Free 100-890 Practice Test

The Cisco DX80 is an all-in-one desktop collaboration device that features a 23-inch Liquid Crystal Display (LCD). The display offers a Full High Definition (FHD) resolution of 1920 x 1080 pixels, which corresponds to a 16:9 aspect ratio. This integrated display is designed to provide a high-quality experience for video conferencing and content sharing directly on the user's desk, combining the functionality of a monitor, camera, speakers, and microphone into a single unit.

A. This is incorrect. The Cisco DX80 features a 23-inch screen, not a 12-inch screen.
B. This is incorrect. The Cisco DX70 model has a 14-inch screen; the DX80 model is larger at 23 inches.
D. This is incorrect. The DX80 has a 23-inch 1080p screen. The 27-inch 4K specification belongs to the Cisco Webex Desk Pro.

1. Cisco. (2021). Cisco DX80 Data Sheet. Cisco Public Information. Retrieved from Cisco.com. In the "Product specifications" section, Table 1, under the "Display" feature, it explicitly states: "23-inch (0.58m) LCD monitor" and "Resolution: 1920 x 1080 (16:9)".

The Cisco Webex Room 55 Dual is an integrated video conferencing system. Its core components include the Cisco Webex Codec Plus and the Cisco Webex Quad Camera. The Codec Plus provides the system's input and output capabilities, which include support for three microphone inputs and two HDMI video outputs for the dual displays. The integrated Quad Camera is a key feature, providing intelligent views like speaker tracking through its four-lens system. These two options accurately describe the technical capabilities of the system's fundamental components.

A. This is incorrect. The integrated Codec Plus supports three microphone inputs, not two.
B. While a wheel kit is available for the floor stand, it is an optional accessory rather than a fundamental component or feature of the base system.
D. This is incorrect. The "Dual" in the product name signifies that it is equipped with two 55-inch displays, not a single one.

1. Cisco Webex Room 55 and Room 55 Dual Data Sheet.

For Answer C (I/O): In the "Product specifications" table, under the "Audio inputs (with Codec Plus)" section, it lists "3 microphone inputs". Under "Video outputs (with Codec Plus)", it lists "2 HDMI outputs". This directly supports the specification of three microphone inputs and two video outputs.

For Answer E (Camera): In the "Features and benefits" section, under "Smart meetings", the document states the system "Brings intelligent views to the video conference room with discreet, integrated cameras. Best overview and speaker tracking capabilities." The "Components" section explicitly lists the "Cisco Webex Quad Camera" as the camera unit, which contains four lenses.

For Answer D (Display): In the "Product specifications" table, under the "Display" section, it specifies, "Two 55-in. (1.4 m) LCD monitors". This refutes the claim of a single display.

For Answer B (Mounting): In the "Ordering information" section, the document lists a "Wheel kit for floor stand (CS-ROOM55-W-KIT=)" as a separate, optional item, not a primary configuration or component.

While current official Cisco documentation specifies a 12-element microphone array for the Webex Desk Pro, earlier technical presentations from the product's initial launch period state it contains an 8-microphone array. Given that 12 is not an option and 8 is, the question most likely refers to this earlier specification. The multi-microphone array is a key feature that enables audio beamforming, which isolates the speaker's voice and reduces background noise for clearer communication.

A. 2: Incorrect. The Webex Desk Pro utilizes a complex multi-element array for advanced audio features like beamforming, which requires significantly more than two microphones.
B. 4: Incorrect. This number does not correspond to any official specification, past or present, for the internal microphone array of the Cisco Webex Desk Pro.
C. 6: Incorrect. There is no official Cisco documentation that specifies a 6-microphone array for this device. This number is not associated with the product's hardware specifications.

1. Supporting Answer (D):

Cisco Live Presentation, BRKCOL-2899, "Deploying the Latest Generation of Cisco Webex Rooms Devices" (2019). Slide 29, in the "Webex Desk Pro" feature list, explicitly states: "8 mic array for beamforming". This historical document supports the choice of 8 from the given options.

2. Contrasting with Current Specification (for clarity):

Cisco, "Cisco Webex Desk Pro Data Sheet". In the "Features and benefits" table, under the "Audio" feature, the document states: "The advanced 12-element microphone array with AI-powered audio features provides exceptional sound quality." This shows the specification has been updated since the product's launch, and the exam question is likely based on older data. (Document ID: C78-741933-13, Published: 12-Mar-2024).

The Cisco TelePresence Management Suite Provisioning Extension (TMSPE) is an optional software feature that extends the capabilities of Cisco TMS. Its primary function is to streamline and centralize the provisioning of users and video conferencing endpoints. TMSPE allows administrators to manage user accounts, create and distribute phone books, and apply configuration templates to devices, often by integrating with Cisco Unified Communications Manager (CUCM). This enables features like self-service provisioning for users, ensuring consistent configurations and simplifying large-scale deployments.

A. This describes the function of the Cisco TMS Booking API, which is a separate component designed to allow external applications and developers to interact with the scheduling system.
B. This functionality is provided by the Cisco TMS Extension for Alternative Endpoints (TMSAE), which is specifically designed to integrate and manage third-party or unmanaged conferencing systems.
D. This describes the Cisco TelePresence Management Suite Extension for Microsoft Exchange (TMSXE), which integrates TMS with Microsoft Exchange to enable meeting scheduling directly from Microsoft Outlook clients. ---

1. Cisco TelePresence Management Suite Provisioning Extension Deployment Guide (15.0), Introduction, Page 5. "Cisco TMSPE is a Cisco TMS extension that allows administrators to provision users for video conferencing from Cisco Unified Communications Manager. Cisco TMSPE provides a single, scalable, and centralized point of administration for video self-provisioning." This document explicitly details the provisioning functions of TMSPE, aligning with the correct answer.

2. Cisco TelePresence Management Suite Extension for Microsoft Exchange Deployment Guide (5.10), Introduction, Page 6. "The Cisco TelePresence Management Suite Extension for Microsoft Exchange (Cisco TMSXE) integrates Cisco TMS with a Microsoft Exchange server, enabling users to schedule meetings with Cisco video conferencing systems and rooms from their Microsoft Outlook clients." This source confirms that option D describes TMSXE, not TMSPE.

3. Cisco TelePresence Management Suite Extension for Alternative Endpoints Deployment Guide (1.2), Introduction, Page 5. "The Cisco TelePresence Management Suite Extension for Alternative Endpoints (Cisco TMSAE) allows Cisco TMS to manage and schedule calls that include third-party systems." This source confirms that option B describes TMSAE, not TMSPE.

4. Cisco TelePresence Management Suite API Guide (15.0), Introduction, Page 7. "The Cisco TMS Booking API allows external applications to book conferences through Cisco TMS." This source confirms that option A describes the Booking API, not TMSPE.

Cisco Unified Communications Manager (CUCM) is the core call-processing and session-management component of the Cisco Collaboration architecture. It operates within an IP-based network, managing signaling and call control for endpoints like IP phones and VoIP gateways. This functionality is fundamentally based on packet telephony, where voice is digitized, encapsulated into IP packets, and transported over a data network. CUCM establishes, maintains, and terminates calls between these packet-based endpoints, making it a packet telephony network device.

A. as a media switched telephone network device: CUCM is primarily a signaling and call control device (a call agent). The actual media (RTP stream) is typically switched directly between endpoints, not through the CUCM server itself.
C. as a circuit-switched telephone network device: Circuit-switching is a legacy technology used by the traditional PSTN, which establishes a dedicated, end-to-end circuit for a call. CUCM operates on modern packet-switched IP networks.
D. as a public switched telephone network device: CUCM is an enterprise-grade, private system. While it interfaces with the Public Switched Telephone Network (PSTN) via gateways, it is not a component of the PSTN itself.

1. Cisco Unified Communications Manager System Guide, Release 14: "Cisco Unified Communications Manager is the powerful call-processing component of the Cisco Unified Communications solution. Cisco Unified Communications Manager software extends enterprise telephony features and functions to packet telephony network devices such as IP phones, media processing devices, VoIP gateways, and multimedia applications."

Source: Cisco Official Documentation.

Reference: Cisco Unified Communications Manager System Guide, Release 14, "System Overview" chapter, "Cisco Unified Communications Manager" section.

2. Cisco Collaboration System 12.x Solution Reference Network Designs (SRND): "Cisco Unified Communications Manager (Unified CM) provides session and call control for the Cisco Collaboration solution. It is the core component for call and session management... Unified CM routes calls and sessions between endpoints registered to it and between gateways to other networks such as the PSTN." This description firmly places it in the context of a packet-based IP network.

Source: Cisco Official Documentation (SRND).

Reference: Cisco Collaboration System 12.x Solution Reference Network Designs (SRND), "Core Applications" chapter, "Cisco Unified Communications Manager" section.

3. Voice over IP Fundamentals, 2nd Edition (Cisco Press): This foundational text explains the architecture of VoIP systems, distinguishing between packet-switched voice networks and circuit-switched PSTN. It identifies call agents like CUCM as the central intelligence in a packet telephony system, responsible for call setup and feature control.

Source: Official Vendor Documentation (Cisco Press).

Reference: Voice over IP Fundamentals, 2nd Edition, Chapter 1, "Introduction to Voice over IP," section on "VoIP and Traditional Voice."

Cisco Unified Communications Manager (CUCM) serves as the core call control and session management component within the Cisco Collaboration architecture. Its primary signaling and device control function is to process calls by establishing, maintaining, and terminating sessions between endpoints. This includes call setup, where it handles digit analysis, routing, and signaling to connect the call, and call tear-down, where it manages the signaling required to properly end the session and release resources. The actual media (audio/video) stream, however, is typically exchanged directly between the endpoints (peer-to-peer) and does not flow through the CUCM call processing agent.

A. quick dialing: This is a user-facing convenience feature built upon the core call processing logic, not the fundamental signaling function itself.
B. phone book service: This is a directory function. While CUCM integrates with directory services, providing the phone book is not its primary signaling and control task.
E. transmission of audio/video media between endpoints: CUCM negotiates the media path via signaling, but the actual RTP media stream flows directly between endpoints, not through the CUCM server.

1. Cisco Collaboration System 12.x Solution Reference Network Design (SRND) Guide, Chapter: "Collaboration Solution Components," Section: "Cisco Unified Communications Manager (Unified CM)."

This document states, "Unified CM is the core component for call control in a Cisco Collaboration solution... Its fundamental role is to process calls between endpoints... This call processing function includes tasks such as call setup, routing, and teardown..."

2. Cisco Unified Communications Manager System Guide, Release 14, Chapter: "System-Level View of Cisco Unified Communications Manager," Section: "Cisco Unified Communications Manager."

This guide explains, "Cisco Unified Communications Manager provides functions such as call processing, signaling, and device control... The call processing component sets up and tears down calls between endpoints..."

3. Implementing and Operating Cisco Collaboration Core Technologies (CLCOR) 350-801 Official Cert Guide, Chapter 2: "Cisco Unified CM Architecture," Section: "CUCM Core Functions."

This official guide (published by Cisco Press, an approved vendor source) details that CUCM's core functions are signaling and call control, which encompass the entire lifecycle of a call from setup to tear-down, while the media path is established separately.

Video conferencing solutions developed before the 1990s were largely impractical for widespread use due to two primary factors: exorbitant cost and poor technical performance. The systems required specialized, bulky hardware and dedicated, high-capacity transmission lines (e.g., T1 or multiple ISDN lines), which were extremely expensive to lease and operate. Concurrently, the compression and transmission technology of the era was limited, resulting in low-resolution, low-frame-rate video. This produced a jerky, unclear, and often unreliable image, which significantly undermined the user experience and the effectiveness of the communication, making it an unviable solution for most organizations.

C. They produced high-resolution image quality. Early systems were severely limited by bandwidth and processing power, resulting in very low-resolution, not high-resolution, images.
D. They had not been invented yet. The concept and working prototypes, such as AT&T's Picturephone, existed as early as the 1960s, well before the 1990s.
E. They operated at a high frame rate. These systems operated at very low frame rates (often below 15 fps), which caused unnatural, jerky motion, not a smooth, high frame rate.

1. Noll, A. M. (1992). Anatomy of a failure: Picturephone revisited. Telecommunications Policy, 16(4), 307-316.

Section: "Picturephone I System" and "Economic Considerations" (pp. 308-312). This article details the technical limitations, including the "small, coarse black-and-white image," and the prohibitive costs, stating the service was "very expensive" and required costly dedicated transmission lines, which were major factors in its commercial failure.

DOI: https://doi.org/10.1016/0308-5961(92)90032-U

2. Fischer, C. S. (1992). America Calling: A Social History of the Telephone to 1940. University of California Press.

Chapter 7: The Picturephone (in later editions or related works by the author on the topic). Fischer's historical analysis of telecommunications technology discusses the failure of early videophones, attributing it to the high price point for consumers and businesses, and the low quality of the video that did not justify the expense.

3. Cisco Press. (2015). CCNA Collaboration CICD 210-060 Official Cert Guide.

Chapter 1: Introduction to Voice and Video Collaboration. While focusing on modern systems, the historical context provided in introductory sections often acknowledges the limitations of earlier technologies. It notes that the evolution to modern IP-based systems was driven by the need to overcome the high costs of legacy PSTN and ISDN circuits and the poor quality of early video codecs. This implicitly supports that high cost and low quality were the problems to be solved.

The Cisco Meeting Server (CMS) 1000 and 2000 platforms are designed to support single-server, all-in-one deployments. Official Cisco documentation for a "Single Combined Server Deployment" explicitly defines the standard set of co-located core components. This configuration includes the Call Bridge, XMPP Server, Database, and the Web Bridge. The Web Bridge is the essential component that serves the web application to browsers, enabling WebRTC clients (like the Cisco Meeting App) to join meetings. It works directly with the XMPP server for signaling and the Call Bridge for media, making it a fundamental part of a complete single-server solution.

A. Turn Server: While the TURN Server can be co-located, it is a supplementary component required only for specific network scenarios involving NAT traversal, not a fundamental part of the core functional group.
B. Load Balancer: The load balancer is a logical function for distributing traffic in scaled, multi-server deployments and is not a software component that runs on the CMS 1000/2000 appliance itself.
D. H.323 Gateway: This is an optional interoperability component for connecting with legacy H.323 endpoints. It is not considered part of the standard core component set for a typical single-server deployment. ---

1. Cisco, "Cisco Meeting Server Deployment Guide for Single Combined Server Deployments (3.1)": In the "About this deployment" section, the guide explicitly states: "A single combined deployment is where all the Cisco Meeting Server components (Call Bridge, Web Bridge, Database, and XMPP server) are installed on one server." This directly identifies the Web Bridge as the fourth core component in this standard configuration.

(Reference: Cisco Meeting Server Deployment Guide for Single Combined Server Deployments 3.1, Chapter: Introduction, Section: "About this deployment")

2. Cisco, "Cisco Meeting Server 1000 Data Sheet": The product overview describes the intended use of the appliance: "The Cisco Meeting Server 1000 is a single-server solution for smaller deployments. It runs all the required components on a single Cisco UCS C220 M5 Rack Server: Call Bridge, Web Bridge, and XMPP server." This further reinforces that the Web Bridge is a primary, expected component for this platform.

(Reference: Cisco Meeting Server 1000 Data Sheet, Section: "Product overview")

3. Cisco, "Cisco Meeting Server API Reference Guide (3.8)": The guide lists the configurable components that can be enabled on a server. While turn and h323gateway are listed as valid, deployable components, the deployment guides (References 1 & 2) specify the core group for a single-server solution, which centrally includes webbridge.

(Reference: Cisco Meeting Server API Reference Guide 3.8, Section: "API objects", Subsection: "servers / server / components")

Cisco Unified Communication Manager (CUCM) supports two primary signaling protocols for its IP phone endpoints. The first is the Skinny Client Control Protocol (SCCP), a lightweight, Cisco-proprietary protocol originally developed for Cisco IP phones. The second is the Session Initiation Protocol (SIP), an open-standard IETF protocol that has become the industry standard for VoIP signaling. CUCM can register and manage Cisco IP phones running either SCCP or SIP firmware, allowing administrators to choose the protocol that best fits their network and feature requirements.

B. H.323: H.323 is a signaling protocol suite primarily used by CUCM for communication with gateways and for trunks to other IP-PBX systems, not for IP phone registration.
C. MGCP: Media Gateway Control Protocol (MGCP) is used by CUCM to control and manage media gateways, such as analog voice gateways, not for IP phone signaling.
D. RTP: Real-time Transport Protocol (RTP) is a transport protocol used to carry the actual voice and video media streams, not a signaling protocol for call setup or control.

1. Cisco Collaboration System 12.x Solution Reference Network Design (SRND)

Reference: Chapter: "Unified CM Endpoints", Section: "Endpoint Protocols"

Content: "Cisco Unified CM supports a wide variety of endpoint types, including Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP), and Java/XML and HTML-based applications. Most Cisco endpoints are available with either SIP or SCCP firmware loads." This directly confirms that SIP and SCCP are the supported protocols for endpoints (phones).

2. System Configuration Guide for Cisco Unified Communications Manager, Release 14

Reference: Chapter: "Cisco Unified IP Phone Configuration", Section: "Phone Configuration Checklist"

Content: The guide details the configuration steps for adding new phones to CUCM. In the "Phone Configuration" window, a mandatory field is "Phone Button Template," which is associated with a specific protocol (SCCP or SIP). Furthermore, the "Device Protocol" field for many phone models explicitly allows the selection of either SIP or SCCP, confirming both are supported.

3. Cisco Unified Communications Manager Administration Guide, Release 12.5(1)SU1

Reference: Chapter: "Device Settings", Section: "Phone Button Template Configuration"

Content: "Phone button templates are associated with a specific phone model and protocol (SCCP or SIP). Cisco Unified Communications Manager comes with a default set of phone button templates for each phone model and protocol." This statement from the official administration guide confirms that CUCM manages phones based on either the SCCP or SIP protocol.