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ONT – Optimizing Converged Cisco Networks : 642-845 Exam

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

Exam Description Introduction
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.

Exam Topics
The following information provides general guidelines for the content likely to be included on the exam. However, other related topics may also appear on any specific delivery of the exam. In order to better reflect the contents of the exam and for clarity purposes the guidelines below may change at any time without notice.

Describe Cisco VoIP implementations.
Describe the functions and operations of a VoIP network (e.g., packetization, bandwidth considerations, CAC, etc.).
Describe and identify basic voice components in an enterprise network (e.g. Gatekeepers, Gateways, etc.)

Describe QoS considerations.
Explain the necessity of QoS in converged networks (e.g., bandwidth, delay, loss, etc.).
Describe strategies for QoS implementations (e.g. QoS Policy, QoS Models, etc.).

Describe DiffServ QoS implementations.
Describe classification and marking (e.g., CoS, ToS, IP Precedence, DSCP, etc.).
Describe and configure NBAR for classification.
Explain congestion management and avoidance mechanisms (e.g., FIFO, PQ, WRR, WRED, etc.).
Describe traffic policing and traffic shaping (i.e., traffic conditioners).
Describe Control Plane Policing.
Describe WAN link efficiency mechanisms (e.g., Payload/Header Compression, MLP with interleaving, etc.).
Describe and configure QoS Pre-Classify.

Implement AutoQoS.
Explain the functions and operations of AutoQoS.
Describe the SDM QoS Wizard.
Configure, verify, and torubleshoot AutoQoS implementations (i.e., MQC).

Implement WLAN security and management.
Describe and Configure wireless security on Cisco Clients and APs (e.g., SSID, WEP, LEAP, etc.).
Describe basic wireless management (e.g., WLSE and WCS). Configure and verify basic WCS configuration (i.e., login, add/review controller/AP status, security, and import/review maps).
Describe and configure WLAN QoS.

“ONT – Optimizing Converged Cisco Networks”, also known as 642-845 exam, is a Cisco certification.
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With the complete collection of questions and answers Q&as with Expert Explanations, Pass4sure has assembled to take you through 217 Q&As to your 642-845 Exam preparation. In the 642-845 exam resources, you will cover every field and category in CCNP helping to ready you for your successful Cisco Certification.

QUESTION 16:
You want to ensure the highest level of audio quality in the PassGuide VOIP
network. Which two statements are true about the digital audio in a VoIP network?
(Select two)
A. Standard encoding techniques create an uncompressed digital data rate of 8000 bps.
B. Two methods of compression are u-law and a-law
C. Standard encoding techniques create an uncompressed digital data rate of 64,000 bps.
D. Two methods of quantization are linear and logarithmic.
E. Standard encoding techniques create an uncompressed digital data rate of 4000 bps.
F. Voice quality is not a concern if compression is not used.
Answer: C, D
Explanation:
Each sample is encoded in the following way:
One polarity bit: Indicates positive versus negative signals
Three segment bits: Identify the logarithmically sized segment number (0-7)
Four step bits: Identify the linear step within a segment.
Because 8000 samples per second are taken for telephony, the bandwidth that is needed
per call is 64 kbps. This is the reason why traditional, circuit-based telephony networks
use time-division-multiplexed lines, combining multiple channels of 64 kbps each
(digital signal level 0 [DS-0]) in a single physical
QUESTION 17:
Call Admission Control is being used on the PassGuide VOIP WAN. Which two
statements are true about the function of CAC? (Select two)
A. CAC solves voice congestion problems by using QoS to give priority to UDP traffic.
B. CAC prevents oversubscription of WAN resources that is caused by too much voice
traffic.
C. CAC artificially limits the number of concurrent voice calls.
D. CAC provides guaranteed voice quality on a link.
E. CAC is used to control the amount of bandwidth that is taken by a call on a link.
F. CAC allows an unlimited number of voice calls while severely restricting, if
necessary, other forms of traffic.
Answer: B, C
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
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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 18:
You need to calculate the bandwidth required to support VOIP on one of the remote
PassGuide locations. What is the minimum bandwidth required to support a single
uncompressed telephony call at the standard sampling rate and sample size?
A. 16 kbps
B. 80 kbps
C. 64 kbps
D. 96 kbps
E. 48 kbps
F. 32 kbps
G. None of the above
Answer: B
Explanation:
Assume a G.711 (Uncompressed) VoIP codec at the default packetization rate (50 pps).
A new VoIP packet is generated every 20 ms (1 second / 50 pps). The payload of each
VoIP packet is 160 bytes; with the IP, UDP, and RTP headers (20 + 8 + 12 bytes,
respectively) included, this packet become 200 bytes in length. Converting bits to bytes
requires multiplying by 8 and yields 1600 bps per packet. When multiplied by the total
number of packets per second (50 pps), this arrives at the Layer 3 bandwidth requirement
for uncompressed G.711 VoIP: 80 kbps. This example calculation corresponds to the first
row of Table 2-1.
Table 2-1 Voice Bandwidth (Without Layer 2 Overhead)
Bandwidth
Consumption
Packetization
Interval
Voice Payload
in Bytes
Packets Per
Second
Bandwidth Per
Conversation
G.711 20 ms 160 50 80 kbps
G.711 30 ms 240 33 74 kbps
G.729A 20 ms 20 50 24 kbps
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G.729A 30 ms 30 33 19 kbps
Reference: http://www.informit.com/articles/article.aspx?p=357102
QUESTION 19:
PassGuide uses FXO interfaces on their VOIP gateways. 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 voice-mail systems
D. Analog trunks that connect a gateway to a central office (CO) or private branch
exchange (PBX)
E. None of the above.
Answer: D
Explanation:
Gateways use different types of interfaces to connect to analog devices, such as phones,
fax machines, or PBX or public switched telephone network (PSTN) switches. Analog
interfaces used at the gateways include these three types:
FXS: The FXS interface connects to analog end systems, such as analog phones or
analog faxes, which on their side use the FXO interface. The router FXS interface
behaves like a PSTN or a PBX, serving phones, answering machines, or fax machines
with line power, ring voltage, and dial tones. If a PBX uses an FXO interface, it can also
connect to a router FXS interface. In this case, the PBX acts like a phone.
FXO: The FXO interface connects to analog systems, such as a PSTN or a PBX, which
on their side use the FXS interface. The router FXO interface behaves like a phone,
getting line power, ring voltage, and dial tones from the other side. As mentioned, a PBX
can also use an FXO interface toward the router (which will then use an FXS interface),
if the PBX takes the role of the phone.
E&M: The E&M interface provides signaling for analog trunks. Analog trunks
interconnect two PBX-style devices, such as any combination of a gateway (acting as a
PBX), a PBX, and a PSTN switch. E&M is often defined to as “ear and mouth,” but it
derives from the term “earth and magneto.” “Earth” represents the electrical ground, and
“magneto” represents the electromagnet used to generate tones.
QUESTION 20:
Analog interfaces are being utilized in a number of the PassGuide VOIP gateways.
Which two voice gateway analog-interface statements are true? (Select two)
A. An analog fax machine can connect to a Foreign Exchange Office (FXO) interface.
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B. A router can use a Foreign Exchange Office (FXO) interface to connect to a PSTN.
C. A router can use a Foreign Exchange Station (FXS) interface to connect to a PBX.
D. An analog telephone can connect to a Foreign Exchange Station (FXS) interface.
Answer: B, D