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CCVP Prep: Cisco IP Telephony Essentials

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Session Goals
1. Cisco Certified Voice Professional 2. To grasp the “big picture” behind VoIP 3. To deliver key concepts and configurations related to the current CVOICE certification exam
Exam Tip!

4. To demonstrate on live equipment the methods used to configure a working VoIP network

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Disclaimer
This Session Will Strictly Adhere to Cisco’s Rules of Confidentiality We may not be able to address your specific question If you have taken the exam please refrain from asking questions from the exam We will be available after the session to direct you to resources to assist with specific questions or to provide clarification

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Cisco Certified Voice Professional
“Earning a CCVP certification validates a robust set of skills in implementing, operating, configuring, and troubleshooting a converged IP network. The certification content focuses on Cisco Systems CallManager, quality of service (QoS), gateways, gatekeepers, IP phones, voice applications, and utilities on Cisco routers and Cisco Catalyst switches.”
642-432 CVOICE 642-444 CIPT 4.X (Windows)/642-445 CIPT 5.X (Linux) 642-642 QoS 642-425 IPTT (Retiring 10.2.2007)/642-426 TUC 642-453 GW/GK
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Exam Topics 642-432
Describe the similarities and differences between legacy TDM and VoIP Configure a router so that a basic IP Telephony call can be completed Troubleshoot a basic IP telephony configuration Successfully integrate a basic IP telephony network into the existing PSTN network Describe the basic operation and components involved in a VoIP call Successfully connect a basic VoIP network to the PSTN
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Exam Format
Test Practical Implementation Skills of VoIP Question formats
Declarative Procedural Complex procedural (simulation) Drag and drop

Avoided question formats
“Trick questions” Version dependencies

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Exam Format—Declarative
A Declarative Exam Item Tests Simple Recall of Pertinent Facts:
What is the optimal end-to-end delay that should be achieved in a VoIP network? 1. 20 ms 2. 100 ms 3. 150 ms 4. 400 ms

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Exam Format—Procedural
A Procedural Exam Item Tests the Ability to Apply Knowledge to Solve a Given Issue:
You are the network technician at Widgets Inc. Your newly appointed trainee wants to know what the connection tie-line command emulates. What will your reply be?

1. A temporary connection to a PBX. 2. A permanent connection to a PBX. 3. A temporary connection to the PSTN. 4. A permanent connection to the PSTN. 5. A permanent connection to the network.

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Exam Format—Complex Procedural
A Complex Procedural Exam Item Tests the Ability to Apply Multiple Knowledge Points to Solve a Given Issue: Configure POTS and VoIP dial-peers to ensure the analog phone can reach the PSTN. The phone number of the analog phone connected to port 1/0/0 is 4805980111.
Router Router POTS Analog Phone

Workstation
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Exam Format—Drag and Drop
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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Outline
The Vision and Benefits of VoIP The Phases of VoIP Growth Centralized and Decentralized Call Management Dial-Peer Concepts and Configuration

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IP Telephony: The Vision
What networks look like today Why an organization would use IP Telephony What it takes to move to IP Telephony
Phase One Migration Phase Two Migration

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Networks Today

Separate networks for voice, video, and data applications Difficult to integrate applications Voice network not designed to handle data+voice+video demands

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Converged Networks

By combining all communication methods into a single network infrastructure, INSANELY cool things are possible
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Why an Organization Would Use IP Telephony
Hard Cost Savings
Moves/Adds/Changes to the phone system Reduced wiring in new buildings Reduced expenses for telecommuters/branch offices IT staff consolidation Free long distance (Toll Bypass) Application consolidation Lower transmission Costs

Soft Cost Savings
Single inbox for messages (voicemail/fax/email) Extension mobility (saves office space) Internet Website integration into phone system (customers love it) Open architecture (invites multi-vendor integrations)
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Exam Tip!
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Phase One IP Telephony Migration

Digital Phones
WAN

Digital Phones

PBX

PBX

PBX and digital telephones still in place Calls routed over the IP WAN rather than PSTN Major benefit: free long distance Major requirement: Quality of Service (QoS) in place
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Phase Two IP Telephony Migration
IP Phone Cisco CallManager Cisco CallManager IP Phone

WAN

IP Phone

IP Phone

Softphone

Softphone

Voice and Data network merged PBX and digital phones sold on Ebay New applications can now integrate into voice network Most newly constructed structures move directly to this phase
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Cisco’s Strategy

Voice

Cisco Unified Communications

Video

Data

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Understanding Packet Telephony
Centralized vs. decentralized call control Packet telephony components Digital vs. analog connections IP telephony call types

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Distributed Call Control

Exam Tip!
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Centralized Call Control
Exam Tip!

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Packet Telephony Components
Digital Phones CallManager

PSTN

PBX

IP Phone Voice-Enabled Switch Voice-Enabled Gateway

WAN

IP Phone

Voice-Enabled Gateway

PC

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VIC Cards: Analog Connections
Most mainline Cisco routers (1800/2800/3800) now support Voice Interface Cards (VICs) which allow for a variety of analog voice connections Analog connections support a single call for each voice port There are three categories of analog voice ports:
Foreign Exchange Station (FXS) Foreign Exchange Office (FXO) Ear and Mouth (E&M)

Cisco VIC-2FXS
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Analog Connections: FXS
FXS Analog Phone

FXS

Fax

FXS

Modem

Provides a direct connection to an analog phone, fax machine, or similar device Exam Tip!
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Analog Connections: FXO
FXO Station Interface PBX

FXO

Telco CO

Allows an analog connection to be directed to the PSTN or station interface of a PBX
Exam Tip!
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Analog Connections: E and M
E&M Trunk Interface PBX

Provides a trunk connection to a PBX

Exam Tip!
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How Far Will Those FXS Ports Go?
A single FXS Port will go a distance of 3,200 feet A single FXS Port can power a loop of up to 8 analog devices (in a home, for example) The total distance of an FXS connection diminishes the more phones you attach to the port

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Digital Voice Ports

T1

Central Office

Typically used in corporate environments with high volume requirements Options of T1/E1 (CAS/CCS) and BRI
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The New All-in-One Module: VWICs
Traditional T1 WIC cards are designed for data and are unable to communicate with legacy voice networks The VWIC T1 module allows you to connect to both data and voice networks

Cisco VWIC-2MF-T1/E1

Exam Tip!
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Installing Voice Ports

Installing voice ports require NM-1V/NM-2V network modules + VIC cards Some 17xx and ISR series routers have built-in VIC slots, but require Packet Voice DSP Module (PVDM)
Exam Tip!
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Local Calls

Exam Tip!
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On-Net Calls

Exam Tip!
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Off-Net Calls

Exam Tip!
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Private Line Automatic Ringdown (AKA Batphone)

Exam Tip!
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On-Net to Off-Net Calls

Exam Tip!
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The Configuration Dividing Line
Breath deep. You are about to enter a whole new world of configurations Get your pens ready! Configuration breakdown:
FXS/FXO physical port configuration Dial-Peer basic concepts and configuration Digital T1/PRI physical port configuration Dial-Peer advanced concepts and configuration

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FXS Port Configuration
Exam Tip!
FXS 1/0/0 Analog Phone Router# configure terminal Router(config)# voice-port 1/0/0 Router(config-voiceport)# description Phone on the dock of a bay Router(config-voiceport)# signal loop-start Router(config-voiceport)# cptone GB Router(config-voiceport)# end Router#

Foundation FXS port configuration Physical PORT configuration mode only assigns aspects that affect the physical port itself, NOT dial-plan information
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FXS Command Reference
Router(config-voiceport)# signal <ground-start | loop-start>

Sets the signaling type; most often loop-start (default) for analog phones, ground-start for PBX-to-PBX connections
Router(config-voiceport)# cptone <locale>

Defines the detection of call-progress tones generated at the local interface
Router(config-voiceport)# ring frequency <25 | 50>

Configures ring frequency (in hertz for OLD analog phone handsets. If incorrect, handset may not ring (or will buzz instead of ring)

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FXS Command Reference (Cont.)
Router(config-voiceport)# ring cadence <pattern | define>

Defines length of ringing voltage. Default to North America. Is automatically set by cptone command
Router(config-voiceport)# busyout

Gives you the ability to busy out a port
Router(config-voiceport)# station id name <15_character_name> Router(config-voiceport)# station id number <15_character_number>

Configures caller-id information for FXS port (may be overridden by FXO port configuration)

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FXO Port Configuration
Exam Tip!
FXO 1/0/0
PSTN

Router# configure terminal Router(config)# voice port 1/0/0 Router(config-voiceport)# description Backup PSTN Connection Router(config-voiceport)# dial-type dtmf Router(config-voiceport)# ring number 1 Router(config-voiceport)# end Router#

Foundation FXO port configuration Physical PORT configuration mode only assigns aspects that affect the physical port itself, NOT dial-plan information
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FXO Command Reference
Router(config-voiceport)# dial-type <dtmf | pulse>

Chooses appropriate dial-type for outbound calls
Router(config-voiceport)# signal <ground-start | loop-start>

Sets the signaling type; most often loop-start (default) for analog phones, ground-start for PBX connections
Router(config-voiceport)# cptone <locale>

Defines the detection of call-progress tones generated at the local interface

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FXO Command Reference (Cont.)
Router(config-voiceport)# ring number <number>

Number of rings before incoming call is answered by the router
Router(config-voiceport)# busyout

Gives you the ability to busy out a port
Router(config-voiceport)# station id name <15_character_name> Router(config-voiceport)# station id number <15_character_number>

Configures caller-id information for FXO port (overrides FXS port configuration)

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Dial Peers: The Routing Table for Voice
Provide the logical configuration and connections for voice ports Used to create a “routing table” for voice calls Can be POTS when connecting to traditional PSTN connections (such as FXS/FXO ports) or VoIP when connecting to a packet network

Exam Tip!
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POTS Dial Peer Configuration
Exam Tip!
FXS 1/0/1 x1111 Router# configure terminal Router(config)# dial-peer voice 1 pots Router(config-dialpeer)# destination-pattern 1111 Router(config-dialpeer)# port 1/0/1

Similar to creating a static route for extension 1111 Destination-pattern command configures the telephone number Port command associates the physical port with the logical extension
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VoIP Dial Peer Configuration
FXS x1111 R4
10.1.1.1/30 10.1.1.2/30

FXS R5 x2222

R4# configure terminal R4(config)# dial-peer voice 2 voip R4(config-dialpeer)# destination-pattern 2222 R4(config-dialpeer)# session target ipv4:10.1.1.2 R5# configure terminal R5(config)# dial-peer voice 2 voip R5(config-dialpeer)# destination-pattern 1111 R5(config-dialpeer)# session target ipv4:10.1.1.1

When destination-pattern is matched, router immediately forwards the call (no further digits are collected)
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Exam Tip!
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Configuring Digital Interfaces
T1/E1 CAS Port Configurations T1/E1 CCS Port Configurations

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Digital T1 CAS Physical Port Configuration
Step 1: Specify Framing, Linecode, and Timing
Router# configure terminal Router(config)# controller t1 1/0 Router(config-controller)# framing < sf / esf > Router(config-controller)# linecode < b8zs / ami >

Exam Tip!

Router(config-controller)# clock source < internal / line >

Framing and Linecode are specified by the service provider Timing can be either internal if your router is providing the clocking, or line if it is receiving the clock rate from an external source
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Digital T1 CAS Physical Port Configuration
Step 2: Create ds0-group
Router# configure terminal Router(config)# controller t1 1/0 Router(config-controller)# ds0-group <number> timeslots <1-24> type <signaling>

Ds0-group number is locally significant, timeslots must match timeslots in use Signaling must match between end devices
E&M signaling types typically connect to PBX trunk lines & telco equipment FXS signaling types typically connect to PBX equipment FXO signaling types typically connect to the C

Once you have created the ds0-group, the router will create a voice port using the ds0-group number you have chosen. For example:
T1 1/0, ds0-group 5 becomes voice port 1/0:5 T1 2/1, ds0-group 1 becomes voice port 2/1:1
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T1 CAS Configuration Example

PSTN

T1 1/0

Router# configure terminal Router(config)# controller t1 1/0 Router(config-controller)# ds0-group 1 timeslots 1-24 type fxo-loop-start Router(config-controller)# framing esf Router(config-controller)# linecode b8zs Router(config-controller)# clock source line

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T1 CCS (ISDN PRI) Configuration
T1 1/0 Router# configure terminal Router(config)# isdn switch-type primary-qsig Router(config)# controller t1 1/0 Router(config-controller)# pri-group 1 timeslots 1-24 Router(config-controller)# exit Router(config)# interface serial 1/0:23 Router(config-if)# isdn incoming-voice voice

PBXs are commonly equipped with a PRI interface Most Cisco T1 interfaces can double as an ISDN PRI interface ISDN PRI D-channel is always channel 23 (T1) or 15 (E1)
Exam Tip!
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Dial-Peer Destination Patterns…Let’s Step it Up!
Plus (+) Comma (,) Period (.) Brackets ( [ ] ) T (Optional) Indicates an E.164 standard number Inserts a pause between digits Matches any single digit Indicates a range of numbers Allows a variable-length dial string
Exam Tip!
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Destination Pattern Examples
Matches one telephone number exactly, 5551212 Matches a seven-digit telephone number where the first three digits are 555, the fourth digit can be 5, 6, 7, 8, or 9, and the last three digits can be any valid digit Matches any telephone number where the first digit is 9. T wildcard can vary in length from 0 to 32 digits total
Exam Tip!
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5551212

555[5-9]…

9T

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A Few More Destination Pattern Examples
555[14-6]…

555121[^135]

[1-79]..[^1]5

Exam Tip!
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Practice Question
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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Practice Question
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2

5551199

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555121

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5451181

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Practice Question
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2

5551199

555121

Place Here

5451181

Place Here

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Practice Question
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2

5551199

555121

5451181

Place Here

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Practice Question
Click and drag the dial peer to the phone number it would match. Answers may be used more than once or may not be used at all.
dial-peer voice 5 pots destination-pattern .T port 1/0/0 dial-peer voice 6 pots destination-pattern 555[1-2].. port 1/0/1 dial-peer voice 7 pots destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots destination-pattern 545118. port 1/0/3
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dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 7 pots Place Here destination-pattern 5551 port 1/0/2 dial-peer voice 8 pots

5551199

555121

Place Here 5451181 destination-pattern 545118. port 1/0/3
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Selecting a CODEC
Voice over IP Coder/DECoders (CODECs) dictate the type of compression that will be used when converting voice signals Three primary CODECs are used in the Cisco VoIP environment:
G.711 (Uncompressed–64Kbps) G.723 (Compressed–6.3Kbps) G.729 (Compressed–8Kbps)
Exam Tip!

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Digging Deeper into CODECs
Each CODEC offers different advantages
Voice quality CODEC complexity Coder delay

In the case of G.729, three flavors exist:
G.729 (original)–high complexity, 3.92 MOS G.729a–medium complexity, 3.7 MOS G.729ab–medium complexity, 3.7 MOS, VAD
Exam Tip!

In addition, Cisco routers support CODEC packet loss concealment
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Digging Deeper into CODECs
Each CODEC will also offer a sample size
By default, 20 ms of audio per packet

Increased sample size offers more bandwidth savings
Tradeoff: Increased delay Tradeoff: Increased impact on packet loss
Exam Tip!

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Configuring CODEC Selection
Router#config t Router(config)#dial-peer voice 10 voip Router(config-class)#codec ? g711alaw g711ulaw g723ar53 g723ar63 g723r53 g723r63 g726r16 g729br8 g729r8 gsmfr G.711 A Law 64000 bps G.711 u Law 64000 bps G.723.1 ANNEX-A 5300 bps (contains built-in vad that cannot be disabled) G.723.1 ANNEX-A 6300 bps (contains built-in vad that cannot be disabled) G.723.1 5300 bps G.723.1 6300 bps G.726 16000 bps G.729 ANNEX-B 8000 bps (contains built-in vad that cannot be disabled) G.729 8000 bps GSMFR 13200 bps

The default CODEC used for VoIP dial-peers is G.729
Exam Tip!

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Configuring CODEC Selection (Cont.)
Router#config t Router(config)#voice class codec 99 Router(config-class)#codec preference 1 g711ulaw Router(config-class)#codec preference 2 g729br8 Router(config-class)#exit

Router(config)#dial-peer voice 2000 voip Router(config-dial-peer)#voice-class codec 99 Router(config-dial-peer)#^Z

Exam Tip!

Voice classes allow you to set up ‘failover’ CODECs during VoIP negotiations

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Wrap-Up: Session Goals
1. Cisco Certified Voice Professional 2. To grasp the “big picture” behind VoIP 3. To deliver key concepts and configurations related to the current CVOICE certification exam 4. To demonstrate on live equipment the methods used to configure a working VoIP network
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Q and A

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Recommended Reading
Continue your Cisco Live learning experience with further reading from Cisco Press Check the Recommended Reading flyer for suggested books

Available Onsite at the Cisco Company Store
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Recommended Reading
Continue your Cisco Live learning experience with further reading from Cisco Press Check the Recommended Reading flyer for suggested books

Available Onsite at the Cisco Company Store
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