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Braindumps of 642-845
ONT - Optimizing Converged Cisco Networks

Exam Questions, Answers, Braindumps (642-845)

Passed the exam yesterday. All the questions were from exams.ws study material. I got 92 % marks. That’s sufficient for me. I am satisfied with it.
I hereby submit some questions.

Q: 1
Analog interfaces are being utilized in a number of the Abc 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.
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
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.
Q: 2
Abc 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.
Q: 3
VOIP has been rolled out to every Abc 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 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.
Q: 4
Abc 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.
Q: 5
A Cisco router is being used as a VOIP gateway to convert voice signals in the Abc 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 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.
Q: 6
You need to implement the proper IOS tools to ensure that VOIP works over the Abc 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.
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.
Q: 7
You want to ensure the highest call quality possible for all VOIP calls in the Abc 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.
Q: 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.
Q: 9
Abc 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.

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