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642-845 : Optimizing Converged Cisco Networks Last Updated Thursday, September 18, 2008 with 270 Questions
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
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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.
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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.
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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
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E. The UDP overhead
F. The packet rate
Answer: D, E, F
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