Pass4sure Cisco Certification

Cisco Voice Over IP Exam

Retired July 26, 2008
Exam Number: 642-432
Associated Certifications: CCVP, Cisco IP Communications Support Specialist, and Cisco Rich Media Communications Specialist
Duration: 75 minutes (55 – 65 questions)
Available Languages: English
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Exam Description Exam Topics Recommended Training Additional Resources
Exam Description
The Cisco Voice over IP 642-432 CVOICE exam is associated with the CCVP, Cisco IP Communication Support Specialist, and Cisco Rich Media Communications Specialist certifications. Candidates can prepare for this exam by taking the Cisco Voice Over IP (CVOICE) course. This exam assesses a candidate’s knowledge of how to implement and support data and voice integration solutions at the network-access level. Topics covered include basic IP Telephony operation, router configuration, support, troubleshooting, and integration with an existing PSTN network.

Exam Topics
The following information provides general guidelines for the content likely to be included on the exam. However, other related topics may also appear on any specific delivery of the exam. In order to better reflect the contents of the exam and for clarity purposes the guidelines below may change at any time without notice.

Describe the similarities and differences between PSTN and VoIP including call transport, call signalling and bandwidth requirements
Identify PSTN characteristics, transport
Explain TDM and statistical multiplexing as it relates to telephony
Identify VOIP characteristics, transport
Compare operation of PSTN call signalling to VoIP call signalling

Configure a router so that a basic IP Telephony call can be completed
Configure dial-peer settings for VoIP or POTs
Configure POTs ports (FXS and FXO)
Configure E&M
Explain default dial-peer
Explain matching inbound/outbound dial peers
Configure hunt groups
Explain digit collection, consumptions and manipulations
Configure gateway/gatekeeper
Configure call signalling

Troubleshoot a basic IP telephony configuration
Use show/debug commands to troubleshoot invalid dial-peer configurations
Use show/debug commands to troubleshoot invalid port configuration and operation
Use show/debug commands to troubleshoot invalid signalling configuration
Use show/debug commands to troubleshoot invalid hunt-group configurations
Use show/debug commands to troubleshoot invalid digit manipulation

Successfully integrate a basic IP telephony network into the existing PSTN network
Explain port requirements for connection to a PBX
Explain signalling requirements for connection to a PBX
Choose correct connection type between PBX and VoIP networks
Compare and contrast a traditional dial plan to a VoIP Dial Plan
Identify telephony infrastructure terms (ie. LEC, CLEC, RBOC, ILEC, IXC)

Converse with traditional TDM support personnel using their terminology
Identify TDM signaling
Identify port and line types
Identify addressing signalling

Describe the basic operation and components involved in a VoIP call
Explain RTP, RTCP, cRTP
Explain H323
Explain MGCP
Explain SIP
Explain E&M
Describe the process of analog to digital conversion
Describe the process of compounding and compressing
Explain the process of packetization (frames, codec types)
Choose the appropriate codec for a given situation
Explain the function/operation/purpose of call-legs
Explain voice quality considerations
Explain the purpose and value of QoS

Successfully connect a basic VoIP net to the PSTN
Choose the correct gateway connection to the PSTN
Determine proper signalling options
Describe digit manipulation
Explain E164 addressing
QUESTION 1:
Which type of signaling is DTMF? A. Supervisory
B. Route
C. Informational
D. Address
Answer: D Explanation:
Start dial supervision is the line protocol that defines how the equipment seizes the E&M
trunk and passes the address signaling information such as dual tone multifrequency
(DTMF) digits. Address Signaling
Address signaling typically represents the digits dialed (called party’s number). There are two options used to pass address information. Either Pulse dial (rotary dialing) or Tone
dial (DTMF) can be used. The default for Cisco routers and gateways is DTMF.

QUESTION 2:

You have set up a complex dial plan using translation rules. The following translation rule has been configured.
What output would correspond to the test translation-rule command?
translation-rule 1
rule 0 ^0.. 215550210 rule 1 ^1.. 215550211 rule 2 ^2.. 215550212 rule 3 ^3.. 215550213 rule 4 ^4.. 215550214 rule 5 ^5.. 215550215 rule 6 ^6.. 215550216 rule 7 ^7.. 215550217 rule 8 ^8.. 215550218 rule 9 ^9.. 215550210

A. test translation-rule 1512
The replaced number: 21555021512
B. test translation-rule 1555
The replaced number: 55521555021
C. test translation-rule 1617
The replaced number: 61721555021
D. test translation-rule 1910
The replaced number: 21555021910

TK

Answer: A

Explanation:
New Delhi(2-digit indial range)
!— Only relevant “IOS translation rule” output is presented
!
translation-rule 1
!– The “1″ above is the tag for the set. rule 0 ^0. 1011000
rule 1 ^1. 1011001 rule 2 ^2. 1011002 rule 3 ^3. 1011003 rule 4 ^4. 1011004 rule 5 ^5. 1011005 rule 6 ^6. 1011006 rule 7 ^7. 1011007 rule 8 ^8. 1011008 rule 9 ^9. 1011009
!
!– These rules replace the first digit of a 2-digit number with the corresponding
!– translation. The router looks for a 2-digit number starting with a leading [0-9].
!– The caret, “^” ensures the match only happens at the start of the digit string
!– rather than any occurrence in a digit string. This ensures the router makes the
!– translation only for the leading digits. By default, if an explicit match is made
!– on a digit (in this case the first digit) the router replaces it with the new
!– digits. Therefore, to keep the original numbering, the matched digit needs to be
!– replaced with the same digit at the end of the modified string. Once the call
!– comes in, the called number prepended with 101100 followed by the
!– original 2 digits.
!
voice-port 1/0:1 translate called 1 cptone IN
compand-type a-law
!
!– The translation rule is applied to the voice port where the
!– call comes in to the router. When a call comes in from the
!– telephone network towards the router, the called number
!– is translated before it is matched on any dial peers.
!
dial-peer voice 100 voip destination-pattern 101100..
session target ipv4:main site IP address ip precedence 5
dtmf-relay h245-alphanumeric

TK

!
!– The VoIP dial peer needs to be configured to match on the new numbering plan
This output was captured from the NewDelhi router which shows the translation rules applied while dialing from the NewDelhi site.
NewDelhi- Output
!– It is possible to confirm the translation rules are working:!!NewDelhi#test translation-rule 1 99!– Original called number is “99″The replaced number:
10110099!– Translated to 8 digits

QUESTION 3:
What is the optimal end-to-end delay that should be achieved in a VoIP network? A. 20 ms
B. 100 ms
C. 150 ms
D. 400 ms
Answer: C Explanation:
Delay Specifications

QUESTION 4:
Which three are supervisory signals? (Choose three) A. busy
B. on hook
C. off hook
D. call waiting

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Recommended Training about 642-432 exam PDF
The following courses are the recommended training for 642-432 exam PDF.
642-432 Q & A with Explanations
642-432 Audio Exam
642-432 Study Guide
642-432 Preparation Lab

Exam Number/Code: 642-432
Exam Name: IP Communications Cisco Voice Over IP
VUE Code: 642-432
Questions Type: Single choice, Multiple choice, Simulate,
Question Numbers of Real-exam: 55 – 65 questions

“IP Communications Cisco Voice Over IP”, also known as 642-432 exam, is a Cisco certification.
Preparing for the 642-432 exam? Searching 642-432 Test Questions, 642-432 Practice Exam, 642-432 Dumps?

With the complete collection of questions and answers, Pass4sure has assembled to take you through 72 Q&As to your 642-432 Exam preparation. In the 642-432 exam resources, you will cover every field and category in IP Communications helping to ready you for your successful Cisco Certification.

Questions and Answers : 72 Q&As
Updated: April 25th , 2008
Market Price: $125.99
Member Price: $89.99

Exam Number/Code: 642-432
Exam Name: IP Communications Cisco Voice Over IP
VUE Code: 642-432
Questions Type: Single choice, Multiple choice, Simulate,
Question Numbers of Real-exam: 55 – 65 questions

“IP Communications Cisco Voice Over IP”, also known as 642-432 exam, is a Cisco certification.
Preparing for the 642-432 exam? Searching 642-432 Test Questions, 642-432 Practice Exam, 642-432 Dumps?

With the complete collection of questions and answers, Pass4sure has assembled to take you through 72 Q&As to your 642-432 Exam preparation. In the 642-432 exam resources, you will cover every field and category in IP Communications helping to ready you for your successful Cisco Certification.

Pass4sure 642-432 CVOICE
Questions and Answers : 72 Q&As
Updated: April 25th , 2008
Market Price: $125.99
Member Price: $89.99

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