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QUESTION 225
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) Explanation
QUESTION 226
From the list below, what protocol is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over Multicast or Unicast network services.
Actualtests.com – The Power of Knowing
642-436

A. CAM
B. IPTV
C. STP
D. RTP
E. DMVRP
F. PIM
G. IS-IS

Correct Answer: D Section: (none) Explanation
QUESTION 227
Which statement represents the definition of an MGCP endpoint?
A. The interconnection between packet and traditional telephone networks.
B. Any analog telephony device (PBX, switch, ect).
C. IP hones
D. The gatekeepers in a VoIP network.

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
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. MGCP assumes a connection model where the basic constructs are endpoints and connections. Endpoints are sources or sinks of data and could be physical or virtual. Examples of physical endpoints are:
*
An interface on a gateway that terminates a trunk connected to a PSTN switch (e.g., Class 5, Class 4,
etc.). A gateway that terminates trunks is called a trunk gateway.
*
An interface on a gateway that terminates analog POTS connection to a phone, key system, PBX, etc. A
gateway that terminates residential POTS lines (to phones) is called a residential gateway.
An example of a virtual endpoint is an audio source in an audio- content server. Creation of physical
endpoints requires hardware installation, while creation of virtual endpoints can be done by software.

QUESTION 228
What are the three components in an MGCP environment? (Choose three)
A. Gateway Actualtests.com – The Power of Knowing 642-436
B. Gatekeeper
C. Endpoint
D. Call agent
E. Proxy server

Correct Answer: ACD Section: (none) Explanation Explanation/Reference:
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.
QUESTION 229
With regard to MGCP, what is a call?
A. It is the path between two telephones. Actualtests.com – The Power of Knowing 642-436
B. It is the RTP sessions between the endpoints.
C. It is a connection between an endpoint and the call agent.
D. It is two or more endpoints sharing the same Call ID and the same media stream.

Correct Answer: D Section: (none) Explanation
QUESTION 230
You are the network engineer at Certkiller .com. You are deploying an IP telephony solution using MGCP.
The call agent expects the gateway to use UDP port 2427 but an application on the Certkiller network is
already using that port. You want to use port 4662 instead.
Which command would allow you to change the UDP port that the call agents and gateway communicate
on?

A. Router(config)# mgcp UDP 4662
B. Router(config)# mgcp gateway 4662
C. Router(config)# mgcp call-agent 4662
D. Router(config-dial-peer)#application MGCPAPP 4662
E. Router(config)# mgcp default-package gm-package 4662

Correct Answer: C Section: (none) Explanation QUESTION 231
You are the Voice engineer at Certkiller .com. Numerous Certkiller users complain that they are unable to
complete calls through the MGCP network. You want to verify the extent of the problem by reviewing a
count of the successful and unsuccessful control commands.
Which command should you use?

A. show mgcp
B. show mgcp count
C. show mgcp statistics
D. show call active voice
E. show call history voice

Correct Answer: C Section: (none) Explanation
QUESTION 232
You are the network engineer at Certkiller .com. You want to verify the registration of the gateway with the
call agent.
Which show command should you use?

A. show mgcp Actualtests.com – The Power of Knowing 642-436
B. show call agent
C. show gateway mgcp
D. show endpoint mgcp
E. show call active voice

Correct Answer: A Section: (none) Explanation
QUESTION 233
What identifies an MGCP endpoint?
A. A two part identifier that consists of thetelephone number and local name of the user.
B. A two part identifier that consists of thetelephone number and remote name of the user.
C. A two part identifier that consists of the domain name of the user and the IP address of the gateway.
D. A two part identifier that consists of the local name of the user and the domain name of the gateway.

Correct Answer: D Section: (none) Explanation
QUESTION 234
DRAG DROP Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes.

A.
B.
C.
D.

Correct Answer: Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436

Explanation:
“Server reports back to a UA with destination coordinates” is a function of the a Redirect Server (p. 6-94 of
CVoice version 4.1 class books). Reference pages 6-91 – 6-94 of CVoice version 4.1 class books.

QUESTION 235

For Scalability and ease of management, the decision has been made to centralize the location of all SIP
endpoints in servers.
When phone A wants to call Phone B. it asks Certkiller A how to find Phone B.
What kind of device is Certkiller A?

A. Proxy
B. Redirect
C. Registrar
D. User agent client
E. User agent server

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
SIP ServersSIP servers include:

1.
Proxy server-the proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client’s behalf. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.

Actualtests.com – The Power of Knowing 642-436

2.
Redirectserver-Providesthe client with information about the next hop or hops that a message should take and then the client contacts the next hop server or UAS directly.

3.
Registrar server-Processes requests from UACs for registration of their current location. Registrar servers are often co-located with a redirect or proxy server. Redirect server: A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client
QUESTION 236
What is the function of a SIP location server?
A. Resolves active endpoint addresses
B. Routes service requests
C. Acquires active endpoint addresses
D. Resolves text addresses to IP addresses

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: The correct answer should be “Resolves active endpoint addresses” based on the following from CVoice version 4.1 class books on pages 6-84 and 6-89. A Location Server is defined (on page 6-84) as: An abstraction of a service providing address resolution services to SIP proxy or redirect servers. A location server embodies mechanisms to resolve addresses. On page 6-89 a Registrar Server is described as a server that acquires addresses for the location server.
QUESTION 237

Given the SIP network shown in the diagram identify which three actions are initiated by the UAC (user agent client)? (Choose three)
A. Initiates a SIP requests.
B. Originated the BYE method to indicate call termination.
C. Originates the ACK method to indicate that it has receives a response to its invitation.
D. Contacts the user when a SIP invitation is receives.
E. Returns a response on behalf of the user to the invitation originator.

Correct Answer: ABC Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
Explanation:
1.4.4 SIP Invitation A successful SIP invitation consists of two requests, INVITE followed by ACK. The INVITE (Section 4.2.1) request asks the callee to join a particular conference or establish a two-party conversation. After the callee has agreed to participate in the call, the caller confirms that it has received that response by sending an ACK (Section 4.2.2) request. If the caller no longer wants to participate in the call, it sends a BYE request instead of an ACK.

QUESTION 238

Which characteristic is true about SIP protocol messages?
A. Binary
B. Text-based
C. Numeric
D. Encrypted

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
Format
All SIP messages are either requests from a server or client or responses to a request. The messages are
formatted according to RFC 822,
“Standard for the format of ARPA internet text messages.” For all messages, the general format is:

1.
A start line

2.
One or more header fields

Actualtests.com – The Power of Knowing 642-436

3.
An empty line
4.
A message body (optional)
Each line must end with a carriage return-line feed (CRLF).

QUESTION 239
Upon which protocol model is the SIP protocol based?
A. HTML
B. H.323
C. Q.931
D. MGCP
E. HTPP/WWW

Correct Answer: E Section: (none) Explanation
QUESTION 240
With regard to SIP and SDP, which of the following statements is true?
A. SIP is similar to RAS and SDP is similar to RTP
B. SIP is similar to RTP and SDP is similar to RAS
C. SIP is similar to H.225 and SDP is similar to H.245
D. SIP is similar to H.245 and SDP is similar to H.323
E. SIP is similar to H.323 and SDP is similar to H.225

Correct Answer: C Section: (none) Explanation
QUESTION 241
You are the network engineer at Certkiller .com. You are configuring a connection to a SIP proxy server. Which command would you use to specify the IP address of the server?
A. sip-ua sip-server ipv4:1.2.3.4
B. sip-ua sip-server target:1.2.3.4
C. dial-peer voice 1 voip session target sip:1.2.3.4
D. dial-peer voice 1 voip session target sip-server:1.2.3.4

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 242
Which of the following call control models are based on decentralized call control? (Choose two.)
A. SIP
B. CAS
C. H.323
D. Q.931
E. MGCP

Correct Answer: AC Section: (none) Explanation
QUESTION 243
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 244
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
Actualtests.com – The Power of Knowing 642-436
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 172.16.1.1
B. mgcp call-agent 172.16.1.1
C. application MGCPAPP 172.16.1.1
D. mgcp 5036 172.16.1.1

Correct Answer: B Section: (none) Explanation
QUESTION 245
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
request method initiates a SIP call setup.
What will your reply be?

A. ACK
B. INVITE
C. OPTIONS
D. REGISTER
E. DISCOVER

Correct Answer: B Section: (none) Explanation
QUESTION 246

hostname CK1 ! interface serial 0/0 ip address 172.16.1.1 255.255.255.248
!
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controller t1 framing esp clock source line linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 404555…. session-target ipv4:172.16.1.6 ! dial-peer voice 2 ports destination-pattern 201555…. port 1/0:1 hostname CK2 ! interface serial 0/0 ip address 172.16.1.6 255.255.255.248 ! controller t1 framing esp clock source line linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 201555…. session-target ipv4:172.16.1.1 ! dial-peer voice 2 ports destination-pattern 404555…. port 1/0:1 Use the figure above to answer this question. When extension 201-555-1000 dials 404-555-1200, how are digits manipulated in R1 so they are presented correctly at CK2 ?
A. When extension 201-555-1000 dials 404-555-1200, the digits 404-555 are stripped off prior to matching the outbound POTS dial peer.
B. When extension 202-555-1000 dials 404-555-1200, the digits 404-555 are stripped off by the connection trunk and CK2 receives only 1200.
C. When extension 201-555-1000 dials 404-555-1200, the outbound VoIP dial peer is matched and all digits are sent.
D. When extension 201-555 1000 dials 404-555-1200, CK1 collects the 1200 and Actualtests.com – The Power of Knowing 642-436 prepends the tie-line digits 404555. That number is matched to a VoIP dial peer and sent to the appropriate address.

Correct Answer: D Section: (none) Explanation
QUESTION 247
How 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: B Section: (none) Explanation
QUESTION 248
When impendence is mismatched in a two-wire to four-wire circuit, what is the common result?
A. glare
B. jitter
C. echo
D. clipping

Correct Answer: C Section: (none) Explanation
QUESTION 249
In the connection between a Cisco router and an E&M port on a PBX, which side is generally the Cisco side?
A. loop start
B. trunk circuit
C. switch port
D. signaling unit

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: Analog trunk circuits connect automated systems, such as a private branch exchange (PBX) and the network, such as a central office (CO). The most common form of analog trunking is the E&M interface. E&M Signaling is commonly refer to as “ear & mouth” or “recEive and transMit”, but its origin comes from the term earth and magnet. Earth represents electrical ground and magnet represents the electromagnet used to generate
Actualtests.com – The Power of Knowing 642-436
tone. E&M signaling defines a trunk circuit side and a signaling unit side for each connection similar to the data circuit-terminating equipment (DCE) and data terminal equipment (DTE) reference type. Usually the PBX is the trunk circuit side and the telco, CO, channel-bank, or Cisco voice enabled platform is the signaling unit side. Note:Cisco’s analog E&M interface functions as the signaling unit side, so it expects the other side to be a trunk circuit.
QUESTION 250
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which signal types are used by E&M.
What will your reply be?

A. wink start, delay start, and loop start
B. wink start, loop start, and immediate start
C. wink start, delay start, and immediate start
D. delay start, and loop start, and immediate start

Correct Answer: C Section: (none) Explanation
QUESTION 251

In an effort to consume less bandwidth across the WAN, the decision was made at Certkiller to change the voice packet size. They changed from two voice frames per packet to one voice frame per packet. What effect did this have on Certkiller ‘s voice traffic?
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: B Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 252
You have been forwarded some questions by a prospective VoIP customer who would like to know the
Cisco default sample size for the G.729 codec.
What is it?

A. 40 ms
B. 30 ms
C. 20 ms
D. 10 ms

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: Codec Sample Interval (ms) This is the sample interval at which the codec operates. For example, the
G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
QUESTION 253
What component can be used to compensate for jitter?
A. FIFO queuing
B. Ethernet hubs
C. DSP algorithms
D. Playout delay buffer
E. Transmission medium

Correct Answer: D Section: (none) Explanation
QUESTION 254
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch
office in Delaware. Users at headquarters must be able to call users at the branch office and users at the
branch office must be able to call headquarters.
How many dial peers must you configure to meet these requirements?

A. 1
B. 2
C. 3
D. 4
E. none Actualtests.com – The Power of Knowing 642-436

Correct Answer: D Section: (none) Explanation

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QUESTION 191
In 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 deploued without redundancy checks?
Actualtests.com – The Power of Knowing 642-436
A. 1 byte
B. 2 bytes
C. 3 bytes
D. 4 bytes
E. 20 bytes
F. 40 bytes

Correct Answer: B Section: (none) Explanation
QUESTION 192
What does the PBX use to determine the destination of a call?
A. An ISDN ANI packet
B. A blocked/permitted call list
C. An analysis of the dialled digits
D. Historic requests from the specific phone extension

Correct Answer: C Section: (none) Explanation
QUESTION 193
Which of the following are CS-ACELP coding schemes? (Choose two)
A. G.711
B. G.728
C. G.729
D. Q.931
E. G-729A

Correct Answer: CE Section: (none) Explanation
QUESTION 194
Which of the following is the worst-case compression delay for CD-ACELP?
A. 2.5 ms
B. 5 ms
C. 7.5ms
D. 10 ms
E. 20 ms

Correct Answer: E Section: (none) Explanation
QUESTION 195
Actualtests.com – The Power of Knowing 642-436
What type of connection is considered a call leg?
A. A digital connection
B. A virtual connection
C. A logical connection
D. A physical connection
E. A hardwired connection

Correct Answer: C Section: (none) Explanation
QUESTION 196
To which layer of the OSI model does Q.921 signaling equates to in ISDN?
A. Session
B. Network
C. Transport
D. Data-Link
E. Application

Correct Answer: D Section: (none) Explanation
QUESTION 197
Certkiller has a PBX at corporate HQ and one at a branch office. You to replace the PBX-to-PXB TDM trunk connection with IP connectivity. The PBXs use proprietary signalling method. The following is a partial configuration of the HQ router that connect to the PBX: controller t1 1/0 ds0-group 1 timeslots 1-24 type ext-sig dial-peer voice 1 voip destination-pattern 1001 session target ipv4:10.10.0.1 dial-peer voice 2 pots destination-pattern 2001 port 1/0:1 connection trunk 1001 Which command is missing from the above configuration?
A. transparent-ccs in the voice port configuration
B. signal wink-start in the controller t1 configuration
C. auto-cut-through in the pots dial peer configuration
D. codec clear-channel in the voip dial peer configuration

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 198
You are the network engineer at Certkiller .com. The Certkiller ISDN network has two PBX systems from
different manufactures.
Which protocol allows functionality between these two PBX systems?

A. QSIG
B. Q.921
C. Q.931
D. T-CCS

Correct Answer: A Section: (none) Explanation
QUESTION 199
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which application conveys fax using T.37 fax relay.
What will your reply be?

A. IVR
B. TCL
C. TIFF
D. SNMP
E. SMTP

Correct Answer: E Section: (none) Explanation
QUESTION 200
What will happen when a network link is oversubscribed?
A. The link goes down.
B. All voice calls suffer.
C. Voice packets are fragmented.
D. Excess voice calls are dropped.
E. Data packets are given priority.

Correct Answer: B Section: (none) Explanation
QUESTION 201
Certkiller sells managed IP Phone service to businesses in multi-tenant units. Certkiller has POPs in many
cities, so all of their dial peer patterns are based on 10 digit numbers. Users dial 9 for local calls, followed
by the 7 digital local number. The following dial peer has been configured in a New York POP:
dial-peer voice 595 pots

Actualtests.com – The Power of Knowing
642-436

destination-pattern 595
port 1/0:24
A user dials a local number, 9-638-4422.
What command must be configured in the gateway to allow the call to complete?

A. prefix 595
B. forward-digits 7
C. rule 1 9…….595…….

D. forward 9…….595…….

E. num-exp 9…….595…….
Correct Answer: E Section: (none) Explanation
QUESTION 202
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) Explanation
QUESTION 203
When placing a call from an IP Phone to another IP Phone, how is ringback generated??
A. CallManager generates an RTP stream to 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 204
Actualtests.com – The Power of Knowing
642-436
In the VoIP network above, which protocol provides the necessary sequence numbers so voice packets originating at CK1 are played in the correct order to CK5 ?
A. UDP
B. TCP
C. RTCP
D. RTP
E. CRTP

Correct Answer: D Section: (none) Explanation
QUESTION 205
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
Explanation/Reference:
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
Actualtests.com – The Power of Knowing 642-436
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 206
When an IP phone says “Configuration CM List”, what is it doing?
A. downloading a .cnf.xml file via TFTP
B. retrieving the OS79XX.txt files from TFTP
C. downloading the application load from the TFTP server
D. attempting to register with the first two CallManagers onits list of configure CallManagers

Correct Answer: A Section: (none) Explanation
QUESTION 207
Name two sensitivities that Voice traffic has that data traffic is not necessarily affected by.
A. TPI
B. RFI
C. Delay
D. EMI
E. Jitter
F. Noise

Correct Answer: CE Section: (none) Explanation
QUESTION 208
Actualtests.com – The Power of Knowing 642-436 Your customer would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. The following applications are consuming no more bandwidth than what is in the list on this segment of the network. T1 link 1536 Kbps e-mail 75 Kbps Internet 200 Kbps Oracle 500 Kbps FTP 250 Kbps Total 1025 Kbps The customer has allocated 25% of the WAN link for routing updated and other overhead. Assuming 6 bytes overhead for Frame Relay, no cRTP and using the

A. 729 codec, how many calls could be placed on this link?
B. 2 calls
C. 3 calls
D. 4 calls
E. 5 calls
F. 6 calls

Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Based upon a total bandwidth of 1536 Kbps and 1025 Kbps being used by other applications you can only have 4 calls not 5. The reason is that of the 1536 Kbps of bandwidth only 75% of it is available (or 1152 Kbps). 1152 minus 1025 leaves just 127 Kbps available for voice traffic. Assuming that you are using FRF.12, G.729 (stated in this scenario), and no cRTP (also stated in this scenario) then you will need approximately 28.14 Kbps per call with 5% overhead included (26.8 Kbps without overhead). 26.8 x 5 = 134 Kbps and 28.14 x 5 = 140.7 Kbps. Both exceed the 127 Kbps available for voice. To calculate the required bandwidth reference the “Voice Codec Bandwidth Calculator” available on Cisco’s web site (requires a CCO sign-on to access the calculator).
QUESTION 209
You have set up Call Admission Control for a customer between their headquarters and manufacturing facility over their Frame Relay WAN. You are using the
A. 726r16 codec with a 40 byte sample, CRTP without CRC, and 90 kbps configured as the maximum bandwidth for CAC to use. What will happen when 7 calls try to call the remote office?
B. All the calls will go through without any quality issues. Actualtests.com – The Power of Knowing 642-436
C. Only 4 calls will go through and the remainder will get a reorder tone.
D. Six calls will go through, and the seventh call will be placed on hold until bandwidth is available.
E. Three calls will cross the Frame Relay WAN link, and four will use the PSTN with AAR.

Correct Answer: B Section: (none) Explanation QUESTION 210
You have designed a complex dial plan using digit manipulation. Given the following snippet of your configuration file, what action would you expect to result when a call beginning with the digits “5501” is received? dial-peer voice 1 pots destination-pattern 5501… … prefix port 1/0/0
A. A nine digit number beginning with 5501 will be forwarded.
B. A ten digit number beginning with 5501 will be forwarded.
C. A nine digit number beginning with 5501612 will be forwarded.
D. A ten digit number beginning with 5501612 will be forwarded.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: Destination Pattern The destination pattern associates a dialed string with a specific telephony device. It is configured in a dial peer by using the destination-pattern command. If the dialed string matches the destination pattern, the call is routed according to the voice port in POTS dial peers, or the session target in voice-network dial peers. For outbound voice-network dial peers, the destination pattern may also determine the dialed digits that the router collects and then forwards to the remote telephony interface, such as a PBX, a telephone, or the PSTN. You must configure a destination pattern for each POTS and voice-network dial peer that you define on the router. The destination pattern can be either a complete telephone number or a partial telephone number with wildcard digits, represented by a period (.) character. Each “.” represents a wildcard for an individual digit that the originating router expects to match. For example, if the destination pattern for a dial peer is defined as “555….”, then any dialed string beginning with 555, plus at least four additional digits, matches this dial peer.
QUESTION 211
What transport layer protocol does RTP utilize?
A. TCP Actualtests.com – The Power of Knowing 642-436
B. UDP
C. IP
D. ICMP

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: RTP typically runs on top of UDP to utilize its multiplexing and checksum services. Other transport protocols besides UDP can carry RTP as well. Real-Time Transport Protocol, an Internet protocol for transmitting real-time data such as audio and video. RTP itself does not guarantee real-time delivery of data, but it does provide mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols.
QUESTION 212
You are the network technician at Certkiller .com. VoIP is implemented on the Certkiller network. Your newly appointed Certkiller trainee wants to know what is used to carry VoIP voice packets on this network. What will your reply be?
A. ICMP/IP
B. RTP/TCP
C. RTP/UDP
D. STP/UDP
E. RTP/RCMP

Correct Answer: C Section: (none) Explanation
QUESTION 213
Which lower layer protocol does the Real-Time Protocol (RTP) use?
A. TCP
B. UDP
C. WDP
D. HTTP
E. RTCP

Correct Answer: B Section: (none) Explanation
QUESTION 214
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what TCP’s reliable deliver service provides.
Actualtests.com – The Power of Knowing 642-436
What will your reply be?
A. Connectionless service, flow control, sequenced delivery, and automatic error recovery
B. Flow control, sequenced delivery, automatic error recovery, and transmission window management
C. Unregulated send rate, automatic error recovery, and transmission window management
D. Connectionless service, unregulated send rate, automatic error recovery, and transmission window management

Correct Answer: B Section: (none) Explanation
QUESTION 215
You are the Voice technician at Certkiller , Inc. You want to deploy an IP telephony solution for the
company. The Certkiller network is currently a traditional LAN/WAN based on Frame Relay.
Your CEO has read about the issues of converging both data and voice traffic onto a single network. She
is concerned about the quality of their calls that need to cross the WAN in particularly.
What would you need to implement to ensure QoS for VoIP over Frame Relay?

A. Traffic shaping, priority queuing, Call Admission Control, and Class Based Weighted Fair Queuing
B. Traffic shaping, priority queuing, Call Admission Control, and Weighted Random Early Detection
C. Fragmentation, traffic shaping, priority queuing, Low Latency Queuing, and link efficiency with cRTP.
D. Fragmentation, traffic shaping, priority queuing, Call Admission Control, and Weighted Random Early Detection

Correct Answer: C Section: (none) Explanation
QUESTION 216
On what is system capacity planning based?
A. On calculations and measurements of packet length distributions.
B. On calculations and measurements of busy hour call volume/estimates.
C. On calculations and measurements of the phone costs from phone bills.
D. On calculations and measurements of the total number of calls placed during a month.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 217
You have a customer that is interested in determining the number of VoIP calls their Frame Relay WAN links can support. Each of their Frame Relay WAN links has 54 kbps of bandwidth available outside all other applications and overhead. How many G.726 calls using the 32 kbps codec and 80 byte sample size can be supported?
A. 1
B. 2
C. 3
D. 4

Correct Answer: A Section: (none) Explanation
QUESTION 218
You are the network engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which functions use UDP as their transport mechanism.
What will your reply be? (Choose two)

A. RTP
B. RAS control function
C. call signaling function
D. H.245 control function

Correct Answer: AB Section: (none) Explanation
QUESTION 219
What does gateway require to function as a translating gateway?
A. The capacity to translate the audio.
B. The ability to recognize the call control procedures of both connecting endpoints.
C. The ability to establish separate RTP sessions with the originating and terminating endpoints.
D. The ability to recognize the call control procedures for at least one of the connecting endpoints.

Correct Answer: B Section: (none) Explanation
QUESTION 220
You are the Voice engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
compressed RTP does.
What will your reply be?

Actualtests.com – The Power of Knowing
642-436

A. It significantly reduce packet delay
B. It significantly reduce total bandwidth
C. It significantly reduce Frame Relay overhead
D. It significantly reduce the total number of packets

Correct Answer: B Section: (none) Explanation
QUESTION 221
You are the network engineer at Certkiller .com. You are implementing Frame Relay traffic shaping on the
Certkiller network. Your newly appointed Certkiller trainee wants to know why Frame Relay traffic shaping
is important.
What will your reply be?

A. It ensures that excess traffic above the CIR on the link is dropped.
B. It ensures that voice packets are not trapped behind large data packets.
C. It ensures that the priority of the voice packet is higher than the data packets.
D. It ensures that the RTP headed is reduced in size to reduce the overall size of the voice packet.
E. It ensures that excess traffic above the CIR on the link is not dropped, but is buffered and sent when there is capacity on the link.

Correct Answer: E Section: (none) Explanation
QUESTION 222
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.)

A. CIR
B. LLQ
C. WFQ
D. WRED
E. Fragmentation

Correct Answer: BE Section: (none) Explanation
QUESTION 223
You are the Voice technician at Certkiller .com. The Certkiller network uses RTCP. Your newly appointed Certkiller trainee wants to know what RTCP does.
Actualtests.com – The Power of Knowing 642-436
What will your reply be?
A. It provides independent services irrespective of RTP.
B. It provides compression techniques to save bandwidth.
C. It provides in-band control information for an RTP flow.
D. It provides out-of-band control information for an RTP flow.

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: RTCP provides out-of-band control information for an RTP flow.
QUESTION 224
Which statement is true about the MGCP call agent?
A. Acts only as a recorder of call details.
B. Provides only call signaling and call setup.
C. Manages all aspects of the call and voice stream.
D. Monitors the quality of each call after setup.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
In the MGCP model, the gateways focus on the audio signal translation function, while the Call Agent
handles the signaling and call processing functions.

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