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Exam Number/Code: 642-845
Exam Name: Optimizing Converged Cisco Networks
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ONT – Optimizing Converged Cisco Networks
Exam Number: 642-845 Exam
Associated Certifications: ONT – Optimizing Converged Cisco Networks
Duration: 312 Q&As
Available Language(s): English
Exam Details
The Optimizing Converged Cisco Networks (642-845 ONT) is a qualifying exam for the Cisco Certified Network Professional CCNP®. The ONT 642-845 exam will certify that the successful candidate has important knowledge and skills in optimizing and providing effective QOS techniques for converged networks. The exam topics include implementing a VOIP network, implementing QoS on converged networks, specific IP QoS mechanisms for implementing the DiffServ QoS model, AutoQoS, wireless security and basic wireless management.

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QUESTION 1:
You need to implement QoS for the Certkiller VOIP network. Which three
statements are true about the data traffic characteristics of voice traffic? (Select
three)
A. Voice packets require TCP for rapid retransmission of dropped packets.
B. Latency is not a concern as long as jitter is kept below 30 ms.
C. Voice packets require a fairly constant bandwidth reserved for voice control traffic as
well as for the voice payload.
D. Voice packets do not require a specific type of queuing.
E. Latency must be kept below 150 ms.
F. Voice packets are rather small
Answer: C, E, F
Explanation:
QoS refers to the ability of a network to provide improved service to selected network
traffic over various underlying technologies including Frame Relay, ATM, Ethernet and
802.3 networks, SONET, and IP-routed networks.
QoS features provide improved and more predictable network service by offering the
following services:
1. Dedicated bandwidth
2. Improved loss characteristics
3. Congestion management and avoidance
4. Traffic shaping
5. Prioritization of traffic
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and
delay variation.
Loss causes voice clipping and skips. Industry standard codec algorithms can correct for
up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of
voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could
be lost at any given time. If two successive voice packets are lost, the 30 ms correctable
window is exceeded and voice quality begins to degrade.
Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice
delay becomes too long, the conversation sounds as if two parties are talking over a
satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms
one-way delay budget is acceptable for high voice quality. With respect to delay
variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can
usually compensate for 20 to 50 ms of jitter.
QUESTION 2:
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Certkiller uses G.711 for the VOIP calls. When analog signals are digitized using the
G.711 codec, voice samples are encapsulated into protocol data units (PDUs)
involving which three headers? (Select three)
A. UDP
B. RTP
C. IP
D. TCP
E. Compressed RTP
F. H.323
Answer: A, B, C
Explanation:
When a VoIP device, such as a gateway, sends voice over an IP network, the digitized
voice has to be encapsulated into an IP packet. Voice transmission requires features not
provided by the IP protocol header; therefore, additional transport protocols have to be
used. Transport protocols that include features required for voice transmission are TCP,
UDP, and RTP. VoIP utilizes a combination of UDP and RTP.
QUESTION 3:
VOIP has been rolled out to every Certkiller location. What are three features and
functions of voice (VOIP) traffic on a network? (Select three)
A. Voice traffic is bursty
B. Voice traffic is retransmittable
C. Voice traffic is time-sensitive
D. Voice traffic is bandwidth intensive
E. Voice traffic is constant
F. Voice traffic uses small packet sizes
Answer: C, E, F
Explanation:
The benefits of packet telephony networks include
i. More efficient use of bandwidth and equipment: Traditional telephony networks use
a 64-kbps channel for every voice call. Packet telephony shares bandwidth among
multiple logical connections.
ii. Lower transmission costs: A substantial amount of equipment is needed to combine
64-kbps channels into high-speed links for transport across the network. Packet
telephony statistically multiplexes voice traffic alongside data traffic. This consolidation
provides substantial savings on capital equipment and operations costs.
iii. Consolidated network expenses: Instead of operating separate networks for voice
and data, voice networks are converted to use the packet-switched architecture to create a
642-845
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single integrated communications network with a common switching and transmission
system. The benefit is significant cost savings on network equipment and operations.
iv. Improved employee productivity through features provided by IP telephony: IP
phones are not only phones, they are complete business communication devices. They
offer directory lookups and access to databases through Extensible Markup Language
(XML) applications. These applications allow simple integration of telephony into any
business application. For instance, employees can use the phone to look up information
about a customer who called in, search for inventory information, and enter orders. The
employee can be notified of a issue (for example, a change of the shipment date), and
with a single click can call the customer about the change. In addition, software-based
phones or wireless phones offer mobility to the phone user.
QUESTION 4:
Certkiller is rolling out an H.323 VOIP network using Cisco devices. Which IOS
feature provides dial plan scalability and bandwidth management for H.323 VoIP
implementations?
A. Digital Signal Processors
B. Call Routing
C. Gatekeeper
D. Call Admission Control
E. None of the above
Answer: C
Explanation:
Enterprise voice implementations use components such as gateways, gatekeepers, Cisco
Unified CallManager, and IP phones. Cisco Unified CallManager offers PBX-like
features to IP phones. Gateways interconnect traditional telephony systems, such as
analog or digital phones, PBXs, or the public switched telephone network (PSTN) to the
IP telephony solution. Gatekeepers can be used for scalability of dial plans and for
bandwidth management when using the H.323 protocol.
QUESTION 5:
A Cisco router is being used as a VOIP gateway to convert voice signals in the
Certkiller network. What steps are taken when a router converts a voice signal from
analog to digital form? (Select two)
A. Quantization
B. Serialization
C. Packetization
D. Sampling
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Answer: A, D
Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling
is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale
measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth
requirements. Analog-to-digital conversion is done by digital signal processors (DSPs),
which are located on the voice interface cards. The conversion is needed for calls
received on analog lines, which are then sent out to a packet network or to a digital voice
interface.
QUESTION 6:
You need to implement the proper IOS tools to ensure that VOIP works over the
Certkiller network. Which queuing and compression mechanisms are needed to
effectively use the available bandwidth for voice traffic? (Select two)
A. Priority Queuing (PQ) or Custom Queuing (CQ)
B. Real-Time Transport Protocol (RTP) header compression
C. Low Latency Queuing (LLQ)
D. Class-Based Weighted Fair Queuing (CBWFQ)
E. TCP header compression
F. UDP header compression
Answer: D, E
Explanation:
1. Class-based weighted fair queuing (CBWFQ) extends the standard WFQ functionality
to provide support for user-defined traffic classes. By using CBWFQ, network managers
can define traffic classes based on several match criteria, including protocols, access
control lists (ACLs), and input interfaces. A FIFO queue is reserved for each class, and
traffic belonging to a class is directed to the queue for that class. More than one IP flow,
or “conversation”, can belong to a class.
Once a class has been defined according to its match criteria, the characteristics can be
assigned to the class. To characterize a class, assign the bandwidth and maximum packet
limit. The bandwidth assigned to a class is the guaranteed bandwidth given to the class
during congestion.
CBWFQ assigns a weight to each configured class instead of each flow. This weight is
proportional to the bandwidth configured for each class. Weight is equal to the interface
bandwidth divided by the class bandwidth. Therefore, a class with a higher bandwidth
value will have a lower weight.
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By default, the total amount of bandwidth allocated for all classes must not exceed 75
percent of the available bandwidth on the interface. The other 25 percent is used for
control and routing traffic.
The queue limit must also be specified for the class. The specification is the maximum
number of packets allowed to accumulate in the queue for the class. Packets belonging to
a class are subject to the bandwidth and queue limits that are configured for the class.
2. TCP/IP header compression subscribes to the Van Jacobson Algorithm defined in RFC
1144. TCP/IP header compression lowers the overhead generated by the
disproportionately large TCP/IP headers as they are transmitted across the WAN. TCP/IP
header compression is protocol-specific and only compresses the TCP/IP header. The
Layer 2 header is still intact and a packet with a compressed TCP/IP header can still
travel across a WAN link.
TCP/IP header compression is beneficial on small packets with few bytes of data such as
Telnet. Cisco’s header compression supports Frame Relay and dial-on-demand WAN link
protocols. Because of processing overhead, header compression is generally used at
lower speeds, such as 64 kbps links.
QUESTION 7:
You want to ensure the highest call quality possible for all VOIP calls in the
Certkiller network. Which codec standard would provide the highest voice-quality,
mean opinion score (MOS)?
A. G.711, PCM
B. G.729, CS-ACELP
C. G.729A, CS-ACELP
D. G.728, LDCELP
E. None of the above
Answer: A
Explanation:
When a call is placed between two phones, the call setup stage occurs first. As a result of
this process, the call is logically set up, but no dedicated circuits (lines) are associated
with the call. The gateway then converts the received analog signals into digital format
using a codec, such as G.711 or G.729 if voice compression is being used.
When analog signals are digitized using the G.711 codec, 20 ms of voice consists of 160
samples, 8 bits each. The result is 160 bytes of voice information. These G.711 samples
(160 bytes) are encapsulated into an RTP header (12 bytes), a UDP header (8 bytes), and
an IP header (20 bytes). Therefore, the whole IP packet carrying UDP, RTP, and the
voice payload has a size of 200 bytes. When G.711 is being used, the ratio of header to
payload is smaller because of the larger voice payload. Forty bytes of headers are added
to 160 bytes of payload, so one-fourth of the G.711 codec bandwidth (64 kbps) has to be
added. Without Layer 2 overhead, a G.711 call requires 80 kbps.
642-845
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QUESTION 8:
When a router converts analog signals to digital signals as part of the VoIP process,
it performs four separate steps. From the options shown below, which set of steps
contains the steps in their correct sequence?
A. encoding
quantization
optional compression
sampling
B. optional compression
encoding
sampling
quantization
C. sampling
quantization
encoding
optional compression
D. optional compression
sampling
encoding
quantization
E. sampling
quantization
optional compression
encoding
F. encoding
optional compression
quantization
sampling
G. None of the above
Answer: C
Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling
is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale
measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression:
Optionally, voice samples can be compressed to reduce bandwidth requirements.
Analog-to-digital conversion is done by digital signal processors (DSPs), which are
located on the voice interface cards. The conversion is needed for calls received on
analog lines, which are then sent out to a packet network or to a digital voice interface.
642-845
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QUESTION 9:
Certkiller has determined that during its busiest hours, the average number of
internal VoIP calls across the WAN link is four (4). Since this is an average, the
WAN link has been sized for six (6) calls with no call admission control. What will
happen when a seventh call is attempted across the WAN link?
A. The seventh call is routed via the PSTN.
B. The call is completed, but all calls have quality issues.
C. The call is completed but the seventh call has quality issues.
D. The call is denied and the original six (6) calls remain.
E. The call is completed and the first call is dropped.
F. None of the above.
Answer: B
Explanation:
IP telephony solutions offer Call Admission Control (CAC), a feature that artificially
limits the number of concurrent voice calls to prevent oversubscription of WAN
resources.
Without CAC, if too many calls are active and too much voice traffic is sent, delays and
packet drops occur. Even giving Real-Time Transport Protocol (RTP) packets absolute
priority over all other traffic does not help when the physical bandwidth is not sufficient
to carry all voice packets. Quality of service (QoS) mechanisms do not associate
individual RTP packets with individual calls; therefore, all RTP packets are treated
equally. All RTP packets will experience delays, and any RTP packets may be dropped.
The effect of this behavior is that all voice calls experience voice quality degradation
when oversubscription occurs. It is a common misconception that only calls that are
beyond the bandwidth limit will suffer from quality degradation. CAC is the only method
that prevents general voice quality degradation caused by too many concurrent active
calls.
QUESTION 10:
While planning the new Certkiller VOIP network, you need to determine the size of
the WAN links to use. To do this, you need to calculate the bandwidth required by
each call. Which three pieces of information are used to calculate the total
bandwidth of a VoIP call? (Select three)
A. The serialization of the interface
B. The quantization
C. The TCP overhead
D. The packetization size
642-845
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E. The UDP overhead
F. The packet rate
Answer: D, E, F

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1. Which two QoS fields should be used by R1 and R2 to classify the traffic sent from PC1 to Server1? (Choose two.)

A. CoS

B. IP DSCP C. DE bit
D. priority bits

E. IP precedence

Answer: BE

2. Which classification tool can be used to classify traffic based on the HTTP URL? A. class-based policing
B. policy-based routing (PBR)

C. committed access rate (CAR)

D. network-based application recognition (NBAR) E. dial peers
Answer: D

3. Which two statements are true about the function of CAC? (Choose two.) A. CAC provides guaranteed voice quality on a link.
B. CAC artificially limits the number of concurrent voice calls.

C. CAC is used to control the amount of bandwidth that is taken by a call on a link.

D. CAC prevents oversubscription of WAN resources that is caused by too much voice traffic.

E. CAC allows an unlimited number of voice calls while severely restricting, if necessary, other forms of traffic.

F. CAC solves voice congestion problems by using QoS to give priority to UDP traffic. Answer: BD

4. Which two statements are true about the implementation of QoS? (Choose two.) A. Implementing DiffServ involves the configuration of RSVP.
B. Implementing IntServ allows QoS to be performed by configuring only the ingress and egress devices.

C. Implementing IntServ involves the utilization of RSVP.

D. Traffic should be classified and marked by the core network devices.

E. Traffic should be classified and marked as close to the edge of the network as possible. Answer: CE

5. What three types of interfaces do voice gateways use to connect to analog interfaces? (Choose three.)

A. E and M B. E1 CCS C. FXS
D. FXO E. serial
F. T1 CAS Answer: ACD

6. Which two Cisco router functional planes are protected by Control Plane Policing (CoPP)? (Choose two.)

A. data plane

B. control plane

C. service plane

D. management plane E. hyper plane Answer: AD

7. QoS preclassification is a term used to describe what Cisco IOS feature? A. AutoQoS
B. Modular QoS Command-Line Interface

C. QoS for VPNs D. AutoSecure Answer: C

8. For which two network problems would QoS be a good solution? (Choose two.) A. serialization delay
B. end-to-end delay

C. lack of bandwidth

D. inconsistent port costs

E. routing table convergence issues

F. poor file transfer rates

Answer: BC

9. What are two steps to define a QoS policy? (Choose two.) A. Set a minimum bandwidth guarantee.
B. Increase bandwidth.

C. Determine a specific transfer rate. D. Establish timers.
E. Configure CBWFQ for best-effort traffic. F. Set a maximum bandwidth limit.
Answer: AF

10. Which 802.1x Extensible Authentication Protocol (EAP) type supports authentication using digital certificates?
A. EAP-TLS B. EAP-FAST C. LEAP
D. WPA Answer: A

11. Which three configuration tasks are required to successfully deploy NBAR to recognize TCP and UDP stateful protocols? (Choose three.)
A. Over leased lines, use the multilink ppp command to reduce latency and jitter, and to create Distributed Link

Fragmentation and interleaving.

B. Use the service-policy command to attach a traffic flow to an interface on the router.

C. Use the ip rsvp bandwidth command to set a strict upper limit on the bandwidth NBAR uses, and to guarantee admission of any flows.
D. Use the policy-map command to define one or more QoS policies (such as shaping, policing, and so on) to apply to traffic defined by a class map.
E. Use the random-detect dscp command to modify the default minimum and maximum thresholds for the DSCP

value.

F. Use the class-map command to define one or more traffic classes by specifying the criteria by which traffic is classified.
Answer: BDF

12. Which codec standard would provide the highest voice-quality, mean opinion score (MOS)? A. G.711, PCM
B. G.728, LDCELP

C. G.729, CS-ACELP

D. G.729A, CS-ACELP Answer: A

13. Which two voice-gateway, analog-interface statements are true? (Choose two.)

A. A router can use a Foreign Exchange Office (FXO) interface to connect to a PSTN.

B. A router can use a Foreign Exchange Station (FXS) interface to connect to a PBX.

C. An analog fax machine can connect to a Foreign Exchange Office (FXO) interface. D. An analog telephone can connect to a Foreign Exchange Station (FXS) interface. Answer: AD

14. Refer to the exhibit. Both routers have been configured as VoIP gateways. They must also support traditional telephony devices to connect to analog telephones. Which two configuration changes would correctly support the voice requirements? (Choose two.)

A. On each router, under the dial-peer voice 1 pots configuration, add the port fa0/0 command.

B. On each router, under the dial-peer voice 1 pots configuration, add the port 1/0/0 command.

C. On each router, configure dial-peer voice 1 as a voip connection and configure dial-peer voice 2 as a pots connection.
D. Under the dial-peer voice 1 pots configuration, change the destination pattern of 1111 to 2222 on the R1

router, and 2222 to 1111 on the R2 router.

E. Under the dial-peer voice 2 voip configuration, change the destination pattern of 1111 to 2222 on the R1

router, and 2222 to 1111 on the R2 router.

F. Under the dial-peer voice 2 voip configuration, change the destination target address of 10.2.2.2 to 10.1.1.1 on the R1 router, and the destination target address of 10.1.1.1 to 10.2.2.2 on the R2 router.
Answer: BE

15. Drag each WLSE feature on the left to its benefit on the right.

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Exam Number: 642-845
Associated Certifications: CCNP
Duration: 90 minutes
Available Languages: English
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QUESTION 1
You need to implement QoS for the Certkiller VOIP network. Which three statements are true about the data traffic characteristics of voice traffic? (Select three)

A. Voice packets require TCP for rapid retransmission of dropped packets. B. Latency is not a concern as long as jitter is kept below 30 ms.
C. Voice packets require a fairly constant bandwidth reserved for voice control traffic as well as for the voice payload.
D. Voice packets do not require a specific type of queuing. E. Latency must be kept below 150 ms.
F. Voice packets are rather small
Answer: C, E, F Explanation:
QoS refers to the ability of a network to provide improved service to selected network traffic over various underlying technologies including Frame Relay, ATM, Ethernet and
802.3 networks, SONET, and IP-routed networks.
QoS features provide improved and more predictable network service by offering the following services:
1. Dedicated bandwidth
2. Improved loss characteristics
3. Congestion management and avoidance
4. Traffic shaping
5. Prioritization of traffic
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and delay variation.
Loss causes voice clipping and skips. Industry standard codec algorithms can correct for
up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could
be lost at any given time. If two successive voice packets are lost, the 30 ms correctable window is exceeded and voice quality begins to degrade.
Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes too long, the conversation sounds as if two parties are talking over a satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms
one-way delay budget is acceptable for high voice quality. With respect to delay
variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can usually compensate for 20 to 50 ms of jitter.

QUESTION 2
Certkiller uses G.711 for the VOIP calls. When analog signals are digitized using the
G.711 codec, voice samples are encapsulated into protocol data units (PDUs)
involving which three headers? (Select three)

TK

642-845

A. UDP
B. RTP C. IP
D. TCP
E. Compressed RTP F. H.323
Answer: A, B, C Explanation:
When a VoIP device, such as a gateway, sends voice over an IP network, the digitized voice has to be encapsulated into an IP packet. Voice transmission requires features not provided by the IP protocol header; therefore, additional transport protocols have to be used. Transport protocols that include features required for voice transmission are TCP, UDP, and RTP. VoIP utilizes a combination of UDP and RTP.

QUESTION 3
VOIP has been rolled out to every Certkiller location. What are three features and functions of voice (VOIP) traffic on a network? (Select three)

A. Voice traffic is bursty
B. Voice traffic is retransmittable
C. Voice traffic is time-sensitive
D. Voice traffic is bandwidth intensive
E. Voice traffic is constant
F. Voice traffic uses small packet sizes
Answer: C, E, F Explanation:
The benefits of packet telephony networks include
i. More efficient use of bandwidth and equipment: Traditional telephony networks use
a 64-kbps channel for every voice call. Packet telephony shares bandwidth among multiple logical connections.
ii. Lower transmission costs: A substantial amount of equipment is needed to combine
64-kbps channels into high-speed links for transport across the network. Packet
telephony statistically multiplexes voice traffic alongside data traffic. This consolidation provides substantial savings on capital equipment and operations costs.
iii. Consolidated network expenses: Instead of operating separate networks for voice
and data, voice networks are converted to use the packet-switched architecture to create a single integrated communications network with a common switching and transmission system. The benefit is significant cost savings on network equipment and operations.
iv. Improved employee productivity through features provided by IP telephony: IP phones are not only phones, they are complete business communication devices. They offer directory lookups and access to databases through Extensible Markup Language
(XML) applications. These applications allow simple integration of telephony into any

TK

642-845

business application. For instance, employees can use the phone to look up information
about a customer who called in, search for inventory information, and enter orders. The employee can be notified of a issue (for example, a change of the shipment date), and
with a single click can call the customer about the change. In addition, software-based phones or wireless phones offer mobility to the phone user.

QUESTION 4
Certkiller is rolling out an H.323 VOIP network using Cisco devices. Which IOS feature provides dial plan scalability and bandwidth management for H.323 VoIP implementations?

A. Digital Signal Processors
B. Call Routing
C. Gatekeeper
D. Call Admission Control
E. None of the above
Answer: C Explanation:
Enterprise voice implementations use components such as gateways, gatekeepers, Cisco
Unified CallManager, and IP phones. Cisco Unified CallManager offers PBX-like features to IP phones. Gateways interconnect traditional telephony systems, such as
analog or digital phones, PBXs, or the public switched telephone network (PSTN) to the
IP telephony solution. Gatekeepers can be used for scalability of dial plans and for bandwidth management when using the H.323 protocol.

QUESTION 5
A Cisco router is being used as a VOIP gateway to convert voice signals in the Certkiller network. What steps are taken when a router converts a voice signal from analog to digital form? (Select two)

A. Quantization B. Serialization C. Packetization D. Sampling
Answer: A, D Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling
is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth
Exam 642-845: Optimizing Converged Cisco Networks (ONT)

Related Certifications: CCNP

Duration: 90 minutes

Cost: $150, Available at Pearson VUE Testing Centers

Exam Topics Include:

1. Describe Cisco VoIP implementations.

2. Describe QoS considerations and strategies.

3. Describe DiffServ QoS implementations.

4. Implement Cisco Auto QoS.

5. Implement Cisco device hardening using Cisco security features

6. Describe and Implement WLAN security and management.

The Optimizing Converged Cisco Networks exam, 642-845, is a qualifying exam for the Cisco Certified Network Professional CCNP. This is one of four tests that are required in order to become CCNP certified. The ONT 642-845 exam will certify that the successful candidate has important knowledge and skills in optimizing and providing effective QOS techniques for unified networks. The exam topics include implementing a VOIP network, implementing QoS including AutoQoS, wireless security, and basic wireless management.

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Exam Number/Code: 642-845
Exam Name: ONT – Optimizing Converged Cisco Networks
VUE Code: 642-845
Questions Type: Multiple choice,

Exam : Cisco 642-­845

Title :

ONT Optimizing Converged Cisco

Update : Demo

1. What best describes an FXO interface?

A. analog trunks that provide the Survivable Remote Site Telephony (SRST) feature

B. analog trunks that provide VoIP gateway functionality

C. analog trunks that connect a gateway to plain old telephone service (POTS) device such as analog phones, fax machines, and legacy voice­mail systems
D. analog trunks that connect a gateway to a central office (CO) or private branch exchange (PBX) Answer: D

2. Which two steps are executed in the deployment of Cisco AutoQoS for Enterprise? (Choose two.)

A. The customer uses SNMP statistics to create the policy.

B. QoS policy templates are generated and installed on the interface. C. RTP is used to generate the policy.
D. LLQ, cRTP, and LFI are used to automatically discover the policy.

E. The auto­generated policy is manually optimized before implementation.

F. Auto­discovery is used to determine what traffic is on the interface. Answer: BF

3. Which two statements are true about the application of QoS in a converged network? (Choose two.)

A. End­to­end network delay times that exceed 50 ms for real­time traffic are considered unacceptable.

B. End­to­end network delay times that exceed 250 ms for real­time traffic are considered unacceptable.

C. End­to­end network delay is not a factor as long as the delay is consistent. D. Some packet loss can be corrected by codec algorithms.
E. RSVP handles voice packet retransmission.

F. Fragmentation is a result of packet loss. Answer: BD

4. What are two of the three predefined classes for the Cisco SDM wizard? (Choose two.)

A. business­critical

B. high priority

C. middle class

D. voice

E. best effort F. scavenger Answer: AE

5. A GRE tunnel is configured between a local and a remote site. Where should the service policy be applied to classify packets based on the pretunnel header?
A. In global configuration mode, apply the service policy and use the qos pre­classify command.

B. In global configuration mode, apply the service policy but do not use the qos pre­classify command.

C. Apply the service policy on the physical interface but do not use the qos pre­classify command.

D. Apply the service policy on the tunnel interface but do not use the qos pre­classify command.

E. Apply the service policy on the tunnel interface and use the qos pre­classify command. Answer: D

6. Which statement is true about the capabilities of a voice gateway router?

A. Voice gateways with analog interfaces need a gatekeeper to convert analog signals into digital format before voice is encapsulated into IP packets.
B. Voice gateways need a gatekeeper to interconnect the VoIP networks with the PSTN network.

C. Voice gateways with Cisco Unified CallManager Express (CME) can permanently act as a call agent for IP phones.
D. Voice gateways equipped with a DSP module can take the role of the call agent during WAN failure. Answer: C

7. Refer to the exhibit. What type of interface will connect the voice gateway router R1 to the analog phone and the fax machine?

A. The analog phone will be connected to the FXS port, and the fax machine will be connected to the FXO

port on router R1.

B. The analog phone will be connected to the FXO port, and the fax machine will be connected to the FXS

port on router R1.

C. Both devices will be connected to the FXS ports on router R1. D. Both devices will be connected to the FXO ports on router R1.
E. Both devices will be connected to the E and M ports on router R1.

Answer: C

8. Which statement is true about the configuration of AutoQoS on a Frame Relay interface?

A. AutoQoS can be configured from a different subinterface if the DLCI is already assigned to one subinterface.
B. Multilink PPP (MLP) over Frame Relay must be manually configured on low speed DLCIs.

C. For low­speed Frame Relay DLCIs with Frame Relay­to­ATM Interworking, the AutoQoS cannot be configured if a virtual template is already configured for the DLCI.
D. AutoQoS can be configured on a Frame Relay DLCI only if a map class is attached to the DLCI. Answer: C

9. What are two major sources of delay that can be managed by QoS in voice­enabled networks?

(Choose two.)

A. propagation delay

B. voice packet serialization delay

C. congested egress queues

D. header overhead

E. packets dropped because of CRC errors

Answer: CD

10. Which statement best describes the Cisco 2700 Location Appliance collection of location information for wireless devices?
A. All Wi­Fi devices on the WLAN send directly received signal strength indication (RSSI) information to the Cisco Wireless Location Appliance through SNMP.
B. An AP collects RSSI information which is forwarded to wireless controllers through LWAPP. Wireless controllers forward aggregated RSSI to the location appliance through SNMP.
C. The APs collect received signal strength indication (RSSI) information from all Wi­Fi devices and forward the information to the Cisco Wireless Location Appliance through the LWAPP.
D. The Cisco WLAN controllers bypass the received signal strength indication (RSSI) information from all

Wi­Fi devices to the Cisco Wireless Location Appliance through LWAPP. Answer: B

11. Which two statements about the DiffServ QoS model are true? (Choose two.)

A. DiffServ requires RSVP to set up a path through the network to accommodate the requested QoS.

B. Network traffic is identified by class, and the level of service is chosen for each class.

C. The DiffServ model relies on a fairly simple mechanism to provide QoS over a wide range of equipment.
D. RSVP enables the DiffServ model to provide end­to­end QoS.

E. The DiffServ model is more scalable than the IntServ model.

F. The flow­based approach of the DiffServ model is ideal for large scalable implementations such as the public Internet.
Answer: BE

12. What are two features of WLSE­Express wireless management? (Choose two.)

A. centralized management for autonomous access points

B. centralized management for lightweight access points

C. centralized management that requires an external AAA server

D. centralized management with integrated AAA server

E. integration with the Location Appliance to expand real time tracking to 1500 devices for 30 days

F. integration with the Location Appliance to expand real time tracking to 2500 devices for 30 days

Answer: AD

13. Which three statements are true about the application of QoS models? (Choose three.)

A. The DiffServ model requires applications to signal the network with QoS requirements.

B. The DiffServ model can be used to deliver QoS based upon IP precedence, or source and destination addresses.
C. The DiffServ model requires RSVP.

D. The best effort model is suitable for applications such as file transfer and e­mail

E. The IntServ model requires applications to signal the network with QoS requirements.

F. The IntServ model attempts to deliver a level of service based on the QoS specified by each packet

Answer: BDE

14. Refer to the exhibit. The serial link that connects the two offices has a bandwidth of 128 kbps. What could be done to set aside 64 kbps of bandwidth to transport VoIP traffic over the WAN link?

A. Apply the clock rate 64000 command on both serial interfaces connected to the WAN link.

B. Apply the bandwidth 64 command on both serial interfaces connected to the WAN link.

C. Apply the frame­relay voice­bandwidth 64000 command to enable the QoS Call Admission Control

feature on the serial link.

D. Apply the max­conn 64 command on a perial peer basis on both sides of the serial link.

E. Apply the ip bandwidth­percent eigrp 1 64 command on both sides of the serial link.

F. Apply the auto­cost reference­bandwidth 64 command on both sides of the serial link. Answer: C

15. Which three configuration tasks are required to successfully deploy NBAR to recognize TCP and UDP

stateful protocols? (Choose three.)

A. Over leased lines, use the multilink ppp command to reduce latency and jitter, and to create Distributed

Link Fragmentation and interleaving.

B. Use the service­policy command to attach a traffic flow to an interface on the router.

C. Use the ip rsvp bandwidth command to set a strict upper limit on the bandwidth NBAR uses, and to guarantee admission of any flows.
D. Use the policy­map command to define one or more QoS policies (such as shaping, policing, and so on)

to apply to traffic defined by a class map.

E. Use the random­detect dscp command to modify the default minimum and maximum thresholds for the

DSCP value.

F. Use the class­map command to define one or more traffic classes by specifying the criteria by which traffic is classified.
Answer: BDF

16. What is the most common cause of network congestion?

A. aggregation congestion occurring at the core layer where the different distribution layer devices feed traffic to the core­layer switches
B. speed mismatches when moving from a low­speed LAN environment to a higher speed WAN

environment

C. flapping Ethernet interfaces caused by misconfigured LAN interfaces

D. flapping serial interfaces caused by misconfigured WAN interfaces

E. speed mismatches when moving from a high­speed LAN environment to a lower speed WAN

environment

Answer: E

17. What are two components of the Cisco Autonomous WLAN solution? (Choose two.) A. WCS
B. WLSE C. WLC D. WDS E. LWAPP
Answer: BD

18. What are three security problems with Wi­Fi Protected Access (WPA)? (Choose three.)

A. WPA is based on the outdated IEEE 802.11i standard.

B. WPA uses RSN, which uses the same base encryption algorithm as RC4.

C. WPA requires a hardware upgrade that may not be supported by all vendors. D. WPA uses TKIP, which uses the same base encryption algorithm as WEP.
E. WPA is susceptible to a DoS attack when it receives two packets in quick succession with bad MICs,

forcing the AP to shut down the entire Basic Service Set (BSS) for one minute.

F. WPA is susceptible to a security weakness when preshared keys are used. Answer: DEF

19. Which three characteristics of the traffic flow are taken into consideration when the QoS­for­VPNs feature (QoS pre­classify) provides packet classification and applies appropriate QoS service on tunnel interfaces? (Choose three.)
A. IP precedence bits

B. DSCP bits

C. original port numbers

D. DE bits

E. source IP address

F. destination IP address

Answer: CEF

20. Which two statements about the Wireless Location Appliance are true? (Choose two.)

A. A Wireless Location Appliance acts as a server to one or more Cisco WCSs. It collects, stores, and passes on data from its associated Cisco WLAN controllers.
B. Before using the Web interface, the initial configuration of the Wireless Location appliance must be

done using the command­line interface (CLI).

C. The Cisco 2000, 2700, 4100, and 4400 are examples of Wireless Location Appliances.

D. The Wireless Location Appliance visually displays the location information of WLAN devices and forwards this information to third­party applications using the Simple Network Management Protocol
(SNMP).

E. The Wireless Location Appliance visually tracks up to 15,000 WLAN devices and can store this information for 90 days.
Answer: AB

21. Which two statements about the QoS policy generation phase of the Cisco SDM QoS wizard are true?

(Choose two.)

A. The SDM QoS wizard will create two real­time traffic classes to handle VoIP and voice signaling packets.
B. The SDM QoS wizard will create three real­time traffic classes to handle VoIP, voice signaling, and

video packets.

C. The SDM QoS wizard will create four real­time traffic classes to handle VoIP, voice signaling, video, and video streaming packets.
D. The SDM QoS wizard will create two business­critical traffic classes to handle transactional and

network management packets.

E. The SDM QoS wizard will create three business­critical traffic classes to handle transactional, network management, and routing packets.
F. The SDM QoS wizard will create four business­critical traffic classes to handle transactional, network

management, routing, and best­effort packets.

Answer: AE

22. What are three traffic descriptors typically used to categorize traffic into different classes? (Choose three.)
A. IP precedence

B. media type

C. DSCP

D. outgoing interface E. incoming interface F. DLCI
Answer: ACE

23. Which two statements about queuing mechanisms are true? (Choose two.) A. FIFO queuing is only appropriate for slower serial interfaces.
B. Weighted fair queuing is the default queuing mechanism used for all but slower than E1 rate interfaces.

C. Only one queuing mechanism type can be applied to an interface.

D. Weighted fair queuing does not require the configuration of access lists to classify traffic.

E. Flow­based weighted fair queuing provides for queues to be serviced in a round­robin fashion. Answer: CD

24. Which two statements are true about the implementation of QoS? (Choose two.) A. Implementing DiffServ involves the configuration of RSVP.
B. Implementing IntServ allows QoS to be performed by configuring only the ingress and egress devices.

C. Implementing IntServ involves the utilization of RSVP.

D. Traffic should be classified and marked by the core network devices.

E. Traffic should be classified and marked as close to the edge of the network as possible. Answer: CE

25. What type of services are provided by the Cisco IOS gateways in a VoIP network with Cisco

CallManger functionality?

A. Cisco IOS gateways provide services such as address translation and network access control for the devices on the network.
B. Cisco IOS gateways provide services such as bandwidth management and accounting.

C. Cisco IOS gateways provide media termination and signal translation between the public switched telephone network (PSTN) and IP networks.
D. Cisco IOS gateways provide zone management for all registered endpoints in the network. Answer: C

26. What accurately describes the usage of a Cisco AutoQoS command?

A. On Catalyst switches, the show auto discovery qos command is used to display the data collected during the Auto­Discovery phase.
B. On Catalyst switches, the show auto qos command is used to display packet statistics of all classes that are configured for all service policies.
C. On Cisco routers, the show mls qos maps command is used to verify the CoS­to­DSCP maps for egress packet queuing.
D. On Cisco routers, the show auto qos command is used to display the AutoQoS interface templates, policy maps, class maps, and ACLs.
Answer: D

27. Refer to the exhibit. A router has a 256 kbps Frame Relay circuit connected to interface serial 0/0. As

a large FTP packet is being placed into the hardware transmit queue of interface serial 0/0, a voice packet

is placed into the priority queue of that interface. How, and in what order, will the packets be transmitted?

A. The voice packet will be transmitted first, followed by the FTP packet in its entirety.

B. The voice packet will be transmitted first, followed by the fragmented FTP packet. C. The FTP packet will be fragmented and the voice packet will be interleaved.
D. The FTP packet will be transmitted first in its entirety, followed by the voice packet. Answer: D

28. Cisco AutoQoS takes the interface type and bandwidth into consideration when configuring which three QoS features? (Choose three.)
A. voice compression methodology

B. LLQ C. cRTP
D. routing protocol

E. LFI

F. trust boundary

Answer: BCE

29. Refer to the exhibit. Both routers have been configured as VoIP gateways. They must also support traditional telephony devices to connect to analog telephones. Which two configuration changes would

correctly support the voice requirements? (Choose two.)

A. On each router, under the dial­peer voice 1 pots configuration, add the port fa0/0 command.

B. On each router, under the dial­peer voice 1 pots configuration, add the port 1/0/0 command.

C. On each router, configure dial­peer voice 1 as a voip connection and configure dial­peer voice 2 as a pots connection.
D. Under the dial­peer voice 1 pots configuration, change the destination pattern of 1111 to 2222 on the

R1 router, and 2222 to 1111 on the R2 router.

E. Under the dial­peer voice 2 voip configuration, change the destination pattern of 1111 to 2222 on the

R1 router, and 2222 to 1111 on the R2 router.

F. Under the dial­peer voice 2 voip configuration, change the destination target address of 10.2.2.2 to

10.1.1.1 on the R1 router, and the destination target address of 10.1.1.1 to 10.2.2.2 on the R2 router. Answer: BE

30. What are three considerations when choosing the QoS model to deploy in a network? (Choose three.)

A. the routing protocols being utilized in the network

B. the applications utilizing the network

C. the traffic destinations

D. network addressing scheme

E. cost of implementation

F. the amount of the control needed of the resources

Answer: BEF

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“ONT – Optimizing Converged Cisco Networks”, also known as 642-845 exam, is a Cisco certification.
Preparing for the 642-845 exam? Searching 642-845 Test Questions, 642-845 Practice Exam, 642-845 Dumps?

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Questions and Answers : 217 Q&As
Updated: May 9th , 2008
Market Price: $129.99
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The Optimizing Converged Cisco Networks (642-845 ONT) is a qualifying exam for the Cisco Certified Network Professional CCNP®. The ONT 642-845 exam will certify that the successful candidate has important knowledge and skills in optimizing and providing effective QOS techniques for converged networks. The exam topics include implementing a VOIP network, implementing QoS on converged networks, specific IP QoS mechanisms for implementing the DiffServ QoS model, AutoQoS, wireless security and basic wireless management.

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Brent Stewart, Denise Donohue, Jay Swan

ISBN: 1-58705-327-6

As a final exam preparation tool, the four CCNP Quick Reference Sheets included in this value-priced bundle provide a concise review of all objectives on all four of the new CCNP exams (BSCI 642-901, BCMSN 642-812, ISCW 642-825, and ONT 642-845). These digital Short Cuts provide you with detailed, graphical-based information, highlighting only the key topics in cram-style format.

With these documents as your guide, you will review key concepts required to manage the routers and switches that form the network core, as well as edge applications that integrate voice, wireless, and security into the network. These fact-filled Quick Reference Sheets allow you to get all-important information at a glance, helping you to focus your study on areas of weakness and to enhance memory retention of essential exam concepts.

Table of Contents:

1.      CCNP BSCI Quick Reference Sheets

2.      CCNP BCMSN Quick Reference Sheets

3.      CCNP ONT Quick Reference Sheets

4.      CCNP ISCW Quick Reference Sheets

Brent Stewart, CCNP, CCDP, MCSE, is a network administrator for CommScope and a certified Cisco Systems instructor. He participated in the development of BSCI and has seperately developed trainingmaterial for ICND, BSCI, BCMSN, BCRAN, and CIT. Brent lives in Hickory, NC, with his wife, Karen and children, Benjamin, Kaitlyn, Madelyn, and William.

Denise Donohue, CCIE No. 9566, is manager of solutions engineering for ePlus Technology in Maryland. She is responsible for designing and implementing data and VoIP networks, supporting companies based in the National Capital region. Prior to this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and course director for Global Knowledge and did network consulting for many years. Her CCIE is in Routing and Switching. 

Jay Swan is a senior network engineer for the Southern Ute Indian Tribe Growth Fund in Ignacio, CO. Prior to this position, he was a Cisco instructor and course director for Global Knowledge. Jay has also worked in IT in the higher education and service provider fields. He holds CCNP® and CCSP® certifications.

Lab Portfolio

David Kotfila • Joshua Moorhouse • Ross G. Wolfson, CCIE® No. 16696

CCNP Optimizing Converged Networks (ONT 642-845) Lab Portfolio provides you with opportunities for hands-on practice with optimizing and providing effective QoS techniques in converged networks operating voice, wireless, and security applications. Topics also include implementing a VoIP network, specific mechanisms for implementing the DiffServ QoS model, AutoQoS, wireless security, and basic wireless management.

Those preparing for the Optimizing Converged Cisco Networks (ONT 642-845) certification exam should work through this book cover-to-cover. Or, if you need to quickly review configuration examples, you can go directly to the relevant chapter.

CCNP Optimizing Converged Networks (ONT 642-845) Lab Portfolio includes

21 Labs built to support v5 of the Optimizing Converged Networks course within the Cisco® Networking Academy® curriculum, providing ample opportunity to practice.
2 Challenge and Troubleshooting Labs have been added to the core curriculum labs to test your mastery of the topics.
1 Case Study to give you a taste of what is involved in a fully functioning network, covering all the technologies taught in this course. Even if you do not have the actual equipment to configure these more complex topologies, it is worth reading through these labs to expand your thinking into more complex networking solutions.
 

David Kotfila, CCNP®, CCAI, is the director of the Cisco Networking Academy at Rensselaer Polytechnic Institute (RPI), Troy, New York.

Joshua Moorhouse, CCNP, recently graduated from Rensselaer Polytechnic Institute with a bachelor of science degree in computer science, where he also worked as a teaching assistant in the Cisco Networking Academy. He currently works as a network engineer at Factset Research Systems.

Ross Wolfson, CCIE® No. 16696, recently graduated from Rensselaer Polytechnic Institute with a bachelor of science degree in computer science. He currently works as a network engineer at Factset Research Systems.

Use this Lab Portfolio with:

CCNP ONT Official Exam Certification Guide

ISBN-10: 1-58720-176-3

ISBN-13: 978-1-58720-176-9

CCNP ONT Portable Command Guide

ISBN-10: 1-58720-185-2

ISBN-13: 978-158720-185-1

This book is part of the Cisco Networking Academy Series from Cisco Press®. Books in this series support and complement the Cisco Networking Academy curriculum.