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How are inbound and outbound call legs handled from the perspective of the source router?

A. Only the inbound call leg is established by the source router.
B. Only the outbound call leg is established by the source router.
C. The inbound call leg and outbouond call leg are matched to the same dial peer.
D. The outbound call leg is matched first. Then, once the source is known, an inbound call leg is established.
E. The inbound call leg is matched first. Then, once the destination is known, an outbound call leg is established.

Correct Answer: E Section: (none) Explanation

When placing a call from an IP Phone to another IP Phone, how is ringback generated??

A. CallManager generates an RTP streamto play ringback on the originated phone.
B. CallManager sends a command to the originating IP Phone to play ringback locally.
C. The originating IP Phone plays ringback locally until the RTP stream has been established.
D. The phone is connected to an audio file server that generates the inband ringback tones.

Correct Answer: B Section: (none) Explanation
QUESTION 42 Examine the output. ccm-manager mgcp ! mgcp 5036 ! voice-port 1/0/0 ! voice-port 1/0/1 ! dial-peer voice 1 pots application MGCPAPP port 1/0/0 ! dial-peer voice 2 ports application MGCPAPP port 1/0/1 !

Your customer has sent you their MGCP gateway configuration. They are unable to get the gateway to communicate with the call agent. What command needs to be inserted to resolve the problem?
A. ccm-manager mgcp
B. mgcpcall-agent
C. application MGCPAPP
D. mgcp5036

Correct Answer: B Section: (none) Explanation
QUESTION 43 The Cisco CallManager dial plan architecture is set up to handle two general types of calls. What are they? (Choose all that apply.)
A. External calls through a SAA Gateway
B. External calls through a PSTN gateway or to another Cisco CallManager cluster
C. Internal calls From the source router to the PBX-1
D. Internal calls to Cisco IP phones registered to the Cisco CallManager cluster itself-
E. Internal calls from the IP SoftPhone to the 7200 VXR2
F. External calls through the last downstream CallManager phone set.

Correct Answer: BD
Section: (none)


IP Telephony uses which protocol that does not accommodate re-transmission?

A. RIP (Routing Information Protocol)
B. IP (Internet Protocol)
C. RTP (real time protocol)
D. TCP (Transmission Control Protocol)

Correct Answer: C
Section: (none)


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

A. Gateway
B. Gatekeeper
C. Endpoint
D. Call agent
E. Proxy server

Correct Answer: ACD Section: (none) Explanation
Explanation: A typical MGCP gateway environment connects on one side with a public switched telephone network (PSTN), and on the other side with an IP network. Specialized call agent applications control the flow of media data across the distributed environment. Call agents determine the route that data follows as it flows through the system. Multiple call agents can control call processing and data transfer. These call agents use a separate protocol to synchronize with each other and to send coherent commands to modules under their control. Each call agent usually controls a set of gateway applications, including at least one media gateway. Media gateways convert media signals to an appropriate format depending on whether the signals are directed to a circuit switched network format or a packet switched network. Media gateways primarily perform audio signal translation functions in accordance with call agent commands. Note: Gateways connected to an SS7 controlled network must also include at least one signaling gateway for controlling SS7 signaling. The MGCP connection model consists of endpoints and connections. Endpoints represent physical or virtual sources through which data can flow (for example, PSTN ports on a media gateway). Call agents combine sets of endpoints under their control to create point-to-point or multipoint connections. Connections provide data paths for transferring and processing the data that flows through the gateway environment. In the MGCP model, call control intelligence resides in the call agents, not in the media gateways. In effect, the MGCP standard defines a master/slave relationship between call agents and media gateways, where gateways execute commands sent by the call agents. MGCP is a client-server protocol. The CA handles all aspects of setting up calls to and from endpoints. CAs or control servers provide the feature capabilities that a particular endpoint will be able to use. Endpoints connected to different CAs will likely have a different set of features they can use. Since all of the call control features are in the control server, each control server vendor decides which features are most important, and therefore different control server vendors differ in “essential features.” MGCP relies on a control server, or call agent (CA), to control call progression, tones to apply, and call characteristics. MGCP endpoints carry out instructions from the CA, which controls how calls proceed.
What is the most probable cause of jitter?

A. Variable delay
B. Dropped packets
C. Impedance mismatch
D. Excessive delay

Correct Answer: A Section: (none) Explanation
Jitter in Packet Voice Networks
Jitter is defined as a variation in the delay of received packets. At the sending side,
packets are sent in a continuous stream with the packets spaced evenly apart. Due to
network congestion, improper queuing, or configuration errors, this steady stream can
become lumpy, or the delay between each packet can vary instead of remaining constant.
This diagram illustrates how a steady stream of packets is handled.
When a router receives a Real-Time Protocol (RTP) audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer is also sometimes referred to as the de-jitter buffer.
QUESTION 47 Before voice and video can be placed on a network, it is necessary to ensure that adequate bandwidth exists for all required applications. To begin, the minimum bandwidth requirements for each major application (for example, the voice media streams, video streams, voice control protocols, and all data traffic) should be summed. This sum represents the minimum bandwidth requirement for any given link, and it should consume no more than what percentage of the total bandwidth available on that link?
A. 25%
B. 50%
C. 100%
D. 74%

Correct Answer: D
Section: (none)


There are six major steps for WAN deployment when preparing IP telephony. From the list below,

please select which of the following are valid pre deployment choices. (Choose all that apply.)
A. Choosing Wiring Closets carefully
B. Determining Voice Bandwidht Requirements
C. Assessing Results
D. Selecting the right handset for the IP SoftPhone
E. Analyzing Upgrade Requirements
F. Collecting Information on the Current WAN Environment

Correct Answer: BCEF Section: (none) Explanation
Users are complaining that they are unable to complete a call from 678-555-1212 to
770-555-1111 from Router 1 to Router 2.
Select the correct answer to resolve the problem.

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

Correct Answer: B
Section: (none)


Which lower layer protocol does the Real-Time Protocol (RTP) use?


Correct Answer: B Section: (none) Explanation Explanation/Reference:
QUESTION 51 What is the major advantage of designing and placing VoIP and Internet telephony in a clients organization?
A. It is cheap but you still need a PBX regardless
B. The PSTN is doomed to be EOL in 5 years and this is the replacement.
C. It avoids the tolls charged by ordinary telephone service
D. Even without QoS it is much clearer that PSTN technology.

Correct Answer: C Section: (none) Explanation
QUESTION 52 When they are booted, the Cisco Access Digital Trunk Gateway DT-24+, the Cisco Access Digital Trunk Gateway DE-30+, and the Catalyst 6000 digital gateway are provisioned with Cisco CallManager location information. When these gateways initialize, a list of Cisco CallManager’s, referred to as a ______________ is downloaded to the gateways.
A. Cisco IPSP group
B. Call managed Cisco redundancy group
C. IPNC redundancy group
D. Cisco CallManager redundancy group

Correct Answer: D Section: (none) Explanation
QUESTION 53 You are the network engineer at Certkiller .com. You to connect a Cisco voice gateway to a PBX or the PSTN via ISDN (PRI, QSIG, BRI). What are two attributes of the PBX/PSTN switch that must be known to understand which features to configure on the voice gateway to connect successfully to it? (Choose two)
A. Whether Q.921 or Q.931 is supported by the PBX/PSTN switch.
B. Whether Symmetric mode is supported by the PBX/PSTN switch.
C. Which PRI/BRI switch-type is supported by the PBX/PSTN switch.
D. Whether network or user side is supported by the PBX/PSTN switch.
E. Whether wink, delay dial, or immediate dial is supported by the PBX/PSTN switch.

Correct Answer: CD Section: (none) Explanation
QUESTION 54 What would Receiving an Alarm Indication Signal of Blue indicate on your T1 connection where your voice traffic is going over?
A. Blue means there is an alarm occurring in the building, it is part of your disaster plan.
B. Blue means there is an alarm occurring on the line downstream from the equipment that is connected to the port
C. There is no blue alarm, only red and yellow.
D. Blue means there is an alarm occurring on the line upstream from the equipment that is connected to the port

Correct Answer: D Section: (none) Explanation
QUESTION 55 You are meeting with a customer that has deployed IP telephony at their headquarters location. They would like to 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 WAN upgrade. Which codec would be appropriate for their WAN?
A. G.726
B. G.723.1
C. G.711
D. G.729B

Correct Answer: D Section: (none) Explanation
QUESTION 56 You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch office in Delaware. The branch office is using a 128 kbps Frame Relay link to connect to headquarters. You want to ensure good voice quality on this link. Which two QoS mechanisms should you implement on the Frame Relay interface? (Choose two.)
E. Fragmentation

Correct Answer: BD Section: (none) Explanation

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