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Cisco 642-432 Exam Overview:

Certification Vendor:Cisco
Exam Name:Cisco IP Telephony Part 2 (CIPT2) / Voice Over IP
Exam Number:642-432
Exam Price:USD 200 (approx., varies by region; legacy exam pricing)
Exam Format:Multiple choice, Simulation-based questions
Available Languages:English
Related Certifications:CCNP Voice
CCNA Voice (foundational prerequisite in legacy track)
Cisco IP Telephony
CCVP
Exam Duration:90 minutes
Certificate Validity Period:Retired (legacy Cisco certification exam; no longer offered in current Cisco certification programs)
Recommended Training:Cisco Learning Network (Voice / Collaboration resources)
Exam Registration:Pearson VUE Cisco Exams
Cisco Certification Registration
Sample Questions:Cisco 642-432 Sample Questions
Exam Way:Proctored exam via Pearson VUE testing centers (legacy delivery model)
Pre Condition:Recommended knowledge of Cisco voice fundamentals and CCNA-level networking concepts (legacy CCNA Voice / CCNA Collaboration track recommended at the time)
Official Syllabus URL:https://www.cisco.com/c/en/us/training-events/training-certifications.html

Cisco 642-432 Exam Syllabus Topics:

SectionObjectives
Quality of Service (QoS) for Voice- Voice traffic prioritization
  • 1. Latency, jitter, and packet loss considerations
    • 2. QoS marking and classification (DSCP)
      Cisco Unified Communications Architecture- Call processing design
      • 1. Call routing logic and dial plan concepts
        • 2. Digit manipulation and translation patterns
          - CUCM (Cisco Unified Communications Manager) fundamentals
          • 1. Device pools, regions, and locations
            • 2. System components and architecture overview
              IP Phone Deployment and Features- Endpoint registration and configuration
              • 1. Device configuration templates
                • 2. Cisco IP phone registration process
                  - Advanced telephony features
                  • 1. Conference and transfer services
                    • 2. Call forwarding and hunt groups
                      Voice Gateway and Protocol Integration- PSTN integration
                      • 1. Voice gateways configuration concepts
                        • 2. Dial peers and call legs
                          - VoIP signaling protocols
                          • 1. SIP fundamentals
                            • 2. H.323 call setup and interoperability
                              Call Control and Survivability- SRST (Survivable Remote Site Telephony)
                              • 1. SRST configuration and failover behavior
                                - Redundancy and high availability
                                • 1. CallManager redundancy mechanisms

                                  Cisco Voice Over IP Sample Questions:

                                  Question #1

                                  In a VoIP environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deployed without redundancy checks?

                                  • A. 3 bytes
                                  • B. 40 bytes
                                  • C. 20 bytes
                                  • D. 4 bytes
                                  • E. 1 byte
                                  • F. 2 bytes
                                  Reveal Solution  Discussion  0

                                  Correct Answer: F  🗳️

                                  Question #2

                                  LAB

                                  Examine the following PBX system parameters:
                                  The calling side seizes the line by going off-hook on its E-lead and sends information as DTMF digits.
                                  The voice path is 4-wires, and the voice enabled router is in another building from the PBX.
                                  Select the correct set of commands to allow communication between a voice enabled router and a PBX.

                                  • A. voice port 1/0/0 signal wink-start operation 4-wire type 3
                                  • B. voice port 1/0/0 signal immediate-start operation 4-wire type 2
                                  • C. voice-port 1/0/0 signal delay-dial operation 4-wire type 1
                                  • D. voice port 1/0/0 signal immediate-start operation 4-wire type 4
                                  Reveal Solution  Discussion  0

                                  Correct Answer: B  🗳️

                                  Question #3

                                  Refer to the exhibit.
                                  Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network.
                                  T1 link 1536 kbps
                                  e-mail 75 kbps
                                  internet 200 kbps
                                  Oracle 500 kbps
                                  FTP 250 kbps
                                  Total 1025 kbps
                                  The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no cRTP, and the use of the
                                  G.711 codec.
                                  How many simultaneous calls could be placed on this link?

                                  • A. no more than 20 calls
                                  • B. 0 calls
                                  • C. 1 call
                                  • D. no more than 5 calls
                                  • E. 2 calls
                                  • F. no more than 10 calls
                                  Reveal Solution  Discussion  0

                                  Correct Answer: C  🗳️

                                  Question #4

                                  What is the function of a SIP registrar server?

                                  • A. acquires endpoint addresses
                                  • B. resolves text addresses to IP addresses
                                  • C. routes service requests
                                  • D. resolves active endpoint addresses
                                  Reveal Solution  Discussion  0

                                  Correct Answer: A  🗳️

                                  Question #5

                                  How is CAS different on E1 and T1?

                                  • A. T1 uses the D channel for CAS signaling.
                                  • B. E1 CAS signaling is out-of-band while T1 is in-band.
                                  • C. E1 uses robbed-bit signaling.
                                  • D. T1 has more signaling channels.
                                  Reveal Solution  Discussion  0

                                  Correct Answer: B  🗳️

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