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642-437 Exam

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C C CE E E R R R T T T M M M A A A G G G I I I C C C Demo Edition Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Exam: 642-437 642-437 1 http://www.certmagic.com
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CCCEEERRRTTT MMMAAAGGGIIICCC

Demo Edition

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

Exam: 642-437

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Section 1: Sec One (1 to 20) Details: Topic 1, Main QUESTION: 1 DRAG DROP Your boss, Miss Certmagic, is interested in T.38 and Cisco fax relay. Match the appropriate descriptions with the correct categories.

Answer:

QUESTION: 2 Voice packets are arriving at a destination with a variance of between 20 and 50 ms. If the jitter buffer has a capacity of 30 ms, what is the impact on the audio at the receiving IP phone? A. The IP phone will negotiate, in mid-call, a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps. B. The audio stream at the receiving IP phone will be delayed and garbled. C. The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 msto avoid speech gaps. D. The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps. Answer: B

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QUESTION: 3 A small office needs to provide outbound dialing and inbound DID without the cost of a T1 circuit. All signaling is loop-start. Which analog port configuration will support these requirements? A. voice-port 0/1/0 signal did loop-start ! " dial-peer voice 1 pots incoming called-number ! dial-peer voice 90 pot destination-pattern 9T port 0/1/0 B. voice-port 0/0/0 signal loop-start i voice-port 0/1/0 signal loop-start i dial-peer voice 1 pots incoming called-numberT direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0 C. voice-port 0/0/0/ description fxs-did signal did loop-start i voice-port 0/1/0 description fxo signal loop-start i dial-peer voice 1 pots incoming called-number. direct-inward-dial port 0/0/0 i dial-peer voice 90 pots destination-pattern 9T port 0/1/0 D. voice-port 0/0/0

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signal did loop-start ! " dial-peer voice 1 pots incoming called-number direct-inward-dial i dial-peer voice 90 pots destination-pattern 9T port 0/0/0 Answer: B QUESTION: 4 When configuring Cisco AutoQoS VoIP on a Cisco catalyst switch, how is the configuration performed? A. Each interface will need either the auto qos voip trust cisco-phone or auto qos voip trust trust on each interface, depending on the upstream device. B. Each interface will need either the auto qos voip cisco-phone or auto qos voip trust on each interface, depending on the upstream device C. The auto qos voip command is applied globally in the switch. D. The auto qos voip command is applied to each interface. Answer: B QUESTION: 5 What is the function of class-based marking? A. Marking frames only sets information in the Layer 2 headers of a frame so that the frame can be identified and distinguished from other packets or frames. B. Marking packets based on CoS value, IP precedence value, or DSCP value allows Layer 3 frames to be identified and distinguished from other packets. C. Marking frames based only on CoS value or IP precedence value allows Layer 2 frames to be identified and distinguished from other frames. D. Marking packets or frames sets information in the Layer 2 and Layer 3 headers of a packet so that the packet or frame can be identified and distinguished from other packets or frames. Answer: B

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QUESTION: 6 Exhibit:

You work as a network engineer for Certmagic.com. You study the exhibit carefully. When 5551234 is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the PSTN? A. 51234 B. 1234 C. 555 D. 5551234 E. Null F. 5 Answer: A QUESTION: 7 Which two statements are true regarding SCCP? (Choose two.) A. Endpoints and gateways maintain the dial plan. B. SCCP uses hex messages for communication. C. SCCP may interoperate with H.323 endpoints if it is implemented with Cisco Unified Communications Manager. D. SCCP requires each endpoint or gateway event to be communicated to Cisco Unified Communications Manager E. Endpoints can operate autonomously if communication with Cisco Unified Communications Manager is lost. Answer: C, D

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QUESTION: 8 DRAG DROP Your boss, Miss Certmagic, is interested in VoIP dial peer. In particular, she is interested in the process of when the voice gateway selects an inbound VoIP dial peer by matching the information elements in the setup message with the dial-peer attributes. Select the appropriate elements and put them into the correct order in how a voice gateway matches inbound calls.

Answer:

QUESTION: 9 An access layer switch is configured to extend priority to an IP phone. Cisco Discovery Protocol is enabled on all ports. What are the three possible ways that an IP phone can be instructed to treat the Layer 2 CoS priority value of the attached PC? (Choose three.) A. untrusted

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B. trusted C. trusted IEEE 802.1Q D. configured IEEE 802.1Q E. configured CoS level F. configured DSCP level Answer: B, C, E QUESTION: 10 Exhibit:

You work as a network engineer for Certmagic.com. You study the exhibit carefully. Cisco Unified Communications Manager Express has been partially configured to support 6 IP phones and 12 directory numbers. The Cisco Unified Communications Manager Express will use the IP address 10.1.130.1/24. Which two elements of the configuration are missing from the command output and need to be added so that phones do not auto-register, but can manually register with Cisco Unified Communications Manager Express? (Choose two.) A. no reg-ephone B. create cnf-files C. ip address 10.1.130.1 D. create profile E. no auto-reg-ephone F. ip source-address 10.1.130.1 Answer: E, F QUESTION: 11 DRAG DROP

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You are required to create a dial peer that will match all inbound calls and prevent two-stage dialing on a T1 PRI circuit.

Answer:

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QUESTION: 12 Exhibit:

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You work as a network engineer for Certmagic.com. You study the exhibit carefully. Which class is always present even though it is not in the configuration snip? A. class class-scavenger B. best-effort class C. class best-effort D. class class-default E. default class Answer: D QUESTION: 13 Which two functions are associated with a voice gateway? (Choose two.) A. provides voice-messaging services to connected analog and digital voice circuits B. switches voice channels between connected analog and digital voice circuits

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C. controls opening and closing of logical channels that are used to carry media streams D. interconnects two logically separate VoIP networks E. negotiates endpoint capabilities Answer: B, C QUESTION: 14 DRAG DROP Your boss, Miss Certmagic, is interested in fixed and variable delays. Match the appropriate descriptions with the correct categories.

Answer:

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Explanation: Packetization delay, Serialization delay & Dejitter Buffer delay are Fixed delay types. DSP delay, Queuing and Buffering delay & Network delay are Variable Delay types. Variable delay A) Queuing/Buffering Delay Queuing delay (ßn) is a variable delay and is dependent on the trunk speed and the state of the queue. There are random elements associated with the queuing delay. B) Network Switching Delay The public frame relay or ATM network that interconnects the endpoint locations is the source of the largest delays for voice connections. Network Switching Delays (?n) are also the most difficult to quantify. C) DSP Short for digital signal processing, which refers to manipulating analog information, such as sound or photographs that has been converted into a digital form. DSP also implies the use of a data compression technique. Fixed Delay A)De-Jitter Delay Because speech is a constant bit-rate service, the jitter from all the variable delays must be removed before the signal leaves the network. In Cisco router/gateways this is accomplished with a de-jitter ( n) buffer at the far-end (receiving) router/gateway. The de-jitter buffer transforms the variable delay into a fixed delay. B)Packetization Delay Packetization delay (?n) is the time taken to fill a packet payload with encoded/compressed speech. This delay is a function of the sample block size required by the vocoder and the number of blocks placed in a single frame. Packetization delay can also be called Accumulation delay, as the voice samples accumulate in a buffer before they are released. C) Serialization Delay Serialization delay (?n) is the fixed delay required to clock a voice or data frame onto the network interface. It is directly related to the clock rate on the trunk. At low clock speeds and small frame sizes, the extra flag needed to separate frames is significant. QUESTION: 15 Exhibit:

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You work as a network engineer for Certmagic.com. You study the exhibit carefully. Which type of ephone-dn is configured for the two ephones that are shown? A. shared-line, nonexclusive B. shared-line, overlay C. hunt line D. two directory numbers with one telephone number E. single-line-octo F. Octo-line Answer: B QUESTION: 16 Exhibit:

Network Topology exhibit:

You work as a network engineer for Certmagic.com. You study the exhibits carefully. An administrator is migrating a PBX telephony system to an IP Phone solution using a fixed numbering plan. The extension numbers and PSTN DIDs cannot be changed. Which of the following methods can be used in order to reach the individual extensions at Site B when called via the PSTN? A. The administrator needs to map the last four digits in the DID to the extension numbers and prefix an intersite code. B. No changes are necessary because PSTN calls are preceded with access code 9 C. The administrator needs to map the last four digits in the DID to the extension numbers using translation rules. D. The administrator can add a 1 to the DID for Site B to become 300-555-31xxx.

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E. The administrator needs to map the last four digits in the DID to the extension numbers and prefix a site code. Answer: C QUESTION: 17 A new Cisco 7965 IP phone is installed on a Cisco Unified Communications Manager Express system. When the phone requests the .loads file from the TFTP server, it sees that the versions are different. What does the IP phone do to resolve this issue? A. The IP phone does not boot up and will require manual intervention to factory reset the phone before a new firmware image can be downloaded. B. The IP phone boot requests the XMLDefault.cnf.xml file and boots up. C. The IP phone attempts to obtain the new firmware file image from the TFTP server. D. The IP phone requests the SEP<mac>.cfg file and reboots. Answer: C QUESTION: 18 You are configuring a network to support voice to the PSTN. One important aspect to the configuration is to be able to determine the individual slot, subunit, and port number from the gateway endpoint identifier. Which signaling protocol is appropriate for this situation? A. H.323 B. MGCP C. SIP D. SCCP Answer: B QUESTION: 19 When a Cisco 7965 IP phone downloads an SCCP firmware file package, which file is downloaded first by the IP phone from the TFTP server to describe the files it should request from the TFTP server? A. SCCP45.9-0-2SR1S.loads

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B. cvm45sccp.8-4-0-79.sbn C. term65.default.loads D. term45.default.loads E. dsp45.8-4-0-79.sbn F. dsp45.8-4-0-79.sbn G. cnu45.8-4-0-79.sbn H. apps45.9-0-2ES2.sbn I. jar45sccp.8-4-0-79.sbn Answer: D QUESTION: 20 Which three methodologies are specific types of QoS used on a 512-kb/s point-to-point IP WAN link? (Choose three.) A. RSVP B. header compression C. LLQ D. Multilink PPP E. FRF.12 Answer: B, ?, ?

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