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Cisco Voice Over IP Exam(CVOICE) Number : 642-432 Passing Score : 800 Time Limit : 120 min File Version : 2.75 http://www.gratisexam.com/ CISCO 642-432 Cisco Voice Over IP Exam(CVOICE) 72 Q&A Version 2.75 Important Note, Please Read Carefully Other ExamWorx products A) Offline Testing engine Use the offline Testing engine product topractice the questions in an exam environment. Build a foundation of knowledge which will be useful also after passing the exam. Latest Version We are constantly reviewing our products. New material is added and old material is revised. Free updates are available for 90 days after the purchase. You should check your member zone at ExamWorx and update 3-4 days before the scheduled exam date. Here is the procedure to get the latest version: 1.Go towww.ExamWorx.com 2.Click on Log in 3.The latest versions of all purchased products are downloadable from here. Just click the links.
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Page 1: Cisco Voice Over IP Exam(CVOICE)...CISCO 642-432 Cisco Voice Over IP Exam(CVOICE) 72 Q&A Version 2.75 Important Note, Please Read Carefully Other ExamWorx products A) Offline Testing

Cisco Voice Over IP Exam(CVOICE)

Number: 642-432Passing Score: 800Time Limit: 120 minFile Version: 2.75

http://www.gratisexam.com/

CISCO 642-432

Cisco Voice Over IP Exam(CVOICE)

72 Q&A

Version 2.75

Important Note, Please Read Carefully

Other ExamWorx productsA) Offline Testing engineUse the offline Testing engine product topractice the questions in an exam environment.

Build a foundation of knowledge which will be useful also after passing the exam.

Latest VersionWe are constantly reviewing our products. New material is added and old material is revised. Free updates areavailable for 90 days after the purchase. You should check your member zone at ExamWorx and update 3-4days before the scheduled exam date.

Here is the procedure to get the latest version:

1.Go towww.ExamWorx.com2.Click on Log in3.The latest versions of all purchased products are downloadable from here. Just click the links.

Page 2: Cisco Voice Over IP Exam(CVOICE)...CISCO 642-432 Cisco Voice Over IP Exam(CVOICE) 72 Q&A Version 2.75 Important Note, Please Read Carefully Other ExamWorx products A) Offline Testing

For most updates,it is enough just to print the new questions at the end of the new version, not the wholedocument.

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Page 3: Cisco Voice Over IP Exam(CVOICE)...CISCO 642-432 Cisco Voice Over IP Exam(CVOICE) 72 Q&A Version 2.75 Important Note, Please Read Carefully Other ExamWorx products A) Offline Testing

Exam A

QUESTION 1You are working with a potential customer that would like to integrate its existing PBX telephone system into itsIP network. The accompanying figure shows that the customer has two offices that need to be connected to theIP network so that the customer can exchange telephone calls without using the PSTN. Both PBXs arecurrently connected to T1 ISDN circuits.

Which signaling type will allow you to support your customer?

A. QSIGB. CCSC. CASD. T-CCSE. E&MF. FXO

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

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QUESTION 2You are meeting with a customer that has deployed IP telephony at their headquarters location. They would liketo roll out IP telephony to their regional office as well. They are now using the G.711 codec at headquarters.They want to be able to maximize the number of calls carried without impacting voice quality or forcing a WANupgrade. Which codec would be appropriate for their WAN?

A. G.726B. G.723.1C. G.711D. G.729B

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 3Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is thecorrect diagnosis for the problem?

A. incorrect dial-peer statement in Router 1B. incorrect port statement in Router 1 pots dial peerC. incorrect session-target statement in Router 2D. incorrect destination-pattern in Router 1

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

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QUESTION 4You have been forwarded some questions by a prospective VoIP customer who would like to know the Ciscodefault sample size for the G.729 codec. What is it?

A. 40 msB. 30 msC. 20 msD. 10 ms

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 5Examine the example output.

http://www.gratisexam.com/

hostname GW1!interface Ethernet 0/0ip address 172.16.2.1 255.255.255.0h323-gateway voip interfaceh323-gateway voip id GK1-zone1.abc.com abc.com ipaddr 172.16.2.2h323-gateway voip h323-id GW1h323-gateway voip bind srcaddr 172.16.2.1!dial-peer voice 1 voipdestination-pattern 1212.

session-target ras!dial-peer voice 2 potsdestination-pattern 2125551212no register e164!end

Choose the command that will restore communication with gatekeeper functionality to this device.

A. h323-gateway voip h323-id GK1B. gatewayC. h323-gateway voip bind srcaddr 172.16.2.2D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1

Correct Answer: BSection: (none)Explanation

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Explanation/Reference:

QUESTION 6Which preference key word assigns top precedence to a dial peer in a hunt-group?

A. 0B. priorityC. 1D. high

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 7You are working with a potential customer that would like to integrate its existing PBX telephone system into itsIP network. The accompanying figure shows that the customer has two offices that need to be connected to theIP network so that the customer can exchange telephone calls without using the PSTN. Both PBXs use Wink-Start signaling.

Which signaling type will allow you to support the customer?

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A. QSIGB. CCSC. CASD. T-CCSE. E&MF. FXO

Correct Answer: ESection: (none)Explanation

Explanation/Reference:

QUESTION 8A 9 digit number must be dialed to reach numbers on the PSTN. What process makes sure that the first digit 9is not transmitted as part of the called number?

A. digit alternatingB. digit maskingC. digit manipulationD. digit seizing

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Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 9What is the E.164 numbering plan?

A. a proprietary PBX number planB. the IETF North American number planC. the European PBX standard telephony number planD. the ITU worldwide number plan

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 10Refer to the exhibit. What is the minimum WAN bandwidth required to support three simultaneous VoIP calls inthis network?

A. 19,200 bpsB. 51,600 bpsC. 79,200 bpsD. 247,200 bps

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 11In the connection between a Cisco router and an E&M port on a PBX, which side is generally the Cisco side?

A. loop startB. trunk circuitC. switch portD. signaling unit

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Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 12Which dial plan characteristic shows the most obvious improvement by dropping a number translation step?

A. availabilityB. post-dial delayC. scalabilityD. hierarchical design

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 13What are two basic parameters needed to setup a dial peer connected to the PSTN? (Choose two.)

A. voice portB. signaling typeC. interface bandwidthD. destination pattern

Correct Answer: ADSection: (none)Explanation

Explanation/Reference:

QUESTION 14Which device is used to allow an H.323 stream to transit a firewall?

A. gatekeeperB. gatewayC. proxyD. MCU

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 15You are working with your customer in their lab to test the effect of jitter on voice quality. You have set themaximum playout delay to 40 ms on the voice enabled routers. What will be the impact on voice quality if after

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severe congestion the playout buffer empties and the source sends packets to the buffer faster than they areleaving?

A. There will be no noticeable drop in quality.B. The jitter buffer will adapt to the faster-arriving packets by expanding the buffer size.C. The jitter buffer will speed up delivery of packets to the DSP so that packets are not dropped.D. After the jitter buffer fills up, subsequent packets are discarded.

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 16Refer to the exhibit. Your customer wants to converge voice and data on the existing T1 Frame Relay WAN linkbetween New York and Atlanta. The customer has allocated 25 percent of the WAN link for routing updates andother overhead. You are using 6 bytes of overhead for Frame Relay, no cRTP, and the G.729 codec.

How many calls could be placed on this link?

A. twoB. threeC. fourD. fiveE. sixF. seven

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 17In 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 withoutredundancy checks?

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A. 1 byteB. 2 bytesC. 3 bytesD. 4 bytesE. 20 bytesF. 40 bytes

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 18You have a customer that is interested in determining the number of VoIP calls their Frame Relay WAN linkscan support. Each of their Frame Relay WAN links has 84 kbps of bandwidth available outside all otherapplications and overhead. How many G.729 calls using the 8 kbps codec and 20 byte sample size can besupported?

A. 1B. 2C. 3D. 4

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 19You are working with a potential customer that would like to integrate its existing PBX telephone system into itsIP network. The accompanying figure shows that the customer has two offices that need to be connected to theIP network so that the customer can exchange telephone calls without using the PSTN. Both PBXs use an in-band signaling type.

Which signaling type will allow you to support your customer?

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A. QSIGB. CCSC. CASD. T-CCSE. E&MF. FXO

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 20A network has the following characteristics:

?the use of the G.711 codec with a codec speed of 64 kbps?a 160-byte sample size?the use of Frame Relay without compressed Real-Time Transport Protocol (CRTP) ?FRF.12 with 6 bytes ofoverhead

What minimum WAN bandwidth would be required to support three simultaneous VoIP calls?

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A. 247200 bpsB. 19200 bpsC. 79200 bpsD. 51600 bps

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 21At what point does the MGCP call agent release the setup of the call path to the residential gateways?

A. after the call agent has been notified that an event occurred at the source residential gatewayB. after the call agent has been notified of an event and has instructed the source residential gateway to create

a connectionC. does not release call path setupD. after the call agent has sent a connection request to both the source and destination and has relayed a

modify-connection request to the source so that the source and destination can set up the call pathE. after the call agent has forwarded session description protocol information to the destination from the

source and has sent a modify connection to the destination and a create-connection request to the source

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 22Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions ofthe four telephones attached. Which dial-peer statement will you use?

A. dial-peer voice 1 potsdestination-pattern 5552.[0-5]0

B. dial-peer voice 1 pots

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destination pattern 5552[5-6].0C. dial-peer voice 1 pots

destination-pattern 555[2-5][56]D. dial-peer voice 1 pots

destination-pattern 5552[5-6][05]0

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 23King's Mountain Cable has a pair of voice enabled routers capable of supporting only high complexity codecs.They need to conserve bandwidth on WAN links without major impact to call quality. Which two codecs satisfythese requirements? (Choose two.)

A. G.729B. G.729aC. G.729bD. G.729ab

Correct Answer: BDSection: (none)Explanation

Explanation/Reference:

QUESTION 24Which gateway interface connects to the standard station port of a PBX?

A. FXSB. E&MC. POTSD. FXO

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 25Your customer has forwarded this diagram and configuration. The customer wishes to have a connectionbetween its PBXs, a connection that is created and dropped as required. There is one configuration statementmissing from each router. What are the two missing statements? (Choose two.)

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A. connection trunk 2015551000404555?B. connection trunk 4045551200C. connection tie-line 4045551200D. connection tie-line 404555?E. connection tie-line 2015551000F. connection trunk

Correct Answer: CESection: (none)Explanation

Explanation/Reference:

QUESTION 26Refer to the exhibit. What is the minimum WAN bandwidth required to support three simultaneous VoIP calls inthis network?

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A. 19,200 bpsB. 51,600 bpsC. 79,200 bpsD. 247,200 bps

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 27Which process changes an internal extension into a fully qualified external PSTN number before matching to adial peer?

A. digit maskingB. forward digitsC. number expansionD. prefix extension

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 28Which type of signaling is DTMF?

A. supervisoryB. routeC. informationalD. address

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 29Which type of delay is caused when large data packets block voice packets from the outbound interface?

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A. queuing delayB. serialization delayC. propagation delayD. packetization delay

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 30Drop

A.B.C.D.

Correct Answer: Section: (none)Explanation

Explanation/Reference:

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QUESTION 31How is CAS different on E1 and T1?

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

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 32Which is the best way to achieve a scalable dial plan?

A. group numbers for a particular areaB. variable number of extension digitsC. single number prefixingD. hunt groups

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 33The illustration above shows the timing and order of arrival of voice packets at the end device. Each voicepacket contains 20 ms of voice data. Assuming a play-out buffer of 20 ms, what will happen to voice packetnumber 3?

A. It will be played out in its proper location.B. It will be played out in the order it is received.C. It will be dropped and treated as a lost packet.D. Immediately moved to the front of the play-out buffer.

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

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QUESTION 34Which statement is true about VoIP packet loss?

A. Lost packets are simply retransmitted.B. Even minimal packet loss causes echo.C. IP phones can reconstruct up to three consecutive lost packets.D. Codec algorithms can overcome minimal packet loss.

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 35Drop

A.B.C.D.

Correct Answer: Section: (none)

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Explanation

Explanation/Reference:

QUESTION 36What does a gateway router match to a dialed number when setting up a VoIP call?

A. IP routeB. destination patternC. call legD. session target

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 37Which option is true concerning the MGCP call agent?

A. acts only as a recorder of call detailsB. provides only call signaling and call setupC. manages all aspects of the call and voice streamD. monitors the quality of each call after setup

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 38You have set up a complex dial plan using translation rules. The following translation rule has been configured.What output would correspond to the test translation-rule command?

translation-rule 1rule 0 ^0.. 215550210rule 1 ^1.. 215550211rule 2 ^2.. 215550212rule 3 ^3.. 215550213rule 4 ^4.. 215550214rule 5 ^5.. 215550215rule 6 ^6.. 215550216rule 7 ^7.. 215550217rule 8 ^8.. 215550218rule 9 ^9.. 215550210

A. test translation-rule 1 512The replaced number: 21555021512

B. test translation-rule 1 555The replaced number: 55521555021

C. test translation-rule 1 617The replaced number: 61721555021

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D. test translation-rule 1 910The replaced number: 21555021910

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 39Which three are supervisory signals? (Choose three.)

A. busyB. on hookC. off hookD. call waitingE. ring

Correct Answer: BCESection: (none)Explanation

Explanation/Reference:

QUESTION 40You are working with a potential customer that would like to integrate its existing PBX telephone system into itsIP network. The accompanying figure shows that the customer has two offices that need to be connected to theIP network so that the customer can exchange telephone calls without using the PSTN. Both PBXs use aproprietary signaling type.

Which signaling type will allow you to support your customer?

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A. E&MB. CCSC. CASD. T-CCSE. FXOF. FXS

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 41LAB

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A.B.C.D.

Correct Answer: Section: (none)Explanation

Explanation/Reference:

QUESTION 42Examine 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 1/0/0signal immediate-startoperation 4-wiretype 2

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http://www.gratisexam.com/

B. voice-port 1/0/0signal delay-dialoperation 4-wiretype 1

C. voice port 1/0/0signal wink-startoperation 4-wiretype 3

D. voice port 1/0/0signal immediate-startoperation 4-wiretype 4

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 43Site A uses three-digit internal numbers and Site B uses four-digit internal numbers. All calls to the PSTN arerouted through Site B. What dial plan below best represents provision simplicity?

A. Translate all called numbers within Site A to four digits.B. Translate all called numbers within Site B to three digits.C. Translate all called numbers leaving Site A to ten digits.D. Translate all called numbers at either site to ten digits.

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 44In the VoIP network above, which protocol provides the necessary sequence numbers so voice packetsoriginating at R1 are played in the correct order at R5?

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A. UDPB. TCPC. RTCPD. RTPE. CRTP

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 45To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of deviceis Server B?

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A. proxyB. redirectC. registrarD. user agent clientE. user agent server

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 46Which problem does a-law and mu-law address?

A. quantization produces distortion at higher frequenciesB. encoding loses signal integrity at lower frequenciesC. quantization produces higher signal-to-noise ratios at smaller amplitudesD. encoding produces greater signal loss at higher amplitude

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 47Examine the output.

ccm-manager mgcp!mgcp 5036

!voice-port 1/0/0!voice-port 1/0/1!dial-peer voice 1 potsapplication MGCPAPPport 1/0/0!dial-peer voice 2 potsapplication MGCPAPPport 1/0/1!

Your customer has sent you their MGCP gateway configuration. They are unable to get the gateway tocommunicate with the call agent. What command needs to be inserted to resolve the problem?

A. ccm-manager mgcp 172.16.1.1B. mgcp call-agent 172.16.1.1C. application MGCPAPP 172.16.1.1D. mgcp 5036 172.16.1.1

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Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 48Refer to the exhibit. Enzo's Bikes manufactures high end bicycle frames.

Until recently they sold only to bicycle shops; however, now they are starting to sell to end users. They need away to add two additional sales staff and ensure that the senior sales technician always gets the first call. Drewis the senior sales technician. Bob is the newest sales technician. Bob's phone should always be the last onechosen for incoming sales calls, after Drew and James. Bob's phone should be chosen first only when Drewand James are busy on calls.

Select the correct dial-peer command set for Bob's phone.

A. dial-peer voice 3 potsdestination-pattern 5555110preference 2

B. dial-peer voice 3 potsdestination-pattern 5555110preference 0

C. dial-peer voice 3 potsdestination-pattern 5555110preference firstlast

D. dial-peer voice 3 potsdestination-pattern 5555110preference 3huntstop

E. dial-peer voice 3 potsdestination-pattern 5555110preference high

Correct Answer: ASection: (none)Explanation

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Explanation/Reference:

QUESTION 49What are the three components in an MGCP environment? (Choose three.)

A. gatewayB. gatekeeperC. endpointD. call agentE. proxy server

Correct Answer: ACDSection: (none)Explanation

Explanation/Reference:

QUESTION 50Given the SIP network shown in the diagram, identify which two actions are initiated by the UAS (user agentserver)? (Choose two.)

A. contacts the user when a SIP invitation is receivedB. originates the ACK method to indicate that it has received a response to an invitationC. originates the BYE method to initiate call terminationD. originates the INVITE method including a description of the session parametersE. returns a response on behalf of the user to the invitation originator

Correct Answer: AESection: (none)Explanation

Explanation/Reference:

QUESTION 51The following are the original dial peer configurations for routers R1 and R2:

R1:dial-peer voice 20 voipdestination-pattern 408.......

session target ipv4:192.168.2.254!

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R2:dial-peer voice 21 potsdestination-pattern 4085554321

port 1/0/1!

Which phones can call to the other?

A. Only Phone A can call Phone B.B. Only Phone B can call Phone A.C. Both phones can call each other.D. Neither phone can call the other.

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 52What is the function of a SIP location server?

A. resolves active endpoint addressesB. routes service requestsC. acquires active endpoint addressesD. resolves text addresses to IP addresses

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

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QUESTION 53How do a-law and mu-law handle issues with signal-to-noise ratios related to the measure of amplitude?

A. Use smaller step functions at lower amplitudes.B. Use smaller step functions at higher frequencies.C. Increase code points at lower amplitudes.D. Increase code points at higher frequencies.

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 54One voice packet is lost between Phone A and Phone B. What will be the result to the listener?

A. The call is terminated.B. The listener will experience a gap in the received audio stream.C. The listener will hear the audio normally. Packet loss concealment will make the loss inaudible.D. The listener will hear the audio out of order when the lost packet is retransmitted.

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 55What is the function of a SIP registrar server?

A. resolves active endpoint addressesB. routes service requestsC. acquires endpoint addressesD. resolves text addresses to IP addresses

Correct Answer: C

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Section: (none)Explanation

Explanation/Reference:

QUESTION 56You have a pair of voice enabled routers capable of supporting only medium complexity codecs. You need toconserve bandwidth on WAN links without major impact to call quality. Which two codecs will satisfy theserequirements? (Choose two.)

A. G.729B. G.729aC. G.729bD. G.729ab

Correct Answer: BDSection: (none)Explanation

Explanation/Reference:

QUESTION 57Refer to the exhibit. Select the recommended QoS configuration for the LAN segments of the network in thecampus and branch office.

A. Configure WRR with voice as highest priority.Use ACLs to classify voice traffic.Isolate voice traffic in its own VLAN .Configure access switches to trust traffic from IP phones.

B. Configure a PQ with WRR.Use ACLs to classify voice control traffic.Isolate voice traffic in its own VLAN.Configure access switches to trust traffic from IP phones.

C. Configure a PQ with WRR.Use ACLs to classify voice traffic.Isolate voice traffic in its own VLAN.Configure access switches to trust traffic from IP phones.

D. Configure a PQ with voice control as highest priority.Use ACLs to classify voice control traffic.Isolate voice control traffic in its own VLAN .Configure access switches to trust voice control traffic from IP phones.

Correct Answer: CSection: (none)

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Explanation

Explanation/Reference:

QUESTION 58What is the best description of an MGCP endpoint?

A. the interconnection between packet and traditional telephone networksB. any analog telephony device (PBX, switch, etc.)C. IP phonesD. the gatekeepers in a VoIP network

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 59Which two codecs are required on a Gateway to support CNG and VAD across a WAN link without majorimpact to call quality and bandwidth consumption? (Choose two.)

A. G.729B. G.729aC. G.729bD. G.729abE. G.723

Correct Answer: CDSection: (none)Explanation

Explanation/Reference:

QUESTION 60What is the approximate frequency range of human speech?

A. 20 Hz to 20,000 HzB. 40 Hz to 15,000 HzC. 200 Hz to 9000 HzD. 600 Hz to 5400 Hz

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 61In an effort to consume less bandwidth across the WAN, the decision was made at Acme Chemical Companyto change the voice packet size. They changed from two voice frames per packet to one voice frame perpacket. What effect did this have on Acme's voice traffic?

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A. Per call bandwidth consumption decreased and end-to-end delay increased.B. Per call bandwidth consumption increased and end-to-end delay decreased.C. Per call bandwidth consumption decreased and end-to-end delay decreased.D. Per call bandwidth consumption increased and end-to-end delay also increased.E. There was no effect on voice traffic.

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 62Refer to the exhibit.

Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 FrameRelay WAN link between New York and Atlanta. Currently the following list of applications are consuming nomore bandwidth than what is listed on this segment of the network.

T1 link 1536 kbps

e-mail 75 kbpsinternet 200 kbpsOracle 500 kbpsFTP 250 kbps

Total 1025 kbps

The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like toincrease the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overheadfor Frame Relay, no cRTP, and the use of the G.711 codec.

How many simultaneous calls could be placed on this link?

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A. 0 callsB. 1 callC. 2 callsD. no more than 5 callsE. no more than 10 callsF. no more than 20 calls

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 63Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The threeremote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks toconnect the PBXs today.

What kind of connection should be implemented to allow a two-way permanent connection between R1 andR4?

A. PLARB. trunkC. tie-lineD. answer-mode

Correct Answer: BSection: (none)Explanation

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Explanation/Reference:

QUESTION 64In North America, which E&M signaling type is used most often for geographically separated equipment?

A. Type IB. Type IIC. Type IIID. Type IVE. Type V

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 65Which statement is true about only out-of-band signaling?

A. A signaling bit is robbed from each frame.B. Signaling bits are sent in a special order in a dedicated signaling frame.C. All signaling is directly associated with its corresponding voice frame.D. All voice packets carry their own signaling.

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 66In the connection between a Cisco router and an E&M port on a PBX, which side is generally the Cisco side?

A. loop startB. trunk circuitC. switch portD. signaling unit

Correct Answer: DSection: (none)Explanation

Explanation/Reference:

QUESTION 67In T1 CAS, where are the signaling states and control features carried for Super Frame robbed-bit signaling?

A. 6th and 12th frameB. 6th, 12th, 18th, and 24th frame

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C. the first and seventeenth time slotD. the first and sixteenth time slot

Correct Answer: ASection: (none)Explanation

Explanation/Reference:

QUESTION 68A particular set of H.323 endpoints cannot resolve the IP address of destination endpoints. What is the H.323approved method to fix this problem?

A. Build route tables in gateways on the segment.B. Hard code all IP addresses.C. Introduce a gatekeeper on the segment.D. Choose another signaling protocol.

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 69In E1 CAS, where are framing and signaling states carried?

A. 6th and 12th frameB. 6th, 12th, 18th, and 24th frameC. the first and seventeenth time slotD. the first and sixteenth time slot

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 70Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The threeremote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks toconnect the PBXs today.

Which kind of connection should be implemented to allow calls to be placed from 201-555-1000 to 727-555-1000 so that when the call is completed, network resources are returned for other uses?

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A. PLARB. trunkC. tie-lineD. answer-mode

Correct Answer: CSection: (none)Explanation

Explanation/Reference:

QUESTION 71Which command parameter specifies that the router should not attempt to initiate a trunk connection but shouldwait for an incoming call before establishing the trunk?

A. codec clear-channelB. connection-trunk 404555 answer-modeC. voice-port 1/0:1D. ds0-group timeslots 1-23 type ext-sig

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

QUESTION 72Refer to the exhibit. Will these delays meet the G.114 recommendation for one-way delay?

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A. Yes, they will meet the recommendation for most user applications.B. No, they will not meet the recommendation for general network purposes.C. No, but they will be acceptable for time-sensitive transaction processing.D. Yes, they will meet the recommendation, provided the administrators understand the implications.

Correct Answer: BSection: (none)Explanation

Explanation/Reference:

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