CCT Exam Practice Test Questions and Answers

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The Cisco Webex Room 70 G2 supports 8 microphone inputs.

The TelePresence SX10 Quick Set is considered an all-in-one unit designed to video-enable small collaboration spaces. It is a compact and easy-to-use endpoint that combines a camera, codec, and microphone into a single device, making it ideal for small meeting rooms or huddle spaces. The SX10 Quick Set offers high-quality video and audio capabilities, allowing users to have seamless video conferences in small collaborative environments.

Performing a factory reset on the IP phone is the next best approach to try and recover the phone. This can help to resolve any software or configuration issues that may be preventing the phone from completing the boot-up process. By resetting the phone to its factory default settings, it will clear any potential issues and allow the phone to start up properly.

The ultimate sign that an ethernet cable is not working is when the problem is resolved by replacing the cable. This indicates that the original cable was faulty or damaged, and replacing it solves the connectivity issue. The other options mentioned in the question, such as the cable losing connection, intermittent and slow connection, and connection being restored with a wiggle, are all symptoms of a faulty cable, but replacing the cable is the definitive solution to the problem.

The component shown in the picture that can be found in the Cisco Webex Room 70D system left screen is the Codec.

Subnetting helps to reduce network traffic by dividing a large network into smaller subnets, which allows for more efficient use of network resources. Each subnet operates as its own broadcast domain, meaning that broadcasts are contained within the subnet and not propagated to other subnets. Therefore, the statement "Network traffic is reduced" is true. Additionally, subnetting does indeed equal to a broadcast domain, as each subnet has its own unique network address and broadcasts are limited within that subnet.

When switch port security is enabled and configured, it prevents unauthorized devices from accessing the network by blocking packets with source addresses outside the defined group of addresses. This means that only devices with MAC addresses that have been configured or learned on the secure port will be allowed access. Additionally, if the switch is configured to limit the number of secure MAC addresses to one and assign a single MAC address to a workstation attached to that port, it guarantees that the workstation will have the full bandwidth of the port available to it.

A gateway is a VoIP component that provides translation between VoIP and non-VoIP networks. It acts as a bridge between different types of networks, allowing communication between them by converting the protocols and formats used. In the context of VoIP, a gateway enables communication between VoIP networks and traditional telephone networks, allowing calls to be made and received between the two.

Intelligent Proximity is used for content sharing. This feature allows users to wirelessly share their screen or presentations with other devices, such as projectors or displays. It enables seamless and efficient collaboration by eliminating the need for cables and physical connections. With Intelligent Proximity, users can easily share their content and engage with their audience in a more interactive and dynamic way.

The Cisco TMS Network Integration Extension (NETINT) enables Cisco TMS to integrate Cisco and third-party unmanaged conferencing bridges. This means that the NETINT allows Cisco TMS to connect and manage video conferencing systems from different vendors, providing a centralized management platform for administrators. It streamlines the process of integrating and managing conferencing bridges, making it easier for organizations to set up and maintain their video conferencing network.

The Cisco Meeting Server 2000 does not support additional core components including TURN Server, Load Balancer, Recording and Streaming, and H.323 Gateway because it is primarily designed to handle a large number of calls. These additional components are not necessary for the intended purpose of the server, which is to handle a high volume of calls efficiently.

The Cisco DX80 offers a screen size of 23 inches with a 16:9 aspect ratio.

SIP (Session Initiation Protocol) and H.323 are both peer-to-peer signaling protocols where the end devices have the capability to initiate and terminate calls and understand call control messages. These protocols allow for communication between devices and enable the establishment, modification, and termination of sessions. SCCP, MGCP, and H.248 are not peer-to-peer signaling protocols and do not possess the same level of intelligence in end devices for call initiation and termination.

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The Cisco voice gateway is not used for powering up Cisco IP phones. Cisco voice gateways are devices that connect traditional telephony networks, such as the public switched telephone network (PSTN), to IP-based networks. They are responsible for converting voice signals into digital packets for transmission over the IP network. However, they do not provide power to Cisco IP phones. Cisco IP phones can be powered using a Cisco switch with a Power over Ethernet (PoE) port, a TelePresence codec, or an external power supply.

The correct answer is H.323. H.323 is a communication protocol that was commonly used by TANDBERG's T3 endpoint system. It is a standard protocol for audio, video, and data communication over IP networks and is specifically designed for videoconferencing systems. H.323 provides a set of protocols for call setup, control, and management, as well as for audio and video compression. It has been widely adopted in the industry and is still used in many video conferencing systems today.

H.264 uses the common compression method called "lossy." This means that when encoding video files, H.264 discards some of the less important information in order to reduce the file size. This compression method allows for efficient storage and transmission of video files while maintaining acceptable quality.

The Cisco Webex Room 70 Dual G2 supports both a people-focused experience and a content-focused experience. This room system allows for high-quality video and audio conferencing, making it ideal for collaboration and communication. It also has the capability to display and share content, such as presentations or documents, during meetings. With its dual screens and advanced features, the Cisco Webex Room 70 Dual G2 provides a versatile and immersive experience for both participants and content sharing.

The Cisco Unified Communication Manager functions as a packet telephony network device in the enterprise VoIP telephony infrastructure. This means that it is responsible for handling the transmission of voice data over packet-switched networks, such as IP networks. It converts analog voice signals into digital packets and vice versa, allowing for efficient and cost-effective communication over IP networks.

The Cisco Unified Communication Manager supports SIP and H.323 signaling protocols for the Cisco IP phones. These protocols enable the communication between the IP phones and the Unified Communication Manager, allowing for features such as call setup, call control, and media transmission. SIP (Session Initiation Protocol) is a widely used protocol for initiating, modifying, and terminating real-time sessions involving video, voice, messaging, and other communications applications and services. H.323 is an older protocol that provides similar functionality for multimedia communication over IP networks.

The Cisco Unified Communication Manager provides limited conferencing capabilities through the Cisco IP Voice Media Streaming Application Service. It also supports conferencing using DSP (Digital Signal Processors) resources.

To find the IP address of the Cisco Webex Room 70D system, the user should tap on the system name in the upper left corner of the Touch 10 controller screen. Then, they should select "Settings" and navigate to "About this device". This option will provide the necessary information about the IP address of the system.

The Cisco Unified Communication Manager signaling and device control function provides two features: Call Setup and Call Tear Down. Call Setup refers to the process of establishing a connection between two or more parties in a call. It includes tasks such as call routing, address translation, and call admission control. Call Tear Down, on the other hand, refers to the process of terminating a call and releasing the associated resources. It involves tasks such as releasing the call signaling and media channels, updating call status, and freeing up network resources. These two features are essential for the proper functioning of a communication system.

Cisco Meeting Server supports third party devices such as Microsoft Skype for Business.

When the issue is clearly marked red on the Diagnostics page, it means that there are errors that are classified as critical. This indicates that there are significant problems or malfunctions that need immediate attention and resolution. The red marking serves as a clear indication to the user that urgent action is required to address the identified errors and prevent any further issues or disruptions.

In the web interface, the detailed IP information of Webex Endpoint Devices can be found in the "Status" option of the "Setup" tab.

After replacing a Webex endpoint device codec, three common tasks that need to be performed for operation verification are: Test Call, Camera Operation, and Content Sharing. A test call is necessary to ensure that the new codec is functioning properly and can establish a connection. Camera operation needs to be checked to ensure that the video feed is working correctly. Content sharing should be tested to verify that the device can display and share content effectively. Adjusting display settings and conducting a network connection speed test are not mentioned as necessary tasks for operation verification after replacing the codec.

To transfer the license and generate the option key for the Cisco Webex DX80, the customer administrator needs to provide the serial numbers of the new DX80 and the replaced DX80, as well as the RMA number. These details are necessary to ensure that the license is correctly transferred from the old device to the new one and to generate the option key for activating optional functionalities such as MultiSite on the Cisco Webex DX80. The DX80 part number, current option key, and customer license account number are not required for this specific process.

The Cisco IP phone learns the voice VLAN ID via Cisco Discovery Protocol (CDP) from the switch. CDP is a Cisco proprietary protocol that allows network devices to learn information about each other. In this case, the IP phone uses CDP to communicate with the switch and obtain the voice VLAN ID, which is necessary for prioritizing voice traffic on the network.

The Cisco Meeting Server 1000 and 2000 support the Web Bridge core component in addition to the Call Bridge, XMPP Server, and Database core components. The Web Bridge allows users to join meetings using a web browser, without the need for any additional software or plugins. This enables seamless and convenient access to meetings from any device with a web browser, enhancing the overall user experience and increasing accessibility.

The Cisco IX5000 supports 2 1080-p, 30-fps content-sharing streams.

The default username and passphrase for most Webex Endpoint Devices is "admin" and "blank". This means that when the device is first set up, the administrator can leave the passphrase field empty. This is a common default setting to make it easier for the administrator to access the device initially, but it is recommended to change the passphrase to a secure one to ensure the device's security.

The Cisco Meeting Server 1000 and Cisco Meeting Server 2000 both support the SIP protocol, which is a standard protocol used for initiating, maintaining, modifying, and terminating real-time sessions that involve video, voice, messaging, and other communications applications and services. Additionally, both servers can be deployed on-premises as either an appliance or a virtual machine (VM) application, allowing for flexibility in deployment options.

Cisco Meeting Server 1000 and 2000 are both on-premises conferencing platforms. They are not preconfigured as virtualized servers, but Cisco Meeting Server 1000 runs on a preconfigured UCS 5108 blade server. Additionally, Cisco Meeting Server supports both H.323 and SIP protocols.

The CTL file contains a list of Cisco Unified CM and TFTP servers that the IP phone is authorized to connect to. This information is important for the boot-up process of the Cisco IP phone as it allows the phone to establish a secure connection with the authorized servers. By defining the authorized servers in the CTL file, the IP phone can ensure that it is connecting to trusted and authorized sources for communication and configuration updates.

By selecting Settings> About this device from the Endpoint Device Controller, you can find the basic IP address of Webex Endpoint Devices. This option allows you to access the device settings and view detailed information about the device, including its IP address.

The Cisco Webex Room 55 Dual supports 4x lenses, speaker track, 3 microphone inputs, and 2 video outputs. This means that it is capable of using four lenses for enhanced video quality, has a speaker tracking feature to focus on the active speaker, can connect three microphones for better audio input, and has two video outputs for connecting to multiple displays or devices.

The correct answer is "Packet Loss is a relative measure of the number of packets that were not received compared to the total number of packets transmitted." This statement accurately describes packet loss as a measure of the percentage of packets that were not successfully delivered compared to the total number of packets sent.

The correct answer is "Snap to Whiteboard." This feature allows the camera of the Webex device to automatically frame both the person and the whiteboard when a person next to the whiteboard is speaking. It ensures that the camera focuses on the relevant content and provides a clear view of both the speaker and the whiteboard during presentations or discussions.

To validate the certificates of external servers (HTTPS and syslog), a trusted Certificate Authority (CA) is needed. The "PreInstalled" CA refers to a certificate authority that is already installed and trusted by default on the system. This means that the certificates issued by this CA will be automatically recognized and trusted by the system when validating the certificates of external servers. Therefore, the "PreInstalled" CA is the correct choice for validating the certificates of external servers.

The Cisco Unified Communication Manager functions as a packet telephony network device in the enterprise VoIP telephony infrastructure. This means that it is responsible for handling the transmission of voice packets over the network, ensuring clear and reliable communication for users. It manages call signaling, call control, and call routing, making it an essential component in a VoIP system.

Operators can use Cisco Meeting Management to monitor and manage meetings on Cisco Meeting Server. Cisco Meeting Management provides a comprehensive set of tools and features that allow operators to schedule, start, monitor, and manage meetings on the Cisco Meeting Server platform. It allows operators to view real-time meeting statistics, manage participant lists, control meeting access, and troubleshoot any issues that may arise during a meeting. Cisco Meeting Management provides a user-friendly interface that makes it easy for operators to navigate and perform their meeting management tasks efficiently.

The correct answer states that five speakers need to be removed in order to replace the Cisco TelePresence MX800 Dual system camera. These speakers are located at position 4, 5, and 6 above the left monitor, and position 2 and 3 above the right camera.

When an endpoint fails to register to Cisco Unified CM, two possible causes for MAC address mismatch are the endpoint being unplugged and the MAC address being entered incorrectly. If the endpoint is unplugged, the registration process cannot take place, leading to a mismatch in the MAC address. Similarly, if the MAC address is entered incorrectly, the registration will fail and result in a mismatch. The other options, such as the gateway address missing or DNS server not available, do not directly impact the MAC address registration process.

The Infrastructure collaboration components provide Quality of Service and Security mechanism. Quality of Service ensures that the network can prioritize certain types of traffic, such as voice or video, to ensure a high level of performance. Security mechanisms are implemented to protect the infrastructure from unauthorized access and ensure the confidentiality, integrity, and availability of data and services. Instant messaging and voice-mail systems are not specifically mentioned as being provided by the Infrastructure collaboration components.

Endpoints can register to the Cisco Office Manager. This product allows users to manage and control their Cisco endpoints, such as phones and video devices, from a single interface. It provides a centralized platform for configuring and monitoring these endpoints, making it easier for administrators to manage their Cisco devices efficiently. Cisco Webex is a separate product that offers online meetings, video conferencing, and collaboration tools, but it is not specifically designed for endpoint registration. Cisco Collaboration is a broader term that encompasses various Cisco products and solutions for communication and collaboration, but it does not specifically refer to endpoint registration.

In the web interface, the detailed IP information of Webex Endpoint Devices can be found in the Status option of the Setup tab.

The Cisco Webex Desk Pro supports 6 microphone locations.

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The Cisco Meeting Server and Cisco Meeting Apps have been optimized to be deployed with Cisco Unified Communications Manager. These solutions are designed to enhance collaboration and communication within an organization by providing seamless integration with the Unified Communications Manager platform. By optimizing these tools for deployment with Unified Communications Manager, Cisco aims to provide a comprehensive and efficient communication ecosystem for businesses.

The IM and Presence Service integrates into a full Cisco Unified Communications environment, meaning that it can seamlessly work with other Cisco Unified Communications components. Additionally, it supports the Cisco Jabber platform, allowing users to use the Jabber client for instant messaging and presence features.

The Cisco TMS Provisioning Extension (TMSPE) allows video conferencing network administrators to create and manage user accounts, phone books, configuration templates, and address patterns. This means that administrators can control and customize various aspects of the video conferencing network, such as adding and managing user accounts, organizing phone books, creating configuration templates for different scenarios, and defining address patterns for easier dialing. This feature enhances the functionality of Cisco TMS by providing more control and flexibility to network administrators.

IM and Presence Service supports three modes of operations.

The Cisco TelePresence Management Suite (TMS) is a portal that allows users to manage and monitor their video conferencing network. It provides centralized control for both on-site and remote video systems, allowing for easier management and configuration. Additionally, TMS can integrate with Microsoft Exchange and Outlook, enabling flexible meeting scheduling and enhancing the overall user experience. The complexity of a TMS installation is not determined by the number of Cisco Unified CM in a Cisco Unified CM cluster, so this statement is not applicable.

A gatekeeper in a VoIP network is responsible for providing Call Admission Control (CAC), which ensures that the network has enough resources to handle incoming calls. It also controls and manages the bandwidth to ensure efficient use of network resources. Additionally, a gatekeeper performs address translation, allowing devices with different addressing schemes to communicate with each other. Therefore, the gatekeeper is the correct answer as it encompasses all the mentioned functionalities.

For a successful and secured FTP operation between a Cisco router and a PC, two requirements are necessary. Firstly, there must be sufficient space on the router's Flash memory to accommodate the copied file. This ensures that the file can be transferred without any storage issues. Secondly, a username and password are required to authenticate the FTP session, providing security and ensuring that only authorized users can access and transfer files.

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The correct answer is "xstatus audio microphones". This command is likely to provide more information on the microphones. The "xstatus" command is commonly used to check the status or obtain detailed information about various components or settings. In this case, using "xstatus audio microphones" would specifically retrieve information about the microphones.

The Cisco IX5000 with a dual row has a total of 18 seats available.

Dial peers define the virtual interfaces to and from which a call is established, meaning they specify the routing and signaling information for a call. Voice ports, on the other hand, physically connect the router to telephony devices such as PSTN central office switches, allowing the router to send and receive voice traffic. These two statements accurately describe the roles of dial peers and voice ports in configuring Cisco IOS voice features.

The correct answer is "xconfigaudio microphones mute". This command is likely used to access the audio configuration settings and specifically mute the microphone.

The Cisco Catalyst PoE switch can provide both network connection and power to the IP phone, allowing it to function in the IP telephony network. The switch supports Power over Ethernet (PoE) technology, which enables it to deliver power to connected devices, including IP phones, through the Ethernet cables. The Cisco TelePresence codec, on the other hand, is a video conferencing device and not specifically designed to provide power to IP phones. However, it can still be used to connect an IP phone to the network, but an additional power source would be required for the IP phone to function properly.

The first correct phrase states that a TFTP client listens for write requests from a TFTP server for file download. This means that the client is actively waiting for the server to request a file transfer.

The second correct phrase states that TFTP uses UDP transport. UDP (User Datagram Protocol) is a connectionless protocol that is commonly used for fast and efficient transmission of data over a network. TFTP utilizes UDP for its simplicity and low overhead, making it suitable for applications where reliability is not as critical as speed.

The correct answer is IM and Presence and Cisco Meeting Server. These two applications are included in the Collaboration components. IM and Presence refers to instant messaging and presence status, allowing users to communicate in real-time and see the availability of others. Cisco Meeting Server is a video conferencing solution that enables users to have virtual meetings and collaborate remotely. The other options mentioned, such as Software Clients and Voice Messaging, may also be part of the Collaboration components, but they are not the specific applications mentioned in the question.

The IP phone power adapter with Ethernet port can provide both the network connection and power to the Cisco IP phone, allowing it to function in the IP telephony network. Additionally, the Cisco TelePresence codec can also provide the necessary network connection and power for the IP phone to operate.