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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

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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?

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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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