TestKing Cisco 640-460 Exam Questions & Answers

640-460 Implementing Cisco IOS Unified Communications (IIUC) Exam number/code: 640-460 Exam name: Implementing Cisco IOS Unified Communications (IIUC) Questions & Answers: 141 Q&A Related Certifications: CCNA Voice

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TestKing Cisco 640-460 Exam Questions & Answers

Exam: 640-460 Certification Questions & Answers

Question 1: Which Cisco's Proprietary protocol is used to inform the IP Phone of its voice VLAN ID. A. UDP B. CDP C. TCP D. RTP Answer: B

Question 2: You work as a network administrator at TestKing.com. You study the exhibit carefully. When a call is placed to 2000, phones 1 and 2 are paged. A call to 2001 pages phones 3 and 4. What command is missing so that a call to 2002 pages all four phones? Exhibit:

A. paging group 10 20 30 40 B. paging group 20 21 C. paging groupephones-all D. paging group all E. paging group 2000 2001 2002

TestKing Cisco 640-460 Exam Questions & Answers

Answer: B Explanation: The missing configuration is shown below. It's sample configuration.

Question 3: Which two of the following signaling protocols are peer-to-peer protocols? (Choose 2.) A. H.323 B. MGCP C. SIP D. SCCP Answer: A,C

Question 4: Which protocol provides VoiP packet sequence numbering? A. IP B. TCP C. UDP D. RTP E. RTCP F. G711 Answer: D Explanation: RTP is a critical component of VoIP because it enables the destination device to reorder and retime the voice packets before they are played out to the user. An RTP header contains a time stamp and sequence number, which allows the receiving device to buffer and to remove jitter and latency by synchronizing the packets to play back a continuous stream of sound. RTP uses sequence numbers to order the packets only. RTP does not request retransmission if a packet is lost.

Question 5: Which of the following are Cisco-Supported IP telephony deployment models? Select all that apply. A. Multisite with centralized call processing B. Multisite with distributed call processing C. Single Site D. Clustering over the IP WAN Answer: A,B,C,D Explanation: Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/uc_system/UC7.0.1/system_description/S DMOD.html

Question 6:

TestKing Cisco 640-460 Exam Questions & Answers

The Point North Company has a few slow links in its voice and data network. Which two techniques can be used to reduce delay in voice transmission? (Choose two.) A. buffering voice packets B. compression of IP, RTP, and UDP headers C. FIFO queuing D. increasing priority queue sizes E. fragmentation of large packets Answer: B,E Explanation: To reduce the huge bandwidth overhead caused by the IP, UDP, and RTP headers, RTP header compression ( cRTP ) can be used. The name is bit misleading because cRTP not only compresses the RTP header, but it also compress the IP and UDP headers. When you are configuring the proper fragment size to use on a link, a typical goal is to have a maximum serialization delay of around 10 to 15 ms. Depending on the fragmentation mechanisms being configured, the fragment size is either configured in bytes or in milliseconds. Fig is shown below.

Question 7: The call leg and the dial peer are both physical connections used to complete an end-to-end call. A. FALSE B. TRUE Answer: A Explanation: A traditional voice call over the public switched telephone network (PSTN) uses a dedicated 64-kb/s end-to-end circuit. In contrast, a voice call over the packet network is made up of discrete segments or call legs. A call leg is a logical connection between two routers or between a router and a telephony device. A voice call consists of four call legs: two from the perspective of the originating router and two from the perspective of the terminating router, as shown in the below figure.

Question 8: Which two are considered endpoints in a Cisco Unified Communications solution? (Choose two.) A. call agent B. analog phone C. gateway D. H.323 gatekeeper E. Cisco Unified Communications Manager F. IP telephone Answer: B,F Explanation:

Question 9: Which is the best description of time-division multiplexing?

TestKing Cisco 640-460 Exam Questions & Answers

A. All sources get an interleaved slice of time, which offers the entire frequency range allocated for that timeslot. B. A channel is assigned an exclusive slice of the overall frequency of the circuit for the entire time of its operation. C. On a T1 circuit, this is the process where 24 DS-1 signals are multiplexed into a single DS-0 channel, while on a T3 circuit 24 DS-0 signals are multiplexed into a single DS-3 signal. D. Technology that increases the transmission capabilities by dividing the medium into multiple channels that are each assigned a wavelength based on statistical analysis. E. Individual source signals are combined into a composite signal, which allows a capacity equal to or greater than the sum of the component signals. Answer: A Explanation: TDM allows voice networks to carry multiple conversations at the same time over a single, four-wire path. Because the multiple conversations have been digitized, the numeric values are transmitted in specific time slots (thus the "time-division") that differentiate the separate conversations. Figure below illustrates three separate voice conversations sent over a digital connection.

Question 10: What is the key command to use when deploying the partially automated telephone setup process? A. auto phone-type B. autoqos C. auto telephony-service D. auto start-ephone-dn E. auto start-dn F. auto assign Answer: F Explanation: The auto assign command performed from telephony service configuration mode. This command allows you to specify a range of ephone-dns to distribute to IP phones that register but have no explicit ephone configuration in the CME router. The auto assign command allows you to distribute specific ephone-dn ranges to specific types of phones or to any phone requesting an extension. For example, you could use the syntax in fig below to auto-assign ephone-dns 20 through 24 to Cisco 7940 IP Phones, ephone-dns 25 through 30 to Cisco 7960 IP Phones, and ephone-dns 31 through 39 to any other phone model that attempts to register.

Question 11: Which three are components of a dial plan? (Choose three.) A. call legs B. decentralized control C. centralized control D. call coverage E. digit manipulation F. endpoint addressing Answer: D,E,F Explanation:

TestKing Cisco 640-460 Exam Questions & Answers

A dial plan consists of these components: Endpoint addressing (Numbering Plan): Assigning directory numbers to all endpoints (such as IP phones, fax machines, and analog phones), and applications (such as voice-mail systems, auto attendants, and conferencing systems) enables you to access internal and external destinations. Call routing and path selection: Depending on the calling device, you can select different paths to reach the same destination. Moreover, you can use a secondary path when the primary path is not available. For example, a call can be transparently rerouted over the public switched telephone network (PSTN) during an IP WAN failure. Digit manipulation: In some cases, you need to manipulate the dialed string before routing a call; for example, when a call originally dialed using the on-net access code is rerouted over the PSTN, or when an abbreviated code (such as 0 for the operator) is expanded to an extension. This can occur prior to or after a routing decision has been made. Calling privileges: You can assign different groups of devices to different classes of service, by granting or denying access to certain destinations. For example, you might allow lobby phones to reach only internal and local PSTN destinations, while executive phones could have unrestricted PSTN access. Call coverage: You can create special groups of devices to handle incoming calls for a certain service according to different rules (top-down, circular hunt, longest idle, or broadcast). This also ensures that calls are not dropped without being answered.

Question 12: Identify three quality issues that can result because of a lack of network bandwidth. (Choose 3.) A. Jitter B. Impedance C. Delay D. Packet loss Answer: A,C,D

Question 13: You work as a network administrator at TestKing.com. Your boss, Miss Tess King, is interested in signaling descriptions. Make the appropriate matches. Each option should be used once and only once.

TestKing Cisco 640-460 Exam Questions & Answers

Answer:

Explanation: Signaling techniques can be placed into one of three categories: Supervisory: Involves the detection of changes to the status of a loop or trunk. Once these changes are detected, the supervisory circuit generates a predetermined response. A circuit (loop) can close to connect a call, for example. Addressing: Involves passing dialed digits (pulsed or tone) to a PBX or CO. These dialed digits provide the switch with a connection path to another phone or customer premises equipment (CPE). Informational: Provides audible tones, which indicate certain conditions such as an incoming call or a busy phone, to the user .

TestKing Cisco 640-460 Exam Questions & Answers

Question 14: Approximately what percentage of voice packets can be dropped before voice quality becomes poor? A. Less than or equal to 1% B. 15% C. 1 - 2% D. 5 - 10% Answer: A Explanation: Voice packets drop should be less than or equal to 1% to maintain good voice quality.

Question 15: An analogy telephone is connected to a __________ port on a router? A. FXO B. T1 C. E1 D. FXS Answer: D Explanation: A router typically uses Foreign Exchange Station (FXS) analog interfaces to connect to analog devices such as telephones, fax machines, and modems . See the fig below.

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