Cisco 642-437 dumps - in .pdf

642-437 pdf
  • Exam Code: 642-437
  • Exam Name: Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
  • Updated: Sep 09, 2026
  • Q & A: 257 Questions and Answers
  • PDF Price: $59.99
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  • Exam Code: 642-437
  • Exam Name: Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
  • Updated: Sep 09, 2026
  • Q & A: 257 Questions and Answers
  • PDF Version + PC Test Engine + Online Test Engine
  • Value Pack Total: $119.98  $79.99
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Cisco 642-437 dumps - Testing Engine

642-437 Testing Engine
  • Exam Code: 642-437
  • Exam Name: Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
  • Updated: Sep 09, 2026
  • Q & A: 257 Questions and Answers
  • Software Price: $59.99
  • Testing Engine

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Cisco 642-437 Exam Syllabus Topics:

SectionObjectives
Cisco Unified Border Element (CUBE)- CUBE features and configuration
  • 1. Media flow and protocol interworking
    • 2. SIP trunks and interworking
      Troubleshooting- Troubleshooting voice and QoS issues
      • 1. Common voice call issues
        • 2. Diagnostics and show/debug commands
          Quality of Service (QoS)- QoS mechanisms
          • 1. DiffServ and classification
            • 2. AutoQoS and queuing
              Voice Gateways and Call Routing- Gateway implementation and configuration
              • 1. Dial peers and dial plan implementation
                • 2. Voice gateway interfaces and DSPs
                  Cisco Unified Communications Manager Express- CUCME configuration
                  • 1. Call handling and features
                    • 2. Endpoint addressing and ephone setup
                      VoIP Fundamentals- Voice over IP basics
                      • 1. VoIP components and operation
                        • 2. Voice signaling protocols (H.323, SIP)

                          Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Sample Questions:

                          Question #1

                          Which debug command can be used to show events specific to the Cisco Unified Border Element gateway?

                          • A. debug voip ipipgw
                          • B. debug voip cube events
                          • C. debug ip voip cube
                          • D. debug ip cube
                          Answer: A
                          Question #2

                          DRAG DROP
                          Drag the signaling streams to support SIP Early Offer from the left and drop them in the correct box in the graphic on the right.

                          Answer:


                          Explanation:
                          Call Flow of a Typical sip Session

                          Cisco 642-437 Exam

                          Question #3

                          DRAG DROP
                          Drag the attributes of a scalable numbering plan from the left and place them in the boxes on the right.

                          Cisco 642-437 Exam

                          Answer:


                          Explanation:
                          When designing a large-scale dial plan, Cisco recommends you adhere to the following attributes:
                          Logic distribution: Good dial plan architecture relies on the effective distribution of the dial plan logic among the various components. Devices that are isolated to a specific portion of the dial plan reduce the complexity of the configuration. Each component focuses on a specific task accomplishment. Generally, the local switch or gateway handles details that are specific to the local point of presence (POP). Higher-level routing decisions are passed along to the gatekeepers and PBXs. A well-designed network places the majority of the dial plan logic at the gatekeeper devices.
                          Hierarchical design (scalability): You should attempt to keep the majority of the dial plan logic (routing decisions and failover) at the highest-component level. Maintaining a hierarchical design makes the addition and deletion of number groups more manageable. Scaling the overall network is much easier when configuration changes are made to a single component.
                          Simplicity in provisioning. Keep the dial plan simple and symmetrical when designing a network. Try to keep consistent dial plans on the network by using translation rules to manipulate the local
                          digit dialing patterns. These number patterns are normalized into a standard format or pattern before the digits enter the VoIP core. Putting digits into a standard format simplifies provisioning and dial-peer management.
                          Reduction in postdial delay: Consider the effects of postdial delay in the network when you design a large-scale dial plan. Postdial delay is the time between the last digit dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short postdial delay and to hear ringback within seconds. The more translations and lookups that take place, the longer the postdial delay becomes. Overall network design, translation rules, and alternate pathing affect postdial delay. Therefore, you should efficiently use these tools to reduce postdial delay.
                          Availability and fault tolerance. Consider overall network availability and call success rates when you design a dial plan. Fault tolerance and redundancy within VoIP networks are most important at the gatekeeper level. By using an alternate path you help provide redundancy and fault tolerance in the network.
                          Conformance to public standards: Different geographical locations might impose restrictions to your dial plan. Therefore, familiarize yourself with any such limitations prior to designing your dial plan.

                          Question #4

                          What is the minimum number of RSVP peers that are necessary for a Cisco Unified Border Element to provide Call Admission Control?

                          • A. 1
                          • B. 0
                          • C. 3
                          • D. 2
                          • E. 4
                          Answer: D
                          Question #5

                          What are the three acceptable values for one-way delay, jitter, and packet loss in a VoIP network? (Choose three.)

                          • A. 20 ms for jitter
                          • B. 0-150 ms for delay
                          • C. 0-400 ms for delay
                          • D. 1 packet loss
                          • E. 30 ms for jitter
                          • F. 1 percent packet loss
                          Answer: B,E,F

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