[FREE] Voip Exam Questions And Answers | HOT!
Refer to the exhibit. R1 is configured for NAT as displayed. What is wrong with the configuration? The NAT pool is incorrect. Access-list 1 is misconfigured. Use a static route pointing to the ISP and redistribute it. Use the redistribute static...
[DOWNLOAD] Voip Exam Questions And Answers
The array can include only one value type. A space must separate each value in the array. A semicolon separates the key and list of values. Values are enclosed in square brackets. What is a characteristic of a Trojan horse as it relates to network...
CCNA 3 V7.0 Final Exam Answers Full – Enterprise Networking, Security, And Automation
MGCP gateway C. DSP D. Directory number for phone is assigned to the New York partition. Directory number for phone is assigned to the Seattle partition. What will happen you dial ? The call will require additional dialed digits to be connected B. The call is sent to the first listed partition in the CSS C. You are prompted by the IP phone to choose the destination D.
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Specifies the bandwidth that is used for an audio or video call B. Implements call admission control in a centralized call-processing deployment C. Defines sets of common characteristics for devices D. Contains common configuration options for IP phones E. Sets device load, device pool and phone button template for each auto-registered device type Identify functions of RTCP. Choose two A. RTCP provides out-of-band statistics and control information C.
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VoIP Frequently Asked Questions And Answers
RTCP monitors the quality of packet count, packet loss, delay and jitter D. Can be configured by the administrator or user when the call destination is already engaged C. Configured by the phone user to forward calls regardless of the circumstance D. Configured on the CUC Manager to handle calls forwarded to a Hunt Pilot by a user when no Hunt Pilot is available Presence groups control the destinations that a watcher can monitor. Which statements concerning their operation are true? Choose four A. Conferencing support C. Support for fax messaging D. Support for Tcl scripts E.
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Runs directly on Cisco IOS software Cisco Unified Communications Manager uses call routing logic when selecting which pattern to match from its call routing table. When multiple potentially matching patterns are present how is the destination pattern chosen? It uses closest-match logic to select the best destination pattern B. It matches the dial string C. It matches the most strings among matching patterns D. It uses an identification tag to map to a call routing table dial string What Quality of Service characteristics correspond to the IntServ QoS model? A multiple-service model that can accommodate multiple QoS requirements C. Applications reserve bandwidth for a specific time D. At the user level B. At the system level C. Roles are assigned by default D. In the user template What codec is the least processor intensive? Which steps are valid for the successful configuration of Cisco Extension Mobility? To configure a Default Device Profile, the Default Device Profile for the chosen phone model and protocol must be chosen first, then the product type and device protocol can be configured C.
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To associate users with device profiles, go to the End User Configuration window and choose the device profile or profiles that should be associated with the user in the list of Available Profiles D. Choose three A. Call Admission Control.
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Multiple Choice Questions
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VoIP FAQ: 71 Questions & Answers About All Things VoIP
The basic principle of Voip is very simple. It's the same technology you have probably used already to listen to music over the Internet. Voice sounds are picked up by a microphone and digitized by the sound card. The sounds are then converted to a compressed form, compact enough to be sent in real time over the Internet, using a software driver called a codec.
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Voip Telephony Interview Questions & Answers
The sounds are encoded at the sending end, sent over the Internet and then decoded at the receiving end, where they are played back over the speakers. The only requirements are a connection between the two computers of an adequate speed, and matching codecs at each end. To be usable, a Voip system also needs a method for establishing and managing a connection, for example, calling the other computer, finding out if they accept the call, and closing the connection when a user hangs up. Because Voip allows two way communication, and even conference calls, it's a lot more complicated than simple audio streaming. How calls are managed is the area in which Voip systems fundamentally differ, and two Voip users must be using the same system or compatible ones in order to be able to call each other. Because most Internet users don't have a permanent Internet address IP address, a number like Instead, each user of the service registers with an intermediate server, which maintains a record of their IP address all the time they are connected.
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Voice Over Internet Protocol (VOIP) Interview Questions & Answers
An example of a Voip application that works this way is Picophone. The small size of the PicoPhone application file about 64Kb, barely larger than Windows Notepad demonstrates clearly that the basic principles of Voip are not complicated to implement. Another reason for using an intermediate server is that it eases the problem of getting Voip to work through the firewalls that everyone uses these days. Many firewalls block any data from the Internet that is not sent in response to a specific request.
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This makes it impossible to call another computer direct. Because the called computer did not request any data from the caller, the call request would be blocked. By establishing a connection with a server, the Voip software opens a channel of communication through which other computers can call it. Communication may continue using the server, or information may be passed via the server that allows the two computers to open a direct connection between them and continue using that. Is This Answer Correct?
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IT Fundamentals Multiple Choice Questions And Answers-Information Systems
Which two statements are true regarding SCCP? Note: There can be multiple correct answers to this question. Endpoints can operate autonomously if communication with Cisco Unified Communications Manager is lost. SCCP may interoperate with H. Endpoints and gateways maintain the dial plan. SCCP uses hex messages for communication. A new Cisco Unified Communications Manager Express system has been deployed and the technician is trying to add the first new IP phone to the system.
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Answers To Your VoIP Questions
The phone powers up, but it does not register with the system. The phone has been assigned to an iphone and the correct MAC address is configured. With the information provided, which two of the following does the administrator need to verify to resolve this situation? Verify that the ip helper-address is correctly configured. Verify that telephony-service has been configured. Verify that the ephone has a button assigned. Verify that the tftp-server path has been configured. Verify that the correct phone type files are in the tftp-server path. The voice subnet has an address of
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The big advantage is VoIP may save you money depending on how much you are currently spending for local and long-distance calls. What you will need to do is get the total cost the phone company is charging and compare it against a VoIP plan that interests you. With most plans, you get free calls within the U. International calls usually have very low rates with no connection fees. For both residential customers and businesses that make a lot of long distance and international calls, the savings can be several hundred dollars a year. Another advantage is with the features available with VoIP. Features such as caller ID, call waiting, call forwarding, 3 way conferencing and voice mail are usually included at no extra cost. With the phone company, these services are usually extra. In addition, you can make free phone calls anywhere there is a high speed Internet connection available. That means you can be in another state or even in another country and make calls as if you were back at your home or business.
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You will just need to bring your phone adapter along with you and possibly a phone in case one is not available. Is This Answer Correct?
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Network+ VoIP - ProProfs Quiz
The one-way delay over a km optical fibre link between Manchester and New York: a cannot be estimated b exceeds about ms c cannot be much less than 30ms d should be less than 3 ms 2. The effect of increasing the bandwidth of the connection between two components of a distributed system will be to: a decrease the latency delay without necessarily affecting the occurrence of lost packets. The lowest layer providing host-to-host connection is: a physical layer b data-link layer c network layer d transport layer 5. Which of the following statements is true? The 16 bit checksum included in the IP header will allow: a Any bit-error in the header itself to be detected and corrected b Any bit-error in the header itself to be detected but not corrected. Which one of the following statements is true: a A virtual circuit offers reliable connection-oriented service at the network layer with improved quality of service as compared with traditional IP.
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Why is VoIP telephony more demanding than streaming audio as used for sound clips and Internet radio? How could you avoid the accumulation of delay and distortion due to buffer under-flows and over-flows. How can NTP try to distribute time with high accuracy? How is connectivity achieved at the network layer? How is connectivity achieved at the transport layer? Why does each IP datagram header have a time to live?
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Since routers change IP headers, does the checksum have to change? With VoIP, why can t we forget about a missing packet and go on with next one i. Explain how the VoIP application illustrates the axiom that the latency of many network links is not so low that it can normally be disregarded. Can you relate the two army problem to a scenario different from the releasing of TCP? Why can t the time-stamp within RTP packets be used to estimate one-way delay? Explain why it would be difficult to have a precisely synchronized measure of exact time across all components of a distributed system. Why can t we just listen to the pips on the one-oclock news on the radio and all set our computer clocks at that point in time? While thinking about the pips, if you have a DAB radio why are they always a little late? What is round trip delay and timing jitter and why are these parameters much easier to measure?
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Voice Over Internet Protocol (VoIP) Interview Questions And Answers | 1medicoguia.com
What are the major causes of round trip delay and timing jitter in IP networks? Explain how the transmission control protocol TCP achieves reliable and connection oriented communication over an IP network? In giving your answer, explain: i What a buffer is, by referring to the leaky bucket analogy ii Why input buffers and output buffers are needed when communicating with concurrent A-to-D and D-to-A conversion processes, as implemented on sound cards or equivalent on-board circuitry. Explain how the occurrence of lost packets can be identified at the receiver. How can such jitter be measured for two successive packets conveyed by RTP? Explain how the receiver s jitter-buffer affects the conversation. How would increasing the jitter-buffer size change this effect and what limits the size of jitter-buffer that may be employed with VoIP? Answers and discussion: Multiple choice: 1. Initially IP networks were used for VoIP i as a means of providing cheap or free telephone calls over the public Internet ii to allow private companies, especially large international corporations, to use the dedicated computer network data links owned or leased by them to be use also for telephony.
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CCNA Voice Practice Test | CICD Exam
It is convenient to have just one type of network IP rather than two IP and telephone. Nowadays, the public service telephone service PSTN and mobile telephony is changing universally to IP, with virtual circuits normally based on NPLS, to eliminate the distinction between telephone networks and computer networks. With real time telephony, the round trip delay latency must be less than about ms for natural 2-way conversation and therefore a large jitter-buffer cannot be employed. Streaming can afford much larger jitter-buffer sizes to provide a reservoir of samples in case some packets are excessively delayed. Because a VoIP device cannot be sure when it will get a transmit opportunity i. Mobile phones sleep in this way and save a lot of battery life. VoWiFi traditionally must remain on and consume power all the time. Read Barry sides to address this problem. At either end you need buffers with high water level and low water level markings, referring to the leaky bucket analogy.
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Friday, May 28, 2021
Voip Exam Questions And Answers
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