Cisco Unified Communications Manager - Global Knowledge

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Cisco Unified Communications Manager

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Cisco Unified Communications Manager Joe Rinehart, MBA, CCIE #14256, CCNP/DP/VP

Introduction

Figure 1: Cisco Unified Communications Media Convergence Server www.cisco.com

Cisco took a substantial risk when it acquired Selsius Systems for its call-processing platform, as the technology was still new in 1998. Furthermore, the crowded communications market included well-established telephony giants such as Avaya and Nortel, which already had large followings. Complicating the picture were some of the difficulties experienced by “early adopter” types of customers, which, understandably, created some hesitation on the part of others. Even with those initial challenges, Cisco Unified Communications Manager (CUCM) carved itself a place in the communications landscape and has grown to be a dominant presence (Figure 1). Two proverbial flavors of the call processing platform currently exist, the Express version (the subject of another white paper) and the server-based CUCM, which this white paper will examine.

Server-Based Call Control PSTN

Figure 2: CUCM Logical Layout/Icons

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Selsius Call Manager—and Cisco Call Manager following the acquisition—ran on a Windows-based server platform with Ethernet network connectivity, to perform all of its call-processing tasks (see layout in Figure 2). In order to facilitate all of the necessary functions, a CUCM server interacts with handsets (sometimes referred to as endpoints in the literature) as well as other devices of various types. The functional components of the call processing are as follows:

Figure 3: CUCM Cluster

CUCM Server(s) The CUCM server is the heart of the entire communications system and performs all of the call-control functions. As mentioned, CUCM was originally deployed with the Windows operating system and remained so until CUCM version 5 was released, at which time, Linux Red Hat became the underlying operating system. For a while, other ancillary systems, such as voice messaging (the Unity platform) remained on Windows; now all systems have been migrated to Linux. Redundancy & Failover CUCM systems create redundancy and failover through the use of multiple servers running the same software version, as well as utilizing a central, synchronized database containing all configuration and endpoint information (Figure 3). This entity, referred to as a cluster, shares information between each system, as well as sharing traffic, creating a sense of load balancing as well. One device (the Publisher) maintains read-write access over the database, while the others (Subscribers) have read-only access. In almost every circumstance, the use of a single CUCM server is strongly discouraged because of the lack of redundancy it inevitably creates. Registration Every device operating within the communications environment must have some type of relationship with, or be known by, the CUCM server(s). In some cases, such as voice gateways, this involves explicit configuration (through the web-based administration pages) or a discovery mechanism. While in production environments it is wise to explicitly configure handsets, there are also dynamic discovery mechanisms. In either case, when a handset/IP phone gets plugged into the network infrastructure, it receives an IPv4 address and then attempts to register with the CUCM server. Once that process completes, the phone receives all of its configuration informa-

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tion (user, extension, button assignment, etc.) and becomes an active part of the system. If anything goes wrong with this part of the process, the endpoint will not send or receive calls.

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Figure 4: CUCM Call Processing

Call Processing From a functionality standpoint, CUCM provides all of the call-control processing in order for voice conversations to take place. This includes brokering the registration process (discussed previously), various features (call forwarding, call waiting, conferencing, etc.) and, naturally, the setup of the call itself. Two types of protocols are involved in a CUCM call, bearer (the actual call, discussed later) and signaling. Signaling protocols mimic the call setup processes in traditional telephony, and CUCM supports several of these: • H.323: The oldest and once most dominant call setup/signaling protocol, considered “heavy” (many messages used for setup) and generally being displaced by SIP (see below). • Skinny Call Control Protocol (SCCP): Originally developed by Selsius Systems, this proprietary protocol is typically only used between Cisco handsets and the CUCM system. In many cases, it is also being displaced by SIP. • Session Initiation Protocol (SIP): Most popular VoIP signaling protocol, lightweight and multipurpose, can also be used for video, instant messaging, etc. • Media Gateway Control Protocol (MGCP): Once used extensively by telephone company service providers, this protocol uses a client-server model with intelligence moved away from the end device to the control device, CUCM in this case. Signaling provides various functions in the voice communications process, from supplying dial tone to the handset, to gathering the digits to find the called party, and performing the necessary lookups and initiating the call setup process. Call signaling flows between the IP phone and the CUCM server; at this point, the handsets do not interact directly (Figure 4 above). Once call setup is completed, CUCM steps out of the way, and a voice channel is opened, using the Real-Time Transport Protocol (RTP). CUCM only gets involved again if a feature needs to be enacted or the call must be disconnected.

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Virtualization Support As is the case with many server-based applications in recent years, the CUCM system supports virtualized server instances in deployment. In its traditional design, Cisco promoted the use of pre-built servers and built hardware verification into the installation process. The ability to install CUCM in a virtual environment has been technically possible for some time, but was only supported in the last several years. At first, this involved Cisco’s newly minted Unified Computing series of servers (rack-mounted C series and blade server B series), but other environments were added later.

Handsets Since most, if not all, aspects of VoIP/Unified Communications utilize computer technology, it should come as no surprise that IP phones (also called handsets or endpoints) are, in essence, highly specialized computers. A typical handset contains several components that work together in the voice-calling process, as follows: • Digital Signal Processors: Human speech is inherently analog and thus requires conversion to digital data. Even before the advent of network-based telephony, the conversion of analog to digital was critical to the internal processes of the telephone company. An IP phone contains specialized voice conversion chips that perform this process in both directions (analog to digital and vice versa). • Integrated Switch: Contained within a Cisco IP handset is a small switch that separates voice traffic from data traffic. This allows a PC to be plugged into the back of the phone and only use a single physical port on the cabled infrastructure. • Various Buttons and Controls: Cisco IP phones almost always have display screens of some sort and keys that can be set for functions by the CUCM system itself.

Voice Gateway The CUCM system separates out the various voice communications functions to separate device, unlike the Express version, which typically performs almost all of those tasks on one device. The gateway, a specialized router with telephony interfaces, performs the tasks of sending calls to, and receiving from, the public switched telephone network (PSTN). Installed as any other adapter card in the device (along with DSPs), these interfaces are as follows: • Foreign Exchange Station/FXS: Standard analog telephony devices such as handsets or FAX machines • Foreign Exchange Office/FXO: Standard analog line connecting to the central office (incoming/ outgoing calls) • T1/E1: Digital telephony interface that processes multiple calls simultaneously, up to 24 channels/ conversations • Session Initiation Protocol (SIP) Trunk: A more recent, network-based call control protocol and connection that sends calls over a network rather than using traditional telephony interfaces

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Conclusion Unified Communications, once a cutting-edge approach to voice communications, has gained status as a widely adopted way of communicating. Leveraging a more integrated approach between data networks and voice communications, this technology brings an incredible feature set to even smaller business entities. For most companies, the Cisco Unified Communications Manager, or CUCM, can enable comprehensive voice communications with a wealth of advanced features.

Learn More To learn more about how you can improve productivity, enhance efficiency, and sharpen your competitive edge, Global Knowledge suggests the following courses: CVOICE - Implementing Cisco Unified Communications Voice over IP and QoS v8.0 CIPT1 - Implementing Cisco Unified Communications IP Telephony Part 1 v8.0 CIPT2 - Implementing Cisco Unified Communications IP Telephony Part 2 v8.0 UC-UCS v4.5 - Installing Cisco UC on UCS in a Virtualized Environment ICOMM - Introducing Cisco Voice and UC Administration v8.0 Visit www.globalknowledge.com or call 1-800-COURSES (1-800-268-7737) to speak with a Global Knowledge training advisor.

About the Author Joe Rinehart, MBA, CCIE #14256, CCNP/DP/VP is a professional trainer specializing in technology, business, and social media. He is also a successful speaker and published author, as well as a columnist for the Federal Way Mirror. He is active in the social media space, managing one of the largest groups on LinkedIn, as well as serving on the national steering committee of the Cisco Collaboration Users Group. Joe also served as president of the Seattle Cisco Users Group, serving technology professionals throughout the Puget Sound region. Joe Rinehart MBA, CCIE #14256, CCNP/DP/VP President and Chief Edutainment Officer Gracestone Professionals, LLC [email protected] Twitter: jjrinehart

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