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| Section | Weight | Objectives |
|---|---|---|
| Topic 1: Describe a Dial Plan | 13% | - Dial plan components and functions - Digit manipulation and route patterns - Call routing and PSTN connectivity |
| Topic 2: Implement Cisco Unified Communications Manager Express | 18% | - IP phone registration and provisioning - CUCME setup and configuration - Basic call features and dial plans |
| Topic 3: Implement Voice Gateways | 22% | - Fax and modem support - Analog and digital voice ports - Dial peers configuration - Gateway types and deployment models |
| Topic 4: Implement Gatekeeper and Cisco Unified Border Element | 12% | - Interoperability and security - CUBE deployment and features - Gatekeeper functions and configuration |
| Topic 5: Describe VoIP Call Legs and Signaling | 20% | - H.323, SIP, and MGCP protocols - VoIP architecture and call flow - Codec selection and media transport |
| Topic 6: Implement QoS for Voice Networks | 15% | - Call Admission Control - Classification, marking, and policing - QoS principles and requirements - Queuing mechanisms and AutoQoS |
DRAG DROP
Drag the components from the left and drop them under the appropriate categories on the right.
Cisco 642-437 Exam
Correct Answer:

Explanation:
DSP delay, Packetization delay, Serialization delay & Dejitter Buffer delay are Fixed delay types. Queuing and Buffering delay & Network delay are Variable Delay types.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00800a8993.shtml
Fixed Delay DSP Delay Serialization Delay Dejitter Buffer Delay Processing Delay
Variable Delay Network Delay Queuing Delay
DRAG DROP
Drag the appropriate IOS commands from the left and drop them in the spaces on the right to create a dial peer that will match all inbound calls and prevent two-stage dialing on a T1 PRI circuit. Not all boxes are used and not all options are used.
Correct Answer:

Explanation:
Cisco 642-437 Exam
In the case of Digital Interfaces, when the PBX or central office (CO) switch sends a setup message that contains all the digits necessary to fully route the call, those digits can be mapped to an outbound Voice over IP (VoIP) dial-peer (or hairpin to plain old telephone service (POTS) dial-peer directly). The router/gateway does not present a secondary dial tone to the caller and does not collect digits. It forwards the call directly to the configured destination. In the case of analog interfaces, the user only hears the dial tone once (either local or remote), and then dials the digits and gets through to the destination phone. This is called one stage dialing. When one receives an inbound call from a POTS interface, the Direct Inward Dial (DID) feature in dial-peers enables the router/gateway to use the called number (dialed number identification service (DNIS)) to directly match an outbound dial-peer. When DID is configured on the inbound POTS dial-peer, the called number is automatically used to match the destination pattern for the outbound call leg. The incoming called number command will match the dial-peer that has the DID configured.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml
Which two statements are true regarding SCCP? (Choose two.)
Correct Answer: B,E 🗳️
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When marking traffic for Expedited Forwarding, how does a device that is not capable of recognizing DSCP interpret the EF binary string 101110?
Correct Answer: A 🗳️
Which three methodologies are specific types of QoS used on a 512-kb/s point-to-point IP WAN link? (Choose three.)
Correct Answer: B,C,E 🗳️
Viewing page 1 out of 52 pages.
Viewing questions 1-5 out of 257 questions.
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Implementing Cisco IP Switched Networks (300-115日本語版)
Implementing Cisco IP Routing
Troubleshooting and Maintaining Cisco IP Switched Networks (TSHOOT) (642-832日本語版)
Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
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