 ms) 3.5 @ 5.3 Kbps

Mean Opinion Score (MOS) by people: 5 = Excellent4 = Good3 = Fair2 = Poor1 = Bad

Delay in ( ) includes processing.

Standard Speed MOS Delay + Processing G.711 64 Kbps 4.4 0.75 ms (5 ms)

G.726 32, 24, 16 Kbps 4.2 @ 32 Kbps 1 ms (10 ms)

G.728 16 Kbps 4.2 3 to 5 ms (10 ms)

G.729/A 8 Kbps 4.2 10 ms (14 ms)

G.723.1 6.3, 5.3 Kbps 4 @ 6.3 Kbps 30 ms (37 ms) 3.5 @ 5.3 Kbps

CODEC Voice Scores

VoIP Measurement StandardsPSQM (ITU P.861)/PSQM+ : Perceptual Speech Quality MeasurementMNB (ITU P.861) : Measuring Normalized BlocksPESQ (ITU P.862) : Perceptual Evaluation of Speech QualityPAMS (British Telecom) : Perceptual Analysis Measurement SystemE-Model (ITU G.107)

E-Model StandardITU-T G.107Based on impairment levelsR value translate into MOS (R = 100 is MOS 5)Computes R value

G.711 R = 95 (64 Kbps)G.726 R = 87 (32 Kbps)G.728 R = 87 (16 Kbps)G.729 R = 84 (8 Kbps)G.723.1 R = 80 (6.3 Kbps)

Codec Delay Impairments100

90

84

95

80

70

60

5050 100 500400300200

One-way Delay in ms

R Value

Toll QualityG.723.1 (6.3 Kbps)

G.729 (8 Kbps)

G.711 (64 Kbps)

R Value vs. MOSR Value MOS Score

100

94

0

50

60

70

80

904.4

2.6

3.1

3.6

4.3

4.0

User Perception

Very Satisfied

Satisfied

Some Dissatisfaction

Many Dissatisfied

Almost All Dissatisfied

Awful

G.107 Default Value is 94

VoIP Equipment ChangesLess compressionSmaller packets (10 vs. 20, 30, 60 ms)Turn off silence suppressionLarger jitter bufferElevate priority of voice programs in softphonesUse WINDOWS XP with softphone

Section 10

The Future of VoIP/IPT

What to Evaluate

What does it do for me?Where does it fit?The Total Cost of Ownership (TCO) and calculation elementsConnection to my legacy worldBenefits (savings, market)Technology health

continued

What to Evaluate (continued)

Risk ($, job, liabilities)Standards vs. proprietary productsCompetitive environmentsMarket / vendor stabilityRegulatory and legislative requirements

Wireless Network

Mobile Workstation

Access Points

LAN Switch /Firewall

ServerWorkstation

Workstation Server

Security IssuesHacker (internal or external)Viruses/worms/trojan horsesSpywareUser authenticationAccess authorizationVulnerability during moves/addsDenial of service

Security PositionsAPP Serve rs Web Serve r

Internal IP Network

Firewall

External IP Network

Intrusion Prevention (IPS)

Intrusion Detection(IDS)

Content SwitchVoIP Cal l

Server

VoIP F irewal l

IDS

IDS

G at ewa y wi th F i re wa l l

Where is the Enterprise VoIP Market Going?

Average number of stations: 100+ in 2003, to 150+ in 2004, 170+ in 2005Acceptance for large sites, 1000s of stationsLittle internet usageACD, CTI, and web site integrationThird-party applications support softwareThird-party network management software

continued

Where is the Enterprise VoIP Market Going? (continued)

Cisco tying VoIP, router and LAN switch products together (dependent).Cisco offers Call Manager Express as software in routerVoice-centric vendors not offering new TDM switchesVoice-centric vendors agnostic of IP/LAN networksIndependent IP phone vendors growIncreased vendor interoperability testing

Top Drawbacks of Deploying VoIP(part 1)

0 5 10 30252015

39%

16%

9%

Source: BCR Magazine

Deployment was more difficultthan anticipated

Users complain about voice quality

35 40

The VoIP system was more difficultto manage than anticipated

Upgrading/maintaining traditionalPBXs is too expensive

Have not been able to decreasenetwork staff

Our primary vendor made falseclaims

6%

8%

6%

Top Drawbacks of Deploying VoIP(part 2)

0 5 10 30252015

3%

1%

Source: BCR Magazine

Have not been able to deployvoice functionality

Domestic calls between companysites not cheap enough

35 40

Not easier to deploy integratedapplications

Wiring not cheap enough

10%

1%

1%

1%

Other

Moves / adds / changes notcheap enough

Questions to Ask Your Vendor1. Where is the vendor using its own IP-PBX? How? Are they mission

critical applications?

2. Does the vendor still use legacy PBXs, key systems and phones? If so, why?

3. What is the vendor’s local staff support for the IP-PBX?

4. Where does the customer get the staffing and training to support the IP-PBX?

5. How is the H.323 to SIP migration being supported? When? Forklift upgrade?

6. Does the vendor’s IP-PBX increase/decrease

a. Security

b. Moves / Adds / Changes (MACs)

c. Labor required

d. Ease of use (user and administrator)

e. Management functions