System Engineer Atsi Quiz
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Questions and Answers
- 1.
You are implementing a converged network. You need to run a cable to a computer that is located 250 meters from the current network. You will need to install either switches or repeaters to implement the connection using standard CAT5 UTP cable. How many switches or repeaters would you need?
- A.
1
- B.
2
- C.
3
- D.
4
Correct Answer
B. 2Explanation
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- 2.
Which of the following are protocols used to dynamically configure the IP addresses on IP devices?
- A.
DNS
- B.
BOOTP
- C.
DHCP
- D.
FTP
Correct Answer
C. DHCPExplanation
DHCP stands for Dynamic Host Configuration Protocol. It is a network protocol used to automatically assign IP addresses and other network configuration parameters to devices on a network. DHCP eliminates the need for manual configuration of IP addresses, making it easier to manage and maintain IP devices in a network. DNS (Domain Name System) is a protocol used for translating domain names into IP addresses. BOOTP (Bootstrap Protocol) is an older protocol similar to DHCP but with limited functionality. FTP (File Transfer Protocol) is a protocol used for transferring files over a network. Therefore, the correct answer is DHCP as it is specifically designed for dynamically configuring IP addresses on IP devices.Rate this question:
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- 3.
You are implementing a Voice over IP solution. You want to make sure that your data is delivered in a timely fashion to the receiving device. Which of the following protocols is the right choice?
- A.
HTTP
- B.
SSH
- C.
TCP
- D.
UDP
Correct Answer
D. UDPExplanation
UDP (User Datagram Protocol) is the right choice for delivering data in a timely fashion in a Voice over IP (VoIP) solution. Unlike TCP (Transmission Control Protocol), UDP does not provide reliable delivery or error correction mechanisms. However, it is a lightweight and fast protocol, making it suitable for real-time applications like VoIP. UDP's lack of error checking and retransmission allows for lower latency, ensuring that the data is delivered quickly to the receiving device.Rate this question:
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- 4.
At which layer of the OSI model does IP address management occur?
- A.
Layer 3
- B.
Layer 1
- C.
Layer 2
- D.
Layer 4
Correct Answer
A. Layer 3Explanation
During the night the high contrast between the bright moon and the night's dark skies make the Moon look white.Rate this question:
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- 5.
What is the difference between a frame and a packet?
- A.
The term frame usually refers to the data at Layer 2 that is ready to be transmitted.
- B.
The term packet usually refers to the data at Layer 2 that is ready to be transmitted
- C.
Frames are encrypted, and packets are not
- D.
Packets are encrypted, and frames are not
Correct Answer
A. The term frame usually refers to the data at Layer 2 that is ready to be transmitted.Explanation
A frame is a term used in networking to refer to the data at Layer 2 of the OSI model, which is the data link layer. It is the unit of data that is ready to be transmitted over a network. On the other hand, a packet refers to the data at Layer 3 of the OSI model, which is the network layer. It is a logical unit of data that contains both the payload (actual data) and the necessary control information for routing and delivery. Therefore, the correct answer is that the term frame usually refers to the data at Layer 2 that is ready to be transmitted.Rate this question:
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- 6.
You need a device that will allow you to move data throughout your network based on IP address information. Which of the following devices are you most likely to implement?
- A.
Switch
- B.
Router
- C.
Access Point
- D.
CSU/DSU
Correct Answer
B. RouterExplanation
A router is the most likely device to be implemented in order to move data throughout a network based on IP address information. Routers are specifically designed to connect multiple networks together and make decisions on where to forward data packets based on their IP addresses. They can determine the most efficient path for data transmission and direct it accordingly. Switches, access points, and CSU/DSUs have different functions and may not be suitable for this specific task.Rate this question:
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- 7.
Which of the following reduces bandwidth consumption by taking advantage of the fact that users do not talk 100 percent of the time while on a telephone conversation?
- A.
Chatter removal
- B.
Noise reduction
- C.
Silence suppression
- D.
Silence compression
Correct Answer
C. Silence suppressionExplanation
Silence suppression is the correct answer because it reduces bandwidth consumption by detecting and removing periods of silence during a telephone conversation. Since users do not talk 100 percent of the time, silence suppression helps optimize the use of bandwidth by not transmitting audio data during these silent periods. This technique is commonly used in telecommunication systems to improve efficiency and reduce network congestion.Rate this question:
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- 8.
Which of the following features is generally not found in a traditional PBX?
- A.
Voice mail
- B.
Call forwarding
- C.
Call interruption
- D.
Instant messaging
Correct Answer
D. Instant messagingExplanation
Instant messaging is generally not found in a traditional PBX. A traditional PBX (Private Branch Exchange) is a telephone system that handles incoming and outgoing calls for an organization. It is primarily designed for voice communication and does not typically include features such as instant messaging. Instant messaging is a form of real-time text communication that is commonly found in modern communication platforms, but it is not a standard feature in traditional PBX systems.Rate this question:
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- 9.
You are attaching an ISDN PRI line (E1) to your voice gateway. Which channel carries signaling and control information?
- A.
B Channel
- B.
C Channel
- C.
D Channel
- D.
E Channel
Correct Answer
C. D ChannelExplanation
The D channel carries signaling and control information in an ISDN PRI line. This channel is responsible for transmitting call setup and teardown signals, as well as other signaling messages between the voice gateway and the telephone exchange. The B channel, on the other hand, carries the actual voice or data traffic. The C and E channels are not used for signaling or control purposes in an ISDN PRI line.Rate this question:
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- 10.
What is the resulting data rate of Pulse Code Modulation (PCM), as traditionally defined?
- A.
8 Kbps
- B.
16 Kbps
- C.
56 Kbps
- D.
64 Kbps
Correct Answer
D. 64 KbpsExplanation
Pulse Code Modulation (PCM) is a digital encoding technique used to convert analog signals into digital signals. In PCM, the analog signal is sampled and quantized, and each sample is represented by a binary code. The resulting data rate of PCM is determined by the sampling rate and the number of bits used for each sample. Traditionally, PCM uses a sampling rate of 8 kHz and 8 bits per sample, resulting in a data rate of 64 Kbps (kilobits per second).Rate this question:
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- 11.
How does a Computer attached to an IP phone send data to the switch?
- A.
As tagged (using the voice VLAN)
- B.
As untagged
- C.
As tagged (using the data VLAN)
- D.
As tagged (using the CoS value)
Correct Answer
B. As untaggedExplanation
When a computer is attached to an IP phone, it can send data to the switch as untagged. This means that the data packets from the computer are not marked with any VLAN information. The IP phone acts as a bridge between the computer and the switch, stripping off any VLAN tags from the data packets and forwarding them to the switch. This allows the switch to receive and process the data without any VLAN information attached to it.Rate this question:
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- 12.
Identify the issues facing converged enterprise networks. (Choose four)
- A.
Scarce bandwidth
- B.
Delay issues
- C.
CAS issues
- D.
Jitter
- E.
E&M connectivity issues
- F.
Packet Loss
Correct Answer(s)
A. Scarce bandwidth
B. Delay issues
D. Jitter
F. Packet LossExplanation
The issues facing converged enterprise networks include scarce bandwidth, delay issues, jitter, and packet loss. Scarce bandwidth refers to the limited amount of network capacity available for data transmission, which can lead to congestion and slower network performance. Delay issues occur when there is a delay in transmitting data packets, resulting in slower response times and potential disruptions in real-time communication. Jitter refers to the variation in the delay of packet delivery, which can cause packet loss and affect the quality of voice and video communication. Packet loss is the failure of data packets to reach their destination, resulting in data loss and potential disruptions in communication.Rate this question:
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- 13.
Voice over IP is an application that requires efficient use of bandwidth and reliability. Which technology can be used to help ensure the efficient transport of VoIP traffic?
- A.
RTP
- B.
QoS
- C.
DNS
- D.
RSTP
Correct Answer
B. QoSExplanation
QoS stands for Quality of Service, which is a technology that can be used to help ensure the efficient transport of VoIP traffic. QoS prioritizes and manages network traffic to provide better performance and reliability for specific applications, such as Voice over IP. By giving priority to VoIP traffic, QoS helps to minimize latency, packet loss, and jitter, which are crucial factors for a smooth and uninterrupted VoIP communication. Therefore, QoS is essential in ensuring the efficient use of bandwidth and maintaining the reliability of Voice over IP applications.Rate this question:
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- 14.
Which of the following could be a valid Media Access Control (MAC) address?
- A.
0-0-f-f-1-2
- B.
0e0-8y0-5hkf-fie-184-cq2
- C.
300-180-55f-2fe-514-c82
- D.
00-80-5f-fe-14-c2
Correct Answer
D. 00-80-5f-fe-14-c2Explanation
The given MAC address "00-80-5f-fe-14-c2" could be a valid MAC address because it follows the correct format of six pairs of hexadecimal digits separated by hyphens. Additionally, the first octet starts with "00" which indicates that it is a universally administered address.Rate this question:
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- 15.
You are using private IP addresses for your Windows, Linux and Macintosh systems behind a firewall. What must you do with these addresses in order for them to access the Internet?
- A.
Enable Network Address Translation (NAT)
- B.
Enable Dynamic Host Configuration Protocol (DHCP)
- C.
Enable Simple Network Management Protocol (SNMP)
- D.
Enable a default gateway
Correct Answer
A. Enable Network Address Translation (NAT)Explanation
Private IP addresses are not routable on the internet, so in order for systems with private IP addresses to access the internet, Network Address Translation (NAT) must be enabled. NAT allows the private IP addresses to be translated to a public IP address that can communicate with the internet. This allows for multiple devices with private IP addresses to share a single public IP address, conserving the limited number of available public IP addresses.Rate this question:
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- 16.
Several customers call informing you that a computer responsible for enabling client use of the VoIP network is issuing malformed packets. You suspect that TCP packets with the address of 1718 are malformed, and you need to verify this. Which tool could you use to read the contents of the packets?
- A.
A network analyzer
- B.
The netstat command
- C.
The traceroute utility
- D.
An ARP cache
Correct Answer
A. A network analyzerExplanation
A network analyzer is a tool that can be used to capture and analyze network traffic. It allows you to inspect the contents of packets, including their addresses, headers, and payload. By using a network analyzer, you can examine the TCP packets with the address of 1718 and determine if they are indeed malformed. This tool is commonly used by network administrators and security professionals to troubleshoot network issues and identify potential problems in the network traffic.Rate this question:
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- 17.
Which protocol is generally used to establish calls in SIP implementations?
- A.
UDP
- B.
TCP
- C.
RTP
- D.
H.225
Correct Answer
A. UDPExplanation
SIP (Session Initiation Protocol) is a signaling protocol used for initiating, modifying, and terminating real-time sessions involving video, voice, messaging, and other communications applications and services. It is generally used to establish calls in SIP implementations through the use of UDP (User Datagram Protocol). UDP is a connectionless protocol that provides a lightweight and efficient method for transmitting data packets over the network. Unlike TCP, UDP does not establish a reliable and ordered connection, making it suitable for real-time communication where low latency is crucial. RTP (Real-time Transport Protocol) is used for delivering the actual audio and video data, while H.225 is a protocol used in H.323 systems, not SIP.Rate this question:
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- 18.
When a technician connects a North American T-1 trunk to a European E-1 trunk using a conversion device, which of the following will result?
- A.
No less than 80% of the T-1 can be used.
- B.
No more than 80% of the E-1 trunk can be used.
- C.
No connection can be made between E-1 and T-1.
- D.
The connection will fail, because E-1 uses out-of-band transmission, and T-1 lines use in-band transmission.
Correct Answer
B. No more than 80% of the E-1 trunk can be used.Explanation
When a technician connects a North American T-1 trunk to a European E-1 trunk using a conversion device, no more than 80% of the E-1 trunk can be used. This is because T-1 and E-1 trunks have different transmission rates and signaling formats. T-1 trunks have a transmission rate of 1.544 Mbps, while E-1 trunks have a transmission rate of 2.048 Mbps. The conversion device will not be able to fully utilize the higher transmission rate of the E-1 trunk, resulting in a maximum usage of 80% of the E-1 trunk.Rate this question:
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- 19.
How many B Channels and D Channels does E1 voice line support
- A.
E1 supports 24 B Channels and 1 D Channel
- B.
E1 supports 24 D Channels and 1 B Channel
- C.
E1 supports 30 D Channels and 1 B Channel
- D.
E1 Supports 30 B Channels and 1 D Channel
Correct Answer
D. E1 Supports 30 B Channels and 1 D ChannelExplanation
E1 voice line supports 30 B Channels and 1 D Channel. E1 is a digital transmission format that is commonly used for voice and data communications. It is divided into 32 timeslots, with each timeslot able to carry either a voice channel or data channel. B Channels are used for carrying voice or user data, while the D Channel is used for signaling and control information. Therefore, E1 supports 30 B Channels for voice or data and 1 D Channel for signaling and control.Rate this question:
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- 20.
Which of the following describes an E-1 line used for voice data?
- A.
It uses in-band signaling.
- B.
It provides 30 64-Kbps channels.
- C.
It provides 24 64kbps data channels
- D.
It provides a connection speed of 1.544 Kbps
Correct Answer
B. It provides 30 64-Kbps channels.Explanation
An E-1 line is a digital transmission link used for voice data. It provides a connection speed of 2.048 Mbps and consists of 30 time-division multiplexed channels, with each channel having a bandwidth of 64 Kbps. These channels can be used for voice or data transmission, making the E-1 line suitable for various telecommunications applications.Rate this question:
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- 21.
Your company's users on the primary LAN have been asked to use soft phones on their personal computers to make voice calls. You have implemented a VLAN. What priority should you mark voice and computer data?
- A.
Computer data: 0. Voice data: 6
- B.
Computer data: 0. Voice data: 9
- C.
Computer data: 1. Voice data: 0
- D.
Computer data: 5. Voice data: 6
Correct Answer
A. Computer data: 0. Voice data: 6Explanation
The correct answer is Computer data: 0. Voice data: 6. By marking computer data with a priority of 0 and voice data with a priority of 6, it ensures that voice data is given higher priority in the network compared to computer data. This is important for soft phones to have a smooth and uninterrupted voice call experience, as voice data requires low latency and minimal packet loss. By assigning a higher priority to voice data, it ensures that it is given preferential treatment and is transmitted with minimal delay.Rate this question:
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- 22.
Which of the following is responsible for ensuring that incoming customer calls are properly queued according to the order in which they were received?
- A.
SIP proxy
- B.
Edge router
- C.
Automatic Call Distributer (ACD)
- D.
Gatekeeper
Correct Answer
C. Automatic Call Distributer (ACD)Explanation
The Automatic Call Distributor (ACD) is responsible for ensuring that incoming customer calls are properly queued according to the order in which they were received. ACD systems are commonly used in call centers to efficiently manage and distribute incoming calls to available agents. They can prioritize calls based on various factors such as wait time, caller's identity, or specific criteria set by the organization. ACDs help streamline the call handling process and ensure fair and efficient distribution of customer calls.Rate this question:
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- 23.
Which of the following describes the ability to seamlessly combine e-mail, voice, real-time applications and Web chat features for a call center representative to use?
- A.
Multi-channel blending
- B.
On-screen phone control
- C.
Automatic Call Routing (ACR)
- D.
Computer telephony integration (CTI).
Correct Answer
A. Multi-channel blendingExplanation
Multi-channel blending refers to the ability to integrate different communication channels, such as email, voice, real-time applications, and web chat, into a unified platform for call center representatives. This feature allows representatives to seamlessly switch between different channels during customer interactions, enhancing their efficiency and providing a better customer experience. It enables representatives to handle multiple customer inquiries simultaneously and ensures that all communication channels are integrated and managed effectively.Rate this question:
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- 24.
As a general rule, which of the following statements is correct?
- A.
Decreasing compression increases bandwidth usage and reduces voice quality.
- B.
Increasing compression to reduce bandwidth consumption increases voice quality.
- C.
Increasing compression increases bandwidth consumption and reduces voice quality.
- D.
Increasing compression to reduce bandwidth consumption reduces voice quality.
Correct Answer
D. Increasing compression to reduce bandwidth consumption reduces voice quality.Explanation
Increasing compression reduces the amount of data needed to transmit voice signals, which in turn reduces the bandwidth consumption. However, this reduction in bandwidth comes at the cost of voice quality. As compression increases, the voice signals are more heavily compressed, leading to loss of detail and clarity in the transmitted audio. Therefore, increasing compression to reduce bandwidth consumption ultimately results in a decrease in voice quality.Rate this question:
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- 25.
You have just changed codecs from G.711 to G.729. Which of the following statements explains the resulting change in voice quality?
- A.
Voice quality is unaffected by compression rates
- B.
Voice quality improves as compression increases
- C.
Voice quality degrades as compression decreases
- D.
Voice quality degrades as compression increases
Correct Answer
D. Voice quality degrades as compression increasesExplanation
When changing codecs from G.711 to G.729, the compression rate increases. Higher compression rates result in more data being removed from the audio signal, leading to a loss of quality. Therefore, voice quality degrades as compression increases.Rate this question:
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- 26.
The Real time Transport Protocol (RTP) can run on top of various protocols. Which is the most commonly implemented along with RTP?
- A.
SIP
- B.
UDP
- C.
TCP
- D.
H.323
Correct Answer
B. UDPExplanation
RTP, the Real-time Transport Protocol, is commonly implemented along with UDP, the User Datagram Protocol. UDP is a connectionless protocol that offers low-latency communication, making it suitable for real-time applications like streaming media and voice over IP (VoIP). RTP utilizes the services provided by UDP, such as the ability to send data without establishing a connection and the option to prioritize real-time traffic. TCP, on the other hand, is a connection-oriented protocol that introduces more latency due to its reliability mechanisms, making it less suitable for real-time applications. SIP and H.323 are signaling protocols used for session initiation in VoIP systems, but they do not directly handle the transport of real-time data like UDP does.Rate this question:
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- 27.
Which of the following features is commonly provided in an IP-based PBX, but not in a traditional PBX?
- A.
Signal monitoring
- B.
Soft phone support
- C.
DTMF phone support
- D.
Quality of Service (QoS).
Correct Answer
B. Soft phone supportExplanation
Soft phone support is commonly provided in an IP-based PBX but not in a traditional PBX. Soft phone support allows users to make and receive calls using their computer or mobile device, eliminating the need for a physical telephone. This feature is not available in traditional PBX systems, which rely on physical telephones to make and receive calls. Soft phone support offers flexibility and convenience as users can make calls from anywhere with an internet connection, making it a valuable feature in IP-based PBX systems.Rate this question:
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- 28.
Identify the QoS considerations within a converged network. (Choose two)
- A.
Packets cannot tolerate more than a 150-ms delay
- B.
Videoconferencing streams would require the actual bandwidth of the stream plus an additional 20%
- C.
Jitter delay should not exceed 40 ms
- D.
LFI can be used to save bandwidth on a link
Correct Answer(s)
A. Packets cannot tolerate more than a 150-ms delay
B. Videoconferencing streams would require the actual bandwidth of the stream plus an additional 20%Explanation
The QoS considerations within a converged network include ensuring that packets do not experience a delay of more than 150-ms and that videoconferencing streams have enough bandwidth, including an additional 20% to accommodate for any fluctuations. These considerations are important to maintain the quality of service for real-time applications and ensure a smooth user experience.Rate this question:
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- 29.
Which elements determine that a SIP phone receives the correct date and time information?
- A.
Database Server
- B.
FTP Server
- C.
NTP Server
- D.
DHCP Server
Correct Answer
C. NTP ServerExplanation
The correct answer is NTP Server. NTP (Network Time Protocol) is used to synchronize the time on devices within a network. A SIP phone relies on the NTP server to receive the correct date and time information. The NTP server ensures that all devices in the network have accurate and synchronized time, which is crucial for various applications and services, including SIP communication. The other options, such as Database Server, FTP Server, and DHCP Server, do not directly provide time synchronization functionality.Rate this question:
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- 30.
What is the purpose of Call Admission Control (CAC) in a multi-site deployment interconnected by an IP WAN?
- A.
Used to limit the number of voice calls and maintain bandwidth limits
- B.
Provides the means to prioritize voice over data traffic
- C.
Provides detailed information about call activities and voice quality
- D.
Assigns codecs to gateway interfaces based on VoIP traffic demands
Correct Answer
A. Used to limit the number of voice calls and maintain bandwidth limitsExplanation
Call Admission Control (CAC) is used to limit the number of voice calls and maintain bandwidth limits in a multi-site deployment interconnected by an IP WAN. This ensures that the network resources are efficiently utilized and prevents congestion by controlling the number of simultaneous voice calls that can be made. By managing the bandwidth limits, CAC ensures that the voice traffic does not overwhelm the network and affect the quality of service for other applications. CAC plays a crucial role in maintaining a reliable and high-quality communication system in a multi-site deployment.Rate this question:
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- 31.
You are implementing a VoIP solution, and the client has stipulated that a device must be installed that will register and manage VoIP endpoints. What is this device called?
- A.
MCU
- B.
Gateway
- C.
Gatekeeper
- D.
Firewall
Correct Answer
C. GatekeeperExplanation
A gatekeeper is a device that is responsible for registering and managing VoIP endpoints in a VoIP solution. It controls access to the network and ensures that only authorized endpoints can connect and communicate. The gatekeeper also handles call routing, address translation, and bandwidth management. It acts as a central point of control for the VoIP system, allowing for efficient and secure communication between endpoints.Rate this question:
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- 32.
What's the equipment is needed to build VoIP Solution (Check all that apply)
- A.
IP Telephone
- B.
Storage Server
- C.
SIP Server
- D.
Fiber Cables
- E.
Voice Gateway
Correct Answer(s)
A. IP Telephone
C. SIP Server
E. Voice GatewayExplanation
To build a VoIP solution, the necessary equipment includes an IP Telephone, which is a device used to make and receive calls over the internet. A SIP Server is also required, as it is responsible for handling the signaling and call setup processes in a VoIP network. Additionally, a Voice Gateway is needed to convert the analog voice signals into digital data that can be transmitted over the IP network. Therefore, the correct answer is IP Telephone, SIP Server, and Voice Gateway.Rate this question:
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- 33.
Voice can utilize the lowest bandwidth over IP-WAN Connections when using Codec ….
- A.
G711
- B.
GSM
- C.
G729
- D.
ILBC
Correct Answer
C. G729Explanation
G729 is the correct answer because it is a codec that is specifically designed to compress voice data, allowing it to be transmitted over IP-WAN connections using the lowest possible bandwidth. This codec uses a compression algorithm that reduces the amount of data needed to transmit voice, making it ideal for situations where bandwidth is limited or expensive. By utilizing G729, voice communication can be maintained efficiently and effectively over IP-WAN connections without consuming excessive bandwidth.Rate this question:
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- 34.
You did a site survey and found the customer has 2 sites, one main site at Nasr City with 100 users and the second site with 10 users at Rehab City. Which approach you will follow to implement a voice system with high availability and ensure minimal disruption? (Assuming both sites are connected over WAN)
- A.
Install 2 Voice Instances at Rehab City and let the users on Nasr City connect with Voice System.
- B.
Install 2 Voice Instances at Nasr City and let the users on Rehab City connect with Voice System.
- C.
Install 1 Voice Instance at Nasr City as an active instance and the other Instance at Rehab City as a backup instance, and let all users connect to the active instance at Nasr City.
- D.
Install 2 Voice Instances at Nasr City and let all users in both sites connect to it and additionally consider extra 1 Voice Instance System at Rehab City for failover.
Correct Answer
D. Install 2 Voice Instances at Nasr City and let all users in both sites connect to it and additionally consider extra 1 Voice Instance System at Rehab City for failover.Explanation
The best approach to implement a voice system with high availability and minimal disruption is to install 2 Voice Instances at Nasr City and let all users in both sites connect to it. Additionally, considering an extra 1 Voice Instance System at Rehab City for failover ensures that there is a backup in case the primary system fails. This setup allows for redundancy and ensures that users at both sites have access to the voice system without any disruption.Rate this question:
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- 35.
What does Virtualization Technology mean?
- A.
It refers to running multiple operating systems on a computer system simultaneously
- B.
It refers to dynamically distribute the voice calls on Linear ACD queue
- C.
It refers to dynamically distribute IP Addresses across managed network
- D.
It refer to host your application with a cloud service provider like AWS
Correct Answer
A. It refers to running multiple operating systems on a computer system simultaneouslyExplanation
Virtualization technology refers to running multiple operating systems on a computer system simultaneously. This means that a single physical computer can host and run multiple virtual machines, each with its own operating system. This allows for better utilization of hardware resources, increased flexibility, and improved efficiency in managing and deploying multiple operating systems on a single physical machine.Rate this question:
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- 36.
What does Virtualization Machine (VM) mean?
- A.
It refer to a Virtual System running on the top on another system with access to configured resources like RAM, CPU, Disk ..etc
- B.
It refers to a Physical Server with an Operating System running on your managed IP network.
- C.
It refers to purchasing a domain from domain provider (like GoDaddy) to build your own website
Correct Answer
A. It refer to a Virtual System running on the top on another system with access to configured resources like RAM, CPU, Disk ..etcExplanation
A Virtualization Machine (VM) refers to a virtual system that runs on top of another system. It has access to configured resources such as RAM, CPU, and Disk. This allows for the creation of multiple virtual machines on a single physical server, enabling efficient utilization of resources and providing isolation between different virtual systems.Rate this question:
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- 37.
One of the clients asks you to access the voice system applications from his home. What is the most secure option should you propose to him?
- A.
Assign Public IP Address to Voice Application directly and access it from Internet
- B.
Assign Public IP Address to Firewall and use port forwarding to map Public IP with the Voice Application Private IP
- C.
Advice Customer to use VPN, so he can connect to Voice Application from his Home
- D.
Ask the client to go to the his office whenever he need to access the Voice Application
Correct Answer
C. Advice Customer to use VPN, so he can connect to Voice Application from his HomeExplanation
The most secure option would be to advise the customer to use a VPN (Virtual Private Network) to connect to the Voice Application from his home. Using a VPN ensures that the connection between the client's home and the Voice Application is encrypted and secure, protecting sensitive data from unauthorized access. This option provides an additional layer of security compared to assigning a public IP address directly to the Voice Application or the Firewall, which could potentially expose the system to external threats. Asking the client to go to the office whenever they need to access the Voice Application is not practical and does not provide a secure solution.Rate this question:
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- 38.
On which of the following x64 editions of Windows Server 2016 does Hyper-V run? (Choose all that apply.)
- A.
Windows Server 2016 Web Edition
- B.
Windows Server 2016 Standard Edition
- C.
Windows Server 2016 Itanium Edition
- D.
Windows Server 2016 Datacenter Edition
Correct Answer(s)
B. Windows Server 2016 Standard Edition
D. Windows Server 2016 Datacenter EditionExplanation
Hyper-V is a virtualization platform that allows running multiple operating systems on a single physical server. It is available on the x64 editions of Windows Server 2016. The correct answer includes Windows Server 2016 Standard Edition and Windows Server 2016 Datacenter Edition. These editions of Windows Server 2016 support Hyper-V and provide the necessary features and capabilities for virtualization. Windows Server 2016 Web Edition and Windows Server 2016 Itanium Edition do not support Hyper-V.Rate this question:
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- 39.
Andy wants to change the memory of a virtual machine that is currently powered up. What does he need to do?
- A.
Shut down the virtual machine, use the virtual machine’s settings to change the memory, and start it again
- B.
Use the virtual machine’s settings to change the memory.
- C.
Pause the virtual machine, use the virtual machine’s settings to change the memory, and resume it
- D.
Save the virtual machine, use the virtual machine’s settings to change the memory, and resume it
Correct Answer
A. Shut down the virtual machine, use the virtual machine’s settings to change the memory, and start it againExplanation
To change the memory of a virtual machine, Andy needs to shut down the virtual machine first. Then, he can use the virtual machine's settings to make the necessary changes to the memory. Finally, he needs to start the virtual machine again for the changes to take effect.Rate this question:
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- 40.
You want to make sure the hard disk space for your virtual machines is occupied only when needed. What type of virtual hard disk would you recommend?
- A.
Physical or pass-through disk
- B.
Fixed-size disk
- C.
Differencing disk
- D.
Dynamically expanding disk
Correct Answer
D. Dynamically expanding diskExplanation
A dynamically expanding disk is recommended in this scenario because it only occupies the hard disk space when it is needed. This means that the virtual hard disk starts small and grows dynamically as more data is added to it. This allows for efficient utilization of storage space, as the disk only takes up as much space as it actually requires. It is a flexible and cost-effective option for virtual machines, as it allows for easy management of disk space and can save storage costs in the long run.Rate this question:
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- 41.
What devices convert voice into VoIP packets? (Choose TWO)
- A.
Hardware IP phone
- B.
Software phone
- C.
Analog phone
- D.
Cell Phone
Correct Answer(s)
A. Hardware IP phone
B. Software phoneExplanation
Hardware IP phones and software phones are devices that can convert voice into VoIP packets. Hardware IP phones are physical devices that are specifically designed for making VoIP calls. They have built-in hardware and software components that enable them to convert analog voice signals into digital packets that can be transmitted over an IP network. Software phones, on the other hand, are applications that can be installed on computers, smartphones, or tablets. They use the device's microphone to capture voice, which is then converted into VoIP packets using software algorithms. Both hardware IP phones and software phones play a crucial role in enabling voice communication over IP networks.Rate this question:
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- 42.
What is the purpose of the IP Phone registration process?
- A.
It provides details to the Registrar server so that the phone can be located on the network.
- B.
It provides details to the Relocation server so that the phone can be located on the network.
- C.
. It assigns the name, IP address, and MAC address to the phone
- D.
It determines a backup device to call if the phone is unavailable
Correct Answer
A. It provides details to the Registrar server so that the phone can be located on the network.Explanation
The purpose of the IP Phone registration process is to provide details to the Registrar server so that the phone can be located on the network. This allows the phone to establish a connection and communicate with other devices on the network. By registering with the Registrar server, the phone is assigned a unique identifier and its location is known, enabling proper routing of calls and messages.Rate this question:
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- 43.
What device in a VoIP environment is used to convert signaling from traditional telephony lines for transmission over the network?
- A.
Registrar Server
- B.
Proxy
- C.
Redirect Server
- D.
User Agent Client
- E.
Gateway
Correct Answer
E. GatewayExplanation
A gateway is a device used in a VoIP environment to convert signaling from traditional telephony lines for transmission over the network. It acts as a bridge between the traditional telephone network and the IP network, allowing communication between different types of networks. The gateway converts the analog signals from the traditional telephony lines into digital packets that can be transmitted over the IP network, and vice versa. This enables users to make and receive calls using their traditional telephones over the internet.Rate this question:
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- 44.
What is the threshold for packet loss in a VoIP environment?
- A.
>1%
- B.
>5%
- C.
- D.
Correct Answer
C.Explanation
In a VoIP environment, the threshold for packet loss is typically set at 1%. This means that if the percentage of lost packets exceeds 1%, it can result in degraded call quality and potential disruptions in the communication. Therefore, maintaining packet loss below this threshold is crucial to ensure a smooth and reliable VoIP experience. A higher threshold, such as 5%, would likely lead to more severe call quality issues and interruptions.Rate this question:
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- 45.
You have been assigned the task of researching audio codecs and were told to choose the one that would provide the highest audio quality. Bandwidth is not an issue as it is to be used within a single office. Which codec fits the request?
- A.
G729
- B.
G726
- C.
G723
- D.
G711
Correct Answer
D. G711Explanation
G711 fits the request because it is a codec that provides uncompressed audio, which means it does not compress or remove any data from the audio signal. This results in the highest audio quality, making it suitable for a scenario where bandwidth is not an issue. G729, G726, and G723 are all codecs that use compression techniques to reduce the size of the audio data, sacrificing some audio quality in the process.Rate this question:
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Our quizzes are rigorously reviewed, monitored and continuously updated by our expert board to maintain accuracy, relevance, and timeliness.
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Current Version
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Apr 06, 2023Quiz Edited by
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Aug 16, 2020Quiz Created by
Youssef
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