Question 1.
You need to implement QoS for the ITCertKeys 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.
ITCertKeys 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 ITCertKeys 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.

Question 4.
ITCertKeys 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 ITCertKeys 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.

Question 6.
You need to implement the proper IOS tools to ensure that VOIP works over the ITCertKeys 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.

Question 7.
You want to ensure the highest call quality possible for all VOIP calls in the ITCertKeys 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.

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.

Question 9.
ITCertKeys 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 ITCertKeys 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
E. The UDP overhead
F. The packet rate

Answer: D, E, F

Explanation:
Packet rate: Packet rate specifies the number of packets sent in a certain time interval. The packet rate is usually specified in packets per second (pps). Packet rate is the multiplicative inverse of the packetization period. The packetization period is the amount of voice (time) that will be encapsulated per packet, and is usually specified in milliseconds. Packetization size: Packetization size specifies the number of bytes that are needed to represent the voice information that will be encapsulated per packet. Packetization size depends on the packetization period and the bandwidth of the codec used. IP overhead: IP overhead specifies the number of bytes added to the voice information during IP encapsulation. When voice is encapsulated into Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and IP, the IP overhead is the sum of all these headers.

Data link overhead: Data-link overhead specifies the number of bytes added during data-link encapsulation. The data-link overhead depends on the used data-link protocol, which can be different per link. Tunneling overhead: Tunneling overhead specifies the number of bytes added by any security or tunneling protocol, such as 802.1Q tunneling, IPsec, Generic Route Encapsulation (GRE), or Multiprotocol Label Switching (MPLS). This overhead must be considered on all links between the tunnel source and the tunnel destination.

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