642-437 study materials - Cisco 642-437 dumps VCE

Cisco 642-437 : Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)

Exam Code: 642-437

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

Updated: Sep 22, 2026

Q & A: 257 Questions and Answers

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Cisco 642-437 Exam Syllabus Topics:

SectionWeightObjectives
Implement Gatekeeper and Cisco Unified Border Element12%- CUBE deployment and features
- Interoperability and security
- Gatekeeper functions and configuration
Describe VoIP Call Legs and Signaling20%- VoIP architecture and call flow
- Codec selection and media transport
- H.323, SIP, and MGCP protocols
Implement Voice Gateways22%- Analog and digital voice ports
- Gateway types and deployment models
- Fax and modem support
- Dial peers configuration
Implement QoS for Voice Networks15%- Classification, marking, and policing
- Call Admission Control
- QoS principles and requirements
- Queuing mechanisms and AutoQoS
Implement Cisco Unified Communications Manager Express18%- CUCME setup and configuration
- Basic call features and dial plans
- IP phone registration and provisioning
Describe a Dial Plan13%- Dial plan components and functions
- Call routing and PSTN connectivity
- Digit manipulation and route patterns

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

Question #1

Which of the following best describes the implementation challenges that are associated with variable-length numbering plans?

  • A. the identification of the number of digits that need to be dialed before the call is routed
  • B. the degree in which the dial plan varies
  • C. the number of trunks that need to be assigned
  • D. the mapping between IP addresses and extension numbers
  • E. the variable number of extensions that need to be implemented
Reveal Solution  Discussion  0

Correct Answer: A  🗳️

Question #2

When you configure Cisco Unified Border Element to support RSVP-based CAC, when are the RSVP path and reservation messages sent and received?

  • A. after the H.245 capabilities negotiation is completed
  • B. immediately after the call setup message is received and the reservation message is received, after H.245 capabilities negotiation is completed
  • C. immediately after the call setup message is received
  • D. after the global command call rsvp-sync is configured
Reveal Solution  Discussion  0

Correct Answer: C  🗳️

Question #3

DRAG DROP
Drag the statement from the left to the protocol name that is associated with it on the right.

Reveal Solution  Discussion  0

Correct Answer:


Explanation:

The Real-Time Transport Protocol (RTP) is an Internet protocol standard that specifies a way for programs to manage the real-time transmission of multimedia data over either unicast or multicast network services. RTP is commonly used in Internet telephony applications. RTP does not in itself guarantee real-time delivery of multimedia data; it does, however, provide the wherewithal to manage the data as it arrives to best effect. RTP combines its data transport with a control protocol (RTCP), which makes it possible to monitor data delivery for large multicast networks. When protocols are used in conjunction, RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. Monitoring allows the receiver to detect if there is any packet loss and to compensate. The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. Since RTP is closely related to RTCP (Real Time Control Protocol) which can be used to control the RTP session, SRTP also has a sister protocol, called Secure RTCP (or SRTCP); SRTCP provides the same security-related features to RTCP, as the ones provided by SRTP to RTP. Utilization of SRTP or SRTCP is optional to the utilization of RTP or RTCP; but even if SRTP/SRTCP are used, all provided features (such as encryption and authentication) are optional and can be separately enabled or disabled. The only exception is the message authentication feature which is indispensably required when using SRTCP. On slow links, it may be advantageous to compress the IP/UDP/RTP headers using Compressed RTP (cRTP). If you use cRTP then the 40 bytes of overhead incurred by the IP/UDP/RTP headers can typically be compressed down to 2 to 4 bytes (2 bytes when no UDP checksums are sent, and 4 bytes when checksums are sent). Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if it carries a lot of RTP traffic. cRTP is supported on serial lines using Frame Relay, HDLC, or PPP encapsulation. It is also supported over ISDN interfaces. CRTP should not be used on links greater than 2 Mbps.

Question #4

When a Session Initiation Protocol user agent client initiates a call to another user agent server, what is the first message type that is sent to the Cisco SIP Proxy Server?

  • A. initiate
  • B. invite
  • C. setup
  • D. trying
Reveal Solution  Discussion  0

Correct Answer: B  🗳️

Question #5

Which command should you use to configure a dial-peer to support T.38 fax relay and to use Cisco fax relay if T.38 fails?

  • A. fax relay t38 backup cisco relay
  • B. fax protocol t38 fallback cisco
  • C. fax relay t38 cisco
  • D. fax t38 fallback cisco
Reveal Solution  Discussion  0

Correct Answer: B  🗳️

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