Cisco 642-432 : Cisco Voice Over IP

  • Exam Code: 642-432
  • Exam Name: Cisco Voice Over IP
  • Updated: Sep 30, 2026
  • Q & A: 72 Questions and Answers

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Cisco 642-432 Exam Syllabus Topics:

SectionObjectives
Topic 1: Call Operation and Components in Voice over IP- Call Operation and Components
  • 1. Explain Quality of Service (QoS)
  • 2. Describe companding and compression processes
  • 3. Explain MGCP
  • 4. Explain packetization and codec types
  • 5. Choose appropriate codecs for different scenarios
  • 6. Explain the function and operation of call legs
  • 7. Explain voice quality considerations
  • 8. Describe analog-to-digital conversion
  • 9. Explain RTP, RTCP, and CRTP
  • 10. Explain SIP
  • 11. Explain H.323
  • 12. Explain E&M signaling
  • 13. Describe the basic operation and components of an IP telephony call
Topic 2: Voice over IP Configuration- Voice over IP Configuration
  • 1. Configure E&M interfaces
  • 2. Explain inbound and outbound dial peer matching
  • 3. Explain digit collection, consumption, and manipulation
  • 4. Configure hunt groups
  • 5. Configure FXS and FXO ports
  • 6. Configure call signaling
  • 7. Configure a router for a basic IP telephony call
  • 8. Configure VoIP and POTS dial peers
  • 9. Configure gateways and gatekeepers
  • 10. Explain default dial peers
Topic 3: Integrating Voice over IP into the PSTN- PSTN Integration
  • 1. Explain E.164 addressing
  • 2. Describe digit manipulation
  • 3. Connect a basic VoIP network to the PSTN
Topic 4: Integrating Voice over IP into Existing PBX Networks- PBX Integration
  • 1. Explain port requirements for PBX connectivity
  • 2. Explain signaling requirements for PBX connectivity
  • 3. Select appropriate connection types between PBX and VoIP networks
  • 4. Integrate a basic IP telephony network with an existing PBX
Topic 5: Voice over IP Technologies- Voice over IP Technologies
  • 1. Compare PSTN call signaling with VoIP call signaling
  • 2. Differentiate VoIP technologies from PSTN technologies
  • 3. Identify VoIP characteristics and transport mechanisms
  • 4. Compare PSTN and VoIP including call transport, signaling, and bandwidth requirements
  • 5. Identify PSTN characteristics and transport mechanisms
  • 6. Explain TDM and statistical multiplexing in telephony

Cisco Voice Over IP Sample Questions:

Question #1

Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The three remote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks to connect the PBXs today.
What kind of connection should be implemented to allow a two-way permanent connection between R1 and R4?

  • A. answer-mode
  • B. trunk
  • C. tie-line
  • D. PLAR
Reveal Solution  Discussion  0

Correct Answer: B  🗳️

Question #2

You are working with a potential customer that would like to integrate its existing PBX telephone system into its IP network. The accompanying figure shows that the customer has two offices that need to be connected to the IP network so that the customer can exchange telephone calls without using the PSTN. Both PBXs use Wink-Start signaling.
Which signaling type will allow you to support the customer?

  • A. FXO
  • B. E&M
  • C. QSIG
  • D. CAS
  • E. T-CCS
  • F. CCS
Reveal Solution  Discussion  0

Correct Answer: B  🗳️

Question #3

Which two codecs are required on a Gateway to support CNG and VAD across a WAN link without major impact to call quality and bandwidth consumption? (Choose two.)

  • A. G.729
  • B. G.729b
  • C. G.729ab
  • D. G.723
  • E. G.729a
Reveal Solution  Discussion  0

Correct Answer: B,C  🗳️

Question #4

Which device is used to allow an H.323 stream to transit a firewall?

  • A. MCU
  • B. gatekeeper
  • C. proxy
  • D. gateway
Reveal Solution  Discussion  0

Correct Answer: C  🗳️

Question #5

You are working with your customer in their lab to test the effect of jitter on voice quality. You have set the maximum playout delay to 40 ms on the voice enabled routers. What will be the impact on voice quality if after severe congestion the playout buffer empties and the source sends packets to the buffer faster than they are leaving?

  • A. The jitter buffer will adapt to the faster-arriving packets by expanding the buffer size.
  • B. There will be no noticeable drop in quality.
  • C. The jitter buffer will speed up delivery of packets to the DSP so that packets are not dropped.
  • D. After the jitter buffer fills up, subsequent packets are discarded.
Reveal Solution  Discussion  0

Correct Answer: D  🗳️

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