CLTECH 100-890: Collaboration Environment Overview

Cisco Unified Communications

People work together in different ways and demand many collaboration tools, such as IP telephony for voice calling, web and video conferencing, voicemail, mobility, desktop sharing, instant messaging and presence, and so on. delivers integration of these tools, which help organizations meet rapidly changing technology as well as new engagement demands to stay competitive and engage with customers and employees across new and emerging ecosystems.

VoIP Functions

Signaling Database services Bearer control Codecs

Cisco TMS Extension for Microsoft Exchange (TMSXE)

enables Cisco TelePresence scheduling through Microsoft Outlook.

Cisco TMS Provisioning Extension (TMSPE)

is a provisioning application for Cisco TMS and Cisco TelePresence Video Communication Server (Cisco VCS). This provisioning application allows video conferencing network administrators to create and manage user accounts, phone books, configuration templates, and address patterns.

IP Phones

provide IP endpoints for voice communications.

Cisco Unified Communications Manager

provides limited conferencing capabilities through the Cisco IP Voice Media Streaming Application Service. The software conference bridge supports G.711 audio by default and has no support for video. Cisco IOS devices can support conferencing by using DSP (Digital Signal Processors) resources, which are hardware installed inside the router itself. The capabilities depend on the cards installed and the router platform. Cisco IOS hardware conferencing is also voice-only.

Multipoint Control Unit (MCU)

provides real-time connectivity for participants in multiple locations to attend the same videoconference or meeting.

Cisco Unified Communications Manager

serves as the software-based call control component of the Cisco Collaboration solution for on-premises deployment. The call control provides for call processing, device control, and administration of the dial plan and features to Cisco integrated telephony applications as well as to third-party applications. Cisco Unified CM replaces traditional private branch exchange (PBX) phone systems and extends enterprise telephony features and functions to packet telephony network devices such as Cisco IP phones, video conferencing endpoints, media-processing gateways, VoIP gateways, and multimedia applications. Cisco Unified CM supports the Session Initiation Protocol (SIP) and the Skinny Client Control Protocol (SCCP) for IP phones, and SIP and H.323 for video conferencing endpoints. SCCP is the Cisco proprietary IP phone protocol. SCCP is supported on the Cisco 8800 Series lP phones. Connections to gateways use SIP, H.323, or Media Gateway Control Protocol (MGCP). The clustering feature of Cisco Unified CM provides a mechanism for seamlessly distributing call processing across the infrastructure of a converged IP network. Clustering provides transparent sharing of resources and features and enables system scalability.

Call Control Consists

the call setup, call signaling, and media processing stages of a call and can have varying degrees of complexity depending on the needs of the user.

Database services

Access to services, such as toll-free numbers or caller ID, requires the capability to query a database to determine whether the call can be placed or information can be made available. include access to billing information, caller name delivery, toll-free database services, and calling-card services. VoIP service providers can differentiate their services by providing access to many unique database services. An example is providing a call notification service that places outbound calls with prerecorded messages at specific times to notify users of such events as school closures, wake-up calls, or appointments.

Cisco Meeting Server Components

Call Bridge: The Call Bridge is the main conferencing engine. It also provides the SIP interface, so that calls can be routed to or from the Call Bridge from external call control, such as the Cisco Unified Communications Manager and Avaya call control device. Web Bridge: Web Bridge is the server role that enables users to manage and connect to their conferencing spaces, as well as allow outside participants into conferences using only a WebRTC-enabled browser. XMPP Server: The XMPP service is used to handle all registration and authentication for Cisco Meeting Apps, including the Web Real-Time Communication (WebRTC) Cisco Meeting Apps client. Database: Allows some configuration, such as dial plan, spaces, and users to be aggregated. TURN Server: Traversal Using Relays Around NAT (TURN) is the server role that sits on the public internet that sends and receives media. It must be reachable by both the external devices on the internet and internal devices, so that audio and video traffic can flow into and out of an organization. Load Balancer: The Load Balancer component is used in a clustered Cisco Meeting Server deployment. It makes efficient use of conference resources and minimizes the number of distribution calls between the Call Bridges that host the same meeting space. Recording and Streaming: Meeting recording and streaming functionality. H.323 Gateway: The H.323 Gateway component converts H.323 messages into SIP for Cisco Meeting Server to handle the H.323 calls. Web Admin: Administration GUI and API access, including access for Cisco Unified CM ad hoc conferencing. The core components supported by Cisco Meeting Server 1000 and 2000 are Call Bridge, Web Bridge, XMPP Server, and Database. The core components supported by Cisco Meeting Server 1000 only are the TURN Server, Load Balancer, Recording and Streaming, and H.323 Gateway. The Cisco Meeting Server 2000 is designed to handle a large number of calls. To support this capability, only the Call Bridge, Web Bridge, and XMPP server, and Database components are available for configuration

Cisco Unified CM Primary Functions

Call processing: Call processing refers to the complete process of establishing calls, routing media channels, and terminating calls, including any billing and statistical collection processes. Signaling and device control: Cisco Unified CM sets up all call connections between call endpoints and directs devices such as phones, gateways, and conference bridges to establish and tear down streaming connections. Endpoint registration: Endpoints registered to the Cisco Unified CM are listed in a database mapping usernames and numbers to IP addresses. Cisco Unified CM can control endpoints registration and capabilities with configuration files. Directory services: Cisco Unified CM stores user information in its own database. It allows users to look up a directory of user numbers. User authentication is performed locally or against an external directory. Synchronization with an external directory such as the Microsoft Active Directory allows for centralized user management. Dial plan administration: The dial plan is a set of configurable lists on which Cisco Unified CM performs digit analysis to route the calls. Users can create scalable dial plans to suit individual needs. Phone Feature Administration: Cisco Unified CM extends services such as hold, transfer, forward, conference, speed dial, Call Park, Extension Mobility, and Device Mobility to endpoints and gateways. Bandwidth management: A feature that controls how much bandwidth a call can use. This bandwidth amount can be set for each call and controlled for each link. For example, calls between site 1 and site 2 cannot exceed a set limit.

IP Phone Registration Process Overview

Cisco IP phones use either the SIP or SCCP signaling protocol to communicate with Cisco Unified Communications Manager. Before the IP phone can make a call, the phone must register on the Cisco Unified CM and request configuration for the device capabilities and phone number. The figure provides an overview of the IP phone registration process.

Endpoints

Cisco is focused on making its collaboration technology easy, convenient, and beneficial to use, with emphasis on the following enhancements to the user experience: Phones: From basic single-line phones to video-capable, multiline, color display models to suit various customer requirements. Cisco TelePresence: Cisco TelePresence technology brings people together in real time without the expense and delay of travel. The Cisco TelePresence portfolio of products includes an array of high-definition video endpoints ranging from individual desktop units to large multiscreen immersive video systems for conference rooms. Software clients: Cisco Jabber for on-premises and Cisco Webex Teams for cloud-based customers.

Conferencing

Cisco offers both cloud-based and on-premises of this capabilites. Cisco Webex: A cloud-based solution incorporating audio, high-definition video, and real-time content-sharing in a platform that provides easy setup and administration of meetings, interactive participation in the meeting, and the ability to join the meeting from any type of device, such as an IP phone, tablet device, or a desktop computer. Cisco Meeting Server: On-premises video, audio, and web communication together in a single place.

Mobile endpoints

Collaboration endpoints for mobile endpoints includes Cisco Jabber and Cisco Webex Teams which are available for Android, iPhone, iPad, Windows.

Cisco Unified Communications Components

Communications gateways: The communications gateways perform various functions, including audio and video media termination and signal conversion, multipoint meeting control, session management, and translation between different VoIP networks or between VoIP and public switched telephone networks (PSTNs). These gateways provide a complete platform for integration into branch office, enterprise, campus, and service provider networks. Call control: The call control provides for call processing, device control, and administration of the dial plan and features. Call control devices include Cisco Unified Communications Manager (Cisco Unified CM) and Cisco Expressway. Unified Communications applications: The Cisco Unified Communications incorporates various advanced applications and services that integrate with Cisco Call Control applications. These applications increase productivity and customer experiences, including Cisco Jabber and Webex Teams for instance messaging and presence, Cisco Unity Connection for messaging and voicemail, and Cisco Contact Center for customer contact and services. Cloud calling: Cisco cloud-based conferencing solution incorporating audio, HD video, and real-time content-sharing that provides easy setup and administration of meetings. Cisco Webex offers calling capabilities from desktop phones and mobile devices. Cisco BroadCloud is a global platform hosted by Cisco that service providers can use to deploy a full suite of calling and collaboration applications. Telephony extensions: Cisco Unified Communications provides many enhanced telephony extensions to the IP phones, applications, and call controldevices. Extensions such as Cisco Office Manager and Cisco Voice Provisioning Tool streamline the routine operational tasks for the users, IP phones, and call control device management. Cisco Unified Call Connectors and Unified Quick Connect extensions provide the QuickDialing and Finding Contacts features, which facilitate collaboration with the easy-to-use interface. Cisco Unified Communications is available as on-premises software, as partner-hosted solutions, or as a service (UCaaS) from cloud providers.

Web Conferencing

Ease-of-use: If the web conferencing software is difficult to use—setting up meetings, signing into meetings, and holding the meetings—you will likely abandon the process before you can reap the benefits. Look for a simple user experience from any device that doesn't require plug-ins or downloads. Full set of features: Can you share screens during a web conference or video conference? Poll the audience? Use a whiteboard to brainstorm new ideas? Record the call? Being able to do more than hold an audio call can elevate your meetings to true collaboration. Security: It's vital to protect your conversations from prying ears and eyes. Be sure your web conferencing solution has built-in multilayer security that doesn't compromise the user experience.

Voice Gateways

Enable Cisco Unified Communications Manager (Cisco Unified CM) to communicate with non-IP telecommunications devices. Cisco Unified CM supports several types of Cisco IP telephony gateways. Gateways use call control protocols to communicate with the PSTN and other non-IP telecommunications devices, such as the PBX. Trunk interfaces specify how the gateway communicates with the PSTN or other external devices by using time-division multiplexing (TDM) signaling. Cisco Unified CM and Cisco gateways use a variety of TDM interfaces, and they vary by gateway model. TDM switching is supported on T1/E1/DS3 interfaces. DS0 channels used in TDM-switched calls can come from the same T1/E1 interface or from different T1/E1 interfaces in the gateway.

Client/Server Protcols/Signaling

H.248, SCCP, and MGCP are examples of this where the endpoints or gateways do not contain call control intelligence but send or receive event notifications to a server commonly referred to as a call agent. For example, when an MGCP gateway detects a telephone that has gone off hook, it does not know to automatically provide a dial tone. The gateway sends an event notification to the call agent, telling the agent that an off-hook condition has been detected. The call agent notifies the gateway to provide a dial tone.

End Point Protocols

In an on-premises solution, Cisco Unified Communications Manager (Cisco Unified CM) is used as the primary call control device. Various endpoints are supported by Cisco Unified CM, including Cisco products and third-party products. Endpoints include IP phones, video endpoints, and gateways, which allow connectivity to PSTN and SIP cloud-based networks. Cisco also offers software clients for mobile endpoints that are based on Android and Apple iOS operating systems and for personal computers (Windows and Mac). Cisco Unified Communications Manager supports SIP and SCCP protocols for phones and telepresence endpoints. Connections to gateways use the SIP, H.323, or MGCP protocol. SIP is the preferred protocol for both gateways and end-user devices. Cisco has two other call control devices: Cisco Unified Communications Manager Express and Cisco Expressway. Cisco Unified Communications Manger Express runs on a Cisco router and is designed for smaller scale on-premises solutions. Cisco Expressway primarily manages connectivity to external networks but can also be used to register SIP and H.323 phones and telepresence devices if required. Cisco endpoints can also be registered to Cisco Webex Cloud services. Customers can also adopt a hybrid solution using both on-premises call control and Cisco Webex Cloud services.

Components of a Video Conferencing System

Intuitive: Users should be able to connect quickly and easily, so that meetings can start on-time—without entering a complex code a dozen times. Reliable and high-quality: Meetings stay connected no matter the device, and all users can clearly hear each other, no matter what device you're using or they're using. Screen sharing: Readable shared screens and applications enable meeting attendees to zoom in to read more closely, as well as to use a digital whiteboard and to edit in real time. Video conferencing: With remote workers spread across geographies and time zones, team building through nonverbal communication is critical in the modern workplace. Integrated instant messaging: Converse before, during, and after meetings to keep work moving forward and to help maximize productivity. Single button sign-on: We've come full circle to the first point and to the frustration of entering complex codes to sign on to meetings. Ensure that your solution makes getting into meetings as simple as possible.

Infrastructure

Routing and switching technologies are the basis for any successful collaboration solution. Adding quality of service (QoS) mechanisms will provide this that can recognize various services and prioritize traffic according to individual needs. Cisco collaboration systems include completely virtualized deployment models where application nodes run as virtual machines on servers. does not end with virtualization. The Cisco collaboration infrastructure also includes security mechanisms that protect every component at each level, and a wide variety of tools, applications, and products to monitor and manage the network.

Peer to Peer Signaling

SIP and H.323 are examples of this protocols where the end devices or gateways contain the intelligence to initiate and terminate calls and interpret call control messages

Call Control

Setting up, tearing down, and managing calls is the primary role of these devices such as Cisco Unified Communications Manager and Cisco Expressway.

TFTP Purpose

TFTP is a critical service for SCCP and SIP IP phones. The phone uses TFTP to download their configuration files, firmware, and other data. Without TFTP, the phones simply do not function properly. When a Cisco Unified CM administrator makes a configuration change to a device, the Cisco Unified CM creates or modifies a configuration file for the device and uploads it to the TFTP server. The TFTP service therefore must be provided by one or more Cisco Unified CM servers in the cluster

Immersive systems

The Cisco TelePresence IX Series turns any conference room into a video collaboration hub by connecting teams, customers, and partners face to face at a moment's notice.

SIP Phone Registration Process

The IP phone obtains the power through the PoE or AC adapter and starts the booting process. The phone loads its locally stored firmware image. The phone learns the voice VLAN ID through the Cisco Discovery Protocol from the switch. The phone uses DHCP to learn its IP address, subnet mask, default gateway, and TFTP server address. The phone contacts the TFTP server and requests the Certificate Trust List (CTL) file. The phone contacts the TFTP server and requests its SEP<mac-address>.cnf.xml configuration file. If the phone has not been provisioned before boot time, it downloads the default configuration XMLDefault.cnf.xml file from the TFTP server. The SIP phone requests a firmware upgrade, if one was specified in the configuration file. This process allows the phone to upgrade the firmware image automatically when required for a new version of Cisco Unified CM. The phone downloads the SIP dial rules configured for that phone. The phone establishes connection with the primary Cisco Unified CM and the TFTP server end to end. The phone registers with the primary Cisco Unified CM server listed in its configuration file. The phone downloads the appropriate localization files from TFTP. The phone downloads the softkey configurations from TFTP. The phone downloads custom ringtones (if any) from TFTP.

SCCP IP phone registration Process

The IP phone obtains the power through the PoE or the AC adapter and starts the booting process. The phone loads its locally stored firmware image. The phone learns the Voice VLAN ID through the Cisco Discovery Protocol from the switch. The phone uses DHCP to learn its IP address, subnet mask, default gateway, and TFTP server address. The phone contacts the TFTP server and requests its configuration file. Each phone has a customized configuration file named SEP<mac_address>.cnf.xml that is created by Cisco Unified CM and uploaded to TFTP when the administrator creates or modifies the phone. The phone registers with the primary Cisco Unified CM server listed in its configuration file. Cisco Unified CM then sends the softkey template to the phone, using messages.

Data Conversion in a VOIP Network

The analog signal from the telephone is digitized into PCM signals by the voice coder-decoder (codec). The PCM samples are then passed to the compression algorithm, which compresses the voice into a packet format for transmission across the IP network. On the far side of the cloud, the exact same functions are performed in reverse order.

Cisco TMS Extension Booking API (TMSBA)

This extension gives developers access to Cisco TMS booking functionality. The API is also used by the Cisco TMS extensions for Microsoft Exchange and IBM Lotus Notes, and the Cisco TMSPE Smart Scheduler.

Voice-Enabled Cisco IOS and Feature Set

When configuring Cisco IOS voice features, it is important to understand the concepts of voice ports and dial peers. In the Cisco implementation of voice, voice ports define the physical interfaces and dial peers define the virtual interfaces to and from which a call is established. Cisco voice ports and dial peers are not specific features; rather, they are the foundations on which all other voice features are built. Voice ports on routers physically connect the router to telephony devices such as telephones, fax machines, PBXs, and PSTN central office (CO) switches. The router's voice-port hardware and software must be configured to transmit and receive the same type of signaling being used by the device with which they are interfacing. That way, calls can be exchanged smoothly between the packet network and the PSTN circuit-switched network. Voice technologies use dial peers to identify call origin and destination and to define the characteristics associated with a call leg. A call leg is a logical connection between two points in the connection. An end-to-end voice call consists of four call legs: two from the perspective of the source router, and two from the perspective of the destination router. The 19XX, 29XX, and 39XX Series routers run the Cisco IOS system, while the 4000 Series routers run IOS XE. IOS XE is an updated version of Cisco IOS, which is based on an open architecture and is more secure and resilient than previous versions of IOS. All new Cisco hardware will eventually run IOS XE. The good news is that the configuration differences between IOS and IOS XE commands are small.

IP phones

Wired and Wireless phone solutions, including the 3900, 6900, 7800, and 8800 series devices.

Room endpoints

allow for customization of meeting rooms with video conferencing to evolve team collaboration. These endpoints include the Cisco Webex Board and Cisco Webex Room series.

Bearer control

are the channels that carry voice calls. Proper supervision of these channels requires that appropriate call connect and call disconnect signaling be passed between end devices. Correct signaling ensures that the channel is allocated to the current voice call and that a channel is properly deallocated when either side terminates the call. Connect and disconnect messages are carried by SS7 in the PSTN network. Connect and disconnect message are carried by SIP, H.323, H.248, or MGCP within the IP network.

Video Codecs

examples of this software codecs are Cinepak, MPEG-2, H.264, MPEG-4, and H.265 or VP8.

Gateway

interfaces with public switched telephone networks (PSTNs) and VoIP networks. also provide physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and private branch exchanges (PBX

Cisco Telepresence Management Suite (TMS)

is a portal for managing and monitoring video conferencing network from a single, structured interface. provides centralized control for on-site and remote video systems, and a deployment and scheduling system for the entire video network. It has an SNMP and HTTPS relationship with both endpoints and infrastructure devices. It can also perform backups, restores, and upgrades, and push configuration templates to any installed devices. automates system configuration for a basic telepresence network, operating right out of the box. behavior is customizable to suit an organization's needs, set up user permissions, and configure a network model so that all of call-handling functionalities are available. As a managing and monitoring application, is not aware of the call signaling that occurs between the call control device and the endpoints. The call control device (for example, Cisco Unified CM and Cisco Expressway) is responsible for the signaling (SIP, H.323, RTP, and so on) and can accept or reject the call setup and media path. From a networking perspective, must be accessible to all the devices in its database. This access includes endpoints possible located outside the firewall. Scalable provisioning: offers rapid, large-scale deployments of up to 100,000 Cisco TelePresence users, endpoints, and soft clients across disparate customer locations, including up to 5,000 direct-managed endpoint and infrastructure devices. Centralized administration: automates and simplifies the management of Cisco TelePresence meetings and Cisco collaboration infrastructure resources, reducing an enterprise's total cost of ownership (TCO). Flexible scheduling: makes scheduling Cisco TelePresence meetings more accessible with a range of tools. The tools include a simple and intuitive web Smart Scheduler, Microsoft Exchange and Outlook integration, and advanced booking capabilities for experienced concierge administrators. One-Button-to-Push (OBTP): This feature makes it easy to join a meeting when scheduling resources on premises with Cisco Meeting Server and Cisco TelePresence Server or with Cisco WebEx Video (Cloud CMR) meetings. Phone book management: The robust and flexible phone book feature in supports synchronization with a wide range of directories, including external sources for easy contact management.

Cisco Meeting Server

is an on-premises high-performance, scalable platform for SIP protocol-based audio, video, and web conferencing. It interoperates with a wide variety of third-party products, including Microsoft Lync or Skype and the Avaya call control device. With the Cisco Meeting Server, people connect regardless of location, device, or technology. software is optimized to run on Cisco Meeting Server 1000 and 2000 appliances, which are based on Cisco Unified Computing System (UCS) technology. Cisco Meeting Server 1000 is a preconfigured version of the virtualized Cisco UCS C220 M4 or M5 Rack Server that supports up to 96 simultaneous 720p HD video conferencing calls and 2200 audio calls. Cisco Meeting Server 2000 runs on a preconfigured Cisco UCS 5108 Blade Server Chassis with eight Cisco UCS B200 M5 Blade Servers that supports up to 700 simultaneous 720p HD video conferencing calls and 3000 audio calls. In addition to the conferencing features, Cisco Meeting Server has extra functionality that can be enabled for more advanced deployments. Cisco Meeting Server supports single-server deployment and cluster deployment.

Signaling

is the capability to generate and exchange control information that will be used to establish, monitor, and release connections between two endpoints. Voice signaling requires the capability to provide supervisory, address, and alerting functionality between nodes. The PSTN network uses Signaling System 7 (SS7) to transport control messages. SS7 uses out-of-band signaling, which is the exchange of call control information in a separate dedicated channel. VoIP presents several options for signaling, including H.323, Session Initiation Protocol (SIP), H.248, Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP). Some VoIP gateways are also capable of initiating SS7 signaling directly to the PSTN network. Signaling protocols are classified as either peer-to-peer or client/server protocols. SIP and H.323 are examples of peer-to-peer signaling protocols where the end devices or gateways contain the intelligence to initiate and terminate calls and interpret call control messages. H.248, SCCP, and MGCP are examples of client/server protocols where the endpoints or gateways do not contain call control intelligence but send or receive event notifications to a server commonly referred to as a call agent. For example, when an MGCP gateway detects a telephone that has gone off hook, it does not know to automatically provide a dial tone. The gateway sends an event notification to the call agent, telling the agent that an off-hook condition has been detected. The call agent notifies the gateway to provide a dial tone.

Quality of Service

is the set of techniques that are used to manage network resources by controlling the bandwidth, delay, delay variation (jitter), and packet loss parameters. Networks that transport a multitude of applications must provide predictable, measurable, and sometimes guaranteed services by managing bandwidth, delay, jitter, and packet loss parameters. Bandwidth is a data transmission rate. It is the maximum amount of information in bits per second that can be transmitted across a network. Delay, which is also referred to as latency, is the finite amount of time that it takes a packet to reach the receiving endpoint after being transmitted from the sending endpoint. In the case of voice, this is the amount of time it takes for a sound to travel from the speaker's mouth to a listener's ear. Jitter, which is also referred to as delay, is the difference in the end-to-end delay between packets. For example, if one packet requires 100 ms to traverse the network from the source endpoint to the destination endpoint and the following packet requires 125 ms to make the same trip, the delay variation is 25 ms. Packet loss is a relative measure of the number of packets that were not received compared to the total number of packets transmitted. During periods of congestion, however, the mechanisms can determine which packets are more suitable to be selectively dropped to alleviate the congestion.

Converged Networks

known also as triple-play services networks, are networks that can transmit data, voice, and video, or any combination of these services over the same networks. are different from traditional networks, where data networks were limited to exchanging character-based information between connected computer systems, and telephone, radio, and television networks were maintained separately from data networks. Every one of these services required a dedicated network, with different communication channels and technologies to carry a communication signal. Each service had its own set of rules and standards to ensure successful communication. In a converged network, there are still many points of contact and many specialized devices, such as personal computers, phones, TVs, and tablet computers. However, there is one common network infrastructure. This network infrastructure uses the same set of rules, agreements, and implementation standards

Application servers

provide services such as voicemail, unified messaging, and Cisco Communications Manager Attendant Console

Codecs

provide the coding and decoding translation between analog and digital facilities. Each codec type defines the method of voice coding and the compression mechanism that is used to convert the voice stream. The PSTN uses TDM to carry each voice call. Each voice channel reserves 64 kbps of bandwidth and uses the G.711 codec to convert an analog voice wave to a 64-kbps digitized voice stream. In VoIP design, codecs might compress voice beyond the 64-kbps voice stream to allow more efficient use of network resources. The most widely used codec in the WAN environment is G.729, which compresses the voice stream to 8 kbps.

Gatekeeper

provides Call Admission Control (CAC), bandwidth control and management, and address translation

Videoconference station

provides access for end-user participation in videoconferencing. The videoconference station contains a video capture device for video input and a microphone for audio input. A user can view video streams and hear audio that originates at a remote user station.

Call agent

provides call control for IP phones, CAC, bandwidth control and management, and address translation. Unlike a gatekeeper, which in a Cisco environment typically runs on a router, a call agent typically runs on a server platform. Cisco Unified Communications Manager is an example this

Collaboration applications

solution incorporates various advanced applications and services that integrate with Cisco Call Control applications. These applications include the following: IM and Presence: The Cisco IM and Presence Service will enable Cisco Jabber, Cisco Unified Communications Manager applications, and third-party applications to provide instant messaging and presence services. Voice messaging: Cisco products provide several voice messaging options for large and small collaboration customers, and the ability to integrate with third-party voicemail systems, using standard protocols. Customer contact: Cisco Contact Center solutions provide intelligent contact routing, call treatment, and multichannel contact management for customer contact centers