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Exam A
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
Correct Answer: CEF Section: (none) Explanation
Explanation/Reference:
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 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
Correct Answer: ABC Section: (none) Explanation
Explanation/Reference:
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
Correct Answer: CEF Section: (none) Explanation
Explanation/Reference:
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
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
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
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
Correct Answer: AD Section: (none) Explanation
Explanation/Reference:
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
Correct Answer: DE Section: (none) Explanation
Explanation/Reference:
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 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
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
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
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
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
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.
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
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
E. The UDP overhead
F. The packet rate
Correct Answer: DEF Section: (none) Explanation
Explanation/Reference:
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.
QUESTION 11
Certkiller uses the distributed call processing model in their VOIP network. Which statement is true about the distributed call control in a VoIP network?
A. The VoIP endpoints have the intelligence to set up and control calls.
B. Call setup and control resides in call agents that are distributed throughout the network.
C. Call setup and control functionality is centralized in one call agent or cluster.
D. Each VoIP device has separate call control, voice packetization, and transport mechanisms.
E. None of the above.
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: Distributed call control is possible where the voice-capable device is configured to support call control directly. This is the case when protocols such as H.323 or SIP are enabled on the end devices. With distributed call control, the devices perform the call setup, call maintenance, and call teardown on their own. With distributed call control, each gateway makes its own, autonomous decisions and does not depend on the availability of another (centralized) device to provide call routing services to the gateway. Because each gateway has its own intelligence, there is no single point of failure. However, each gateway needs to have a local call routing table, which has to be configured manually. Therefore, administration of the distributed call control model is less scalable.
QUESTION 12
A Certkiller branch office has 15 IP phones connected to the main office using the distributed call processing model. Normally, the phones work with great quality. However, during very busy times when all of the agents are on the phone at the same time, the voice quality drops. Words, phrases, or both are
dropped from conversations. What is the most likely cause of the problem?
A. Employees are watching videos over the Internet.
B. Header compression is not being used.
C. Call Admission Control has not been implemented.
D. More bandwidth is required for the office LAN.
E. IP phone traffic is not being classified correctly.
F. Large files are being downloaded over the WAN network.
G. None of the above.
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
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 13
Standard phones connect to Cisco router gateways as shown below:

Study the exhibit above carefully. Routers Certkiller 1 and Certkiller 2 are to be configured as VoIP gateways. On the basis of the information in the exhibit, which interface FastEthernet 0/0 configuration would be valid?
A. Certkiller 1(config-if)# dial-peer voice 1 voip Certkiller 1(config-dial-peer)# destination-pattern 1111 Certkiller 1(config-dial-peer)# port 1/0/0
B. Certkiller 1(config-if)# dial-peer voice 1 pots Certkiller 1(config-dial-peer)# destination-pattern 1111 Certkiller 1(config-dial-peer)# port 1/0/0
C. Certkiller 2(config-if)# dial-peer voice 1 voip Certkiller 2(config-dial-peer)# destination-pattern 1111 Certkiller 2(config-dial-peer)# port 1/0/0
D. Certkiller 2(config-if)# dial-peer voice 1 pots Certkiller 2(config-dial-peer)# destination-pattern 1111 Certkiller 2(config-dial-peer)# port 1/0/0
E. None of the above
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: Voice-Specific Commands
Command Description
dial-peer voicetag type Use the dial-peer voice command to enter the dial peer subconfiguration mode. The tag value is a number that has to be unique for all dial peers within the same gateway. The type value indicates the type of dial peer (for example, POTS or VoIP).
destination-pattern The destination-pattern command, telephone_number entered in dial peer subconfiguration mode, defines the telephone number that applies to the dial peer. A call placed to this number will be routed according to the configuration type and port (in the case of a POTS type dial peer) or session target (in the case of a VoIP type dial peer) of the dial peer.
portport-number The port command, entered in POTS dial peer subconfiguration mode, defines the port number that applies to the dial peer. Calls that are routed using this dial peer are sent to the specified port. The port command can be configured only on a POTS dial peer.
session target ipv4:ip-address The session target command, entered in VoIP dial peer subconfiguration mode, defines the IP address of the target VoIP device that applies to the dial peer. Calls that are routed using this dial peer are sent to the specified IP address. The session target command can be configured only on a VoIP dial peer.
QUESTION 14
Call Admission Control is being utilized in the Certkiller VOIP network. Which two statements are true about CAC? (Select two)
A. CAC is implemented in the call maintenance phase to allocate bandwidth resources.
B. CAC is implemented in the call setup phase to determine the destination of the call.
C. CAC is implemented in the call setup phase to allocate bandwidth resources.
D. CAC uses the Cisco RSVP (Resource Reservation Protocol) Agent to integrate call-processing capabilities with the underlying network infrastructure.
E. CAC is utilized during the call teardown phase to ensure that all resources have been released.
Correct Answer: CD Section: (none) Explanation
Explanation/Reference:
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 15
Part of the Certkiller VOIP network is shown below:

Study the exhibit carefully. Routers Certkiller 1 and Certkiller 2 are to be configured as VoIP gateways. Based on the information shown, which interface FastEthernet 0/0 configuration would be valid?
A. Certkiller 1(config-if)# dial-peer voice 2 voip Certkiller 1(config-dial-peer)# destination-pattern 1111 Certkiller 1(config-dial-peer)# session target ipv4:10.1.1.1
B. Certkiller 2(config-if)# dial-peer voice 2 pots Certkiller 2(config-dial-peer)# destination-pattern 1111 Certkiller 2(config-dial-peer)# session target ipv4:10.1.1.1
C. Certkiller 2(config-if)# dial-peer voice 2 voip Certkiller 2(config-dial-peer)# destination-pattern 1111 Certkiller 2(config-dial-peer)# session target ipv4:10.1.1.1
D. Certkiller 1(config-if)# dial-peer voice 2 pots Certkiller 1(config-dial-peer)# destination-pattern 1111 Certkiller 1(config-dial-peer)# session target ipv4:10.1.1.1
E. None of the above
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation: Voice-Specific Commands
Command Description
dial-peer voicetag type Use the dial-peer voice command to enter the dial peer subconfiguration mode. The tag value is a number that has to be unique for all dial peers within the same gateway. The type value indicates the type of dial peer (for example, POTS or VoIP).
destination-pattern The destination-pattern command, telephone_number entered in dial peer subconfiguration mode, defines the telephone number that applies to the dial peer. A call placed to this number will be routed according to the configuration type and port (in the case of a POTS type dial peer) or session target (in the case of a VoIP type dial peer) of the dial peer.
portport-number The port command, entered in POTS dial peer subconfiguration mode, defines the port number that applies to the dial peer. Calls that are routed using this dial peer are sent to the specified port. The port command can be configured only on a POTS dial peer.
session target ipv4:ip-address The session target command, entered in VoIP dial peer subconfiguration mode, defines the IP address of the target VoIP device that applies to the dial peer. Calls that are routed using this dial peer are sent to the specified IP address. The session target command can be configured only on a VoIP dial peer.
QUESTION 16
You want to ensure the highest level of audio quality in the Certkiller 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.
Correct Answer: CD Section: (none) Explanation
Explanation/Reference:
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 Certkiller 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.
Correct Answer: BC Section: (none) Explanation Explanation/Reference:
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 18
You need to calculate the bandwidth required to support VOIP on one of the remote Certkiller 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
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
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 Packetization Voice Packets Bandwidth Consumption Interval Payload in Per Second Per Bytes 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
G.729A 30 ms 30 33 19 kbps Reference: http://www.informit.com/articles/article.aspx?p=357102
QUESTION 19
Certkiller 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.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
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 Certkiller 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.
Correct Answer: BD Section: (none) Explanation
Explanation/Reference:
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

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