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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
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