Testinside cisco CCVP 642-432

CCVP Cisco Voice Over IP : 642-432 Exam

642-432 CVOICE
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

“Cisco Voice Over IP”, also known as 642-432 exam, is a Cisco certification.
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QUESTION 21
When they are booted, the Cisco Access Digital Trunk Gateway DT-24+, the Cisco
Access Digital Trunk Gateway DE-30+, and the Catalyst 6000 digital gateway are
provisioned with Cisco CallManager location information. When these gateways
initialize, a list of Cisco CallManager’s, referred to as a ______________ is
downloaded to the gateways.
A. Cisco IPSP group
B. Call managed Cisco redundancy group

C. IPNC redundancy group
D. Cisco CallManager redundancy group
Answer: D
QUESTION 22
Which are three required steps in digitizing voice? (Choose three)
A. Companding the signal.
B. Quantizing the amplitude.
C. Filtering the signal.
D. Sampling the soundwave.
E. Encoding the results in binary form.
Answer: B, D, E
The answer on “Question 1″ on page 15 dealing with digitizing voice should be:
1. Quantizing the amplitude
2. Sampling the soundwave
3. Encoding the results in binary form
See page 2-45 of CVoice version 4.1 class books.
Not C: “Filtering” is part of the process to go from digital to analog not analog to digital
(p. 2-47)
QUESTION 23
You are the VoIP engineer at Certkiller .com. A Certkiller user complains that she
gets a busy tone instead of a dial tone when she tries to call another user. You want
to troubleshoot this problem.
What command should you use?
A. show voice dsp
B. show voice path
C. show voice connection
D. show voice port summary
E. show dial-peer voice summary
Answer: A

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