by Brian Morgan, Shane Lisenbea, Michael Popovich

Chapter 1: What Is Unified Communications?

Analysis
Jun 30, 201037 mins

Excerpt from Cisco Unified Presence Fundamentals.

By Brian Morgan, Shane Lisenbea, & Michael Popovich

Published by Cisco Press

ISBN-10: 1-58714-044-6

ISBN-13: 978-1-58714-044-0

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Chapter 1: What Is Unified Communications?

The intent of this chapter is to provide a high-level overview of Cisco Unified Communications (UC) as a collaborative solution that enables its users to perform their job functions more efficiently. Increasing efficiency decreases the amount of time required to perform the same job function. Depending on circumstances, these decreases can be dramatic.

The definition of Unified Communications is different based upon the vendor’s solution being defined and how it is implemented. This type of definition is unique and can confuse the topic because each vendor can have their own nomenclatures, so industry terms and definitions are necessary. The industry’s definition of Unified Communications follows:

Unified communications (UC) is the integration of nonreal-time communication services such as unified messaging (integrated voicemail, e-mail, short message service [SMS], and fax) with real-time communication services such as instant messaging (chat), presence information, IP telephony, video conferencing, call control, and speech control. UC is not a single product but a set of products that provides a consistent unified user interface and user experience across multiple devices and media types.

The root of the preceding definition is that a UC solution is a convergence of real-time and nonreal-time services that are equally usable across multiple means of access.

This chapter briefly covers some basic background information regarding the following:

  • Telephony history

  • Private Branch eXchange (PBX) architecture

  • IP-based call control

Some discussion then follows regarding the user experience as it pertains to legacy telephony technologies then and Cisco UC now and how Cisco is simplifying the means by which companies deploy the various technologies available on a user-by-user basis. By basing the licensing model on the users and their chosen workspace, a more efficient, fluid deployment model can emerge.

Cisco UC is far from simply dial tone. The Time Division Multiplexing (TDM) technologies of past decades are quickly becoming architectural fossils. Many of the traditional TDM vendors have made attempts to seize a part of the UC market, but their successes are largely contained to their so-called loyal-at-all-costs clientele.

Before any discussion of UC can really take place, a frame of reference must be established. To accomplish this, a discussion regarding past and present communications is in order.

Telephony History

Telephony as we know it goes back quite a while. In fact, it goes back farther than most people care to explore. Was Alexander Graham Bell the first to produce the telephone in 1876? Or the more pertinent question, was he the first to file the patent? Much controversy surrounded Bell’s invention. One Elisha Gray might have been very much in favor of Bell meeting with some misfortune. But, alas, it was not to happen. Although Elisha Gray’s patent for the telephone might have proved a workable solution, Bell had one critical element that Gray did not possess. The simple knowledge of speech and linguistics. Bell’s mother was hearing impaired. His grandfather, father, and brother were all noted teachers of elocution and speech (what we know today as the study of linguistics). Bell and his father were largely responsible for what would become the International Phonetic Alphabet. Bell understood what it took to not only speak but to also make speech. Speech and speaking were the core of the family business.

Interestingly, Gray wasn’t the only one who wished Bell ill. Thomas Edison was actually hired to take Bell down by any means necessary. Edison made a number of motions toward that goal, but we ended up with inventions such as the microphone due to his efforts.

History as it was and as we know it are likely two quite different things. For a pointed example of this, spend a few minutes exploring who actually invented the telegraph years before Morse entered the picture.

Suffice to say that Bell, et al, gave us the analog telephony technologies that were lacking to that point. Where did the digital side come in? In 1924, a gentleman by the name of Harry Nyquist, of Bell Labs, was one of two men who developed the first publicly available fax machines. In 1927, he taught us that the number of pulses that could be put through a telegraph channel per unit time is limited to twice the bandwidth of the channel. This would later be adapted into the Nyquist-Shannon theorem that is the basis of nearly all telecommunications. A range of 0–4000 Hz would be sampled 8000 times per second in each channel. Each sample is 8 bits in size. The result of 8000 samples at 8 bits each comes out to 64,000 bps or what we know as a DS0.

The digital hierarchy of T1/E1 is well out of scope for this discussion, but it is relevant because it is still widely used today. These bandwidth rates were set in excess of 80 years ago, and we still know that one timeslot has a bandwidth of 64 kbps. We know that 24 timeslots create a DS1. We know that 30 timeslots create an E1. These are transmitted utilizing a TDM algorithm such that they are all processed in tandem.

Why is history so important? That’s such a clichèd question and it appropriately has a clichèd answer. Those who don’t know history are doomed to repeat it. This isn’t always a bad thing, but it is usually an inefficient thing. The technologies that got us to this point were amazing in their time. They are time tested and highly dependable. But time goes on and technology advances.

Why hasn’t this technology advanced in more than 80 years? The short answer is a resounding, “IT HAS!” It took a forward-thinking company whose comfort zone is the so-called bleeding-edge of technology to bring all the pieces together into a new way of looking at telephony. This book is a discussion of one facet of those advances.

PBX Architecture and Call Control Basics

With the all-too-brief history lesson aside, some discussion of the technologies that got us to this point is in order. The concept of call control refers to the software and logic involved in providing telephony features including, but not limited to, call setup, call tear-down, and the various features we’ve all come to associate with telephony systems (hold, transfer, park, and so on). TDM technologies are still in widespread use throughout the planet. Many have been in production for 15 years or more. The majority of traditional Private Branch eXchange (PBX) switches share the common underlying architecture illustrated in Figure 1-1.

Figure 1-1

General PBX Architecture

As seen in the figure, the essential components of the PBX include

  • Backplane: The central infrastructure that interconnects the telephony components of the PBX

  • Processor: A collective term referring to the switching fabric, path selection algorithms, and other components related to call routing.

  • Line cards: Mission-specific interfaces for digital and analog phones or other endpoints (faxes, modems, and so on)

  • Trunk cards: Mission specific interface for PSTN, applications, interactive voice response (IVR), PBX interconnection (also known as tie lines), and related connectivity

All these components play their part in the overall telephony picture.

Inbound calls enter via trunk lines, whether analog or digital. The manner of entry makes little difference to this general overview. The call signaling is transmitted across the backplane to the processor for path selection. The PBX is simply another type of switch capable of much the same process as a LAN switch or a router. The underlying process is similar. Information enters through an ingress port. This information carries unique identifiers for both the source and destination. Based on the destination information, a route lookup is performed and path selection is made resulting in the selection of an egress port. At that point, a path switch is performed, and the relevant information or portion thereof is sent out the selected port. The processes are somewhat similar, at least in concept, in both the voice and data realms.

The PBX processor makes the pathing decision for the call based on the Called Party Number (CdPN). The Calling Party Number (CgPN), while included in the signaling information, is not utilized for path selection. At this point, the call will be forwarded on to the egress port, be it another trunk port (if the CdPN exists as an extension of another PBX) or to a line card associated with a specific phone that owns the CdPN (resulting in ringing). When the CdPN phone is answered, the voice path is established and the call is active. Figure 1-2 illustrates the path of both the signaling and the voice.

Figure 1-2

Signaling and Voice Path Through the PBX

The essential purpose of a PBX is to provide dial tone and essential call features such as transfer, hold, and so on. After it has provided these base services, the capability of the PBX to expand functionality can rapidly hit a brick wall, so to speak.

IP-Based Call Control

In the past 15 years, a movement away from TDM and toward IP has erupted within the voice networking realm. This move has not been accepted easily by everyone. Some still hold out in their blatant refusal to adopt the future of communications.

The first real application of the idea of sending voice traffic across an IP network came in the form of IP Trunking. That is, the IP network provided a mechanism that made it possible for traditional PBX network administrators to eliminate point-to-point tie lines and utilize the data infrastructure in passing voice traffic between PBXs within the enterprise. Voice traffic would be passed to the IP network via voice gateways (typically H.323 gateways). These gateways are simply routers with additional code built into their operating systems that enable the recognition and routing of voice traffic through the use of statically configured dial peers.

IP Telephony (IPT), as an independent call control capability, began simply as an IP-based PBX. Its initial aim was to provide a means of replicating the services offered by a TDM PBX but on an IP-capable platform. The removal of the TDM backplane from the architecture didn’t change the need for the core components discussed in the previous section. They are still necessary. Obviously, they present themselves in different forms than previously described.

Each component is illustrated in Figure 1-3 as it relates to both the previous discussion and the IPT architecture.

In the simplest possible terms, voice has become another application traversing the data network. That is not to say that it can be treated as simply another application. It still remains highly sensitive to latency and jitter. It must be protected throughout the network as a critical traffic type. The same is true for video traffic, which will be discussed briefly later in this chapter.

The Network Is the Platform

As shown in Figure 1-3, the core components of the traditional PBX architecture are still in existence. What has changed is the way in which these services are offered. The network is the platform that enables all the applications seen as mission-critical. It is now the platform that enables voice, video, and collaborative technologies of which the PBX could only dream.

Figure 1-3

IPT Architectural Overview

As an IP-based entity, the voice network now has all the layers of redundancy and protection afforded to data traffic. This includes convergence capabilities, quality of service (QoS) capabilities, traffic engineering, and more. One of the single largest benefits of an IPT infrastructure is the reduction in overall resources necessary to provide like-for-like services. No longer are tie lines necessary as the network provides dynamic routing capabilities in reaching site-to-site. The entire legacy voice network can be decommissioned and the associated costs reclaimed.

Referring to Figure 1-3, one detail of note is that the trunk and line cards are no longer dedicated resources in a PBX cabinet. They are voice-capable routers and LAN switches, respectively. These devices are not mission-specific to voice. The voice-capable router, for example, is known as an Integrated Services Router (ISR), which also has virtual private networking (VPN), firewall capabilities, voice, IP routing, and more built into it. The ISR can be a fully self-contained, secure system for remote offices. Furthermore, the ISR can provide LAN switch ports with Power over Ethernet (PoE), call control, VPN connectivity (for both clients and point-to-point connections), firewall services, and more all in a single chassis.

The LAN switches are now largely Layer 3-aware and capable of intelligently providing power to various endpoints as needed based on their requirements. The network as the platform has extended not only reachability but also capability of the typical network device. Figure 1-4 illustrates the signaling and voice media pathways in an IPT deployment.

Figure 1-4

Signaling and Media with IP Telephony

Cisco Unified Communications is centered on the concept of Services Oriented Network Architecture (SONA). This enables the intelligence to be placed into and leveraged from the network.

The intelligence leveraged can be accessed through more than just traditional means, that enables a user of a UC-enabled system direct access to information previously restricted to less than mobile components. The services are composed of call control, Presence, voicemail, video, Instant Messaging (IM), and web collaboration.

Cisco Unified Communications Manager

Call control is the body and soul of Unified Communications. Unified Communications Manager performs the Cisco call control functions that enable through signaling and discovery protocols the processing and management of gateways and dial plan, directory, directory integration, application integration, and desperate PBX integration.

From the beginning, the phone system inspired the need to know whom or what might be available or busy. Some phone systems (key systems) took a squared approach that would have a line key that would indicate busy or idle. A Busy Lamp Field (BLF) also translated this function to individual users to show availability. The same BLF indicators were further enhanced to provide speed dial calls to the number it was programmed to.

The BLF, along with the entrance of call centers, initiated the need for presence or the ability to see who is available, when, and on what device.

Cisco Unified Communications Manager (CUCM) is the evolution of Cisco IP-based call control. In relation to the traditional PBX, CUCM provides the processor function. It started its existence in the late 1990s as CallManager. The change to Communications Manager is relatively recent and coincided with the release of CUCM version 6.0 in the latter half of 2007. There have been significant additional changes as well. CallManager has traditionally been a Microsoft Windows-based call control element. CallManager ­version 5.0 was essentially a reengineering of the product resulting in its conversion to a Linux-based appliance.


Note – CallManager version 4.3(2) is the final release of a Windows Server–based call control solution. All future development and releases of the CallManager/Communications Manager will be on the Linux platform.


Like its predecessor, CallManager, the Communications Manager functions in a clustered architecture. This clustered appliance architecture enables the call control functionality to be spread geographically. Each appliance in the cluster is capable of running call control on its own for a varied number of phones if connectivity is lost between the cluster members. The number of phones supported varies by appliance platform (that is, Media Convergence Server [MCS] Server Model). Table 1-1 shows the MCS server options and number of phones supported per server. You need to understand that these are maximum endpoint counts and that the total number can be reduced based on configuration variables such as Computer Telephony Interface (CTI) Ports, voicemail ports, gateways, and various other possibilities.

Table 1-1  Maximum Devices per Server Platform

MCS Server Model

Number of Supported Endpoints

MCS-7815

300

MCS-7816

500

MCS-7825

1000

MCS-7835

2500

MCS-7845

7500

Far from providing simple dial tone, CUCM provides services for voice, video, mobility and presence for up to 30,000 users on a single cluster (assuming the use of MCS-7845 platforms). Chapter 2, “Cisco Unified Presence Overview,” discusses these features in more detail.

Endpoint Network Access

An endpoint is considered to be any device that makes use of the infrastructure to place or receive IP-based calls whether they are audio, video, or a combination of the two. Endpoints making use of the Cisco Skinny Client Control Protocol (SCCP) will be registered to the CUCM for all call control functions. SCCP endpoints are not the only endpoints that CUCM is capable of controlling. As an open system, the CUCM supports Session Initiation Protocol (SIP)-based phones. This includes Cisco IP Phones running SIP loads and third-party SIP-based phones. Additionally, H.323 endpoints are fully compatible with the architecture; however, they might require the installation of an H.323 Gatekeeper to route calls to/from them, but they will easily integrate.

As a bit of internetworking technology review, following are three layers of basic architecture when designing networks:

  • Core layer: The heart of the network where most services and applications will be hosted.

  • Distribution layer: Provides redundant paths to and from the Core from multiple points within the network.

  • Access layer: Houses the LAN switches into which user computers and phones will connect. No user equipment should connect to any other layer.

The aforementioned telephony endpoints can make use of the access layer devices in the network infrastructure, which might include LAN switches, wireless access points, and others where applicable. The end result of this is that the access layer devices are essentially fulfilling the role of the line cards in the traditional PBX model. They provide the phone access to the network resources it requires to place and receive voice and video calls.

Integrated Services Routers

The Cisco core business has always been routing and switching. If routing and switching can be considered the Cisco central nervous system, the gateways are the mouth, eyes, and ears that allow information in and out.

Gateways open up communications between systems via data, voice, and video. With the continued use of industry protocols, such as H.323 and Session Initiation Protocol (SIP) and their capabilities, additional information (such as Presence, IP Phone Applications/Services, and so forth) is now being passed along or is capable of being passed along that further enriches a UC environment.

Before Unified Communications came to be known as an industry term, the roads to the UC path were already being paved.

Phone systems could share calls or call paths through trunking capabilities as with TIE trunks. Networking phone systems began to allow multiple vendor solutions to work together. Networking with voicemail and other applications solutions became necessary. This all points to the early need to federate (that is, translate and exchange) information or share information across multiple entities in a company or across multiple companies.

Call control and messaging marked the beginning of the shift from traditional (or TDM) services to what would become Unified Communications. Software, services, and other applications began a steady migration into the network as the natural side effect of the versatility offered by an IP network.

Microsoft and IBM, along with a few other companies, made or helped create popular e-mail and instant messaging through their suite of desktop software. With e-mail and IM, the top two means of nontelephone-based communication, networking, or communications between multiple companies and vendor solutions became and still is a required way of doing business as the proficiencies gained were too large to ignore.

UC’s second path was solidified when IM clients brought into being the centerpiece of UC that is Presence. The ability to change one’s availability dynamically based on time, schedule, or simple preference caught on and spread quickly.

Gateways have supplied the roads in which UC travel across as they supply the interfaces and protocols that allow for communications and information exchange to happen.

Several gateways can be used for these types of solutions and Cisco has several, but the most common in Cisco are referred to as Integrated Services Routers (ISR).

ISRs and the newly announced ISR G2 routers are multipurpose routers. The ISR routers include 2800 and 3800 model routers in which the ISR G2 routers include 2900 and 3900 router models. They provide the capability to optimize multiple services into a single platform to deliver a consistent user experience. The consolidation of services into a single chassis reduces the overall footprint of the devices traditionally necessary to provide those same services. Services in the ISR platform include

  • Routing: Traditional internetwork routing capabilities.

  • Switching: LAN switching blades and modules.

  • Unified Communications: Fully self-contained or Survivable Call Control. That is, the ISR can stand on its own as a primary call control entity, or it can provide resiliency services to phones that exist in the site where it is located. If connectivity to the primary call control entities is lost, specific code can be invoked on the router that enables the local phones to utilize it for call control purposes until connectivity is restored.

  • Integrated security: Full firewall and VPN connectivity.

  • Application performance: Bandwidth/WAN optimization and acceleration.

  • Management: GUI administration and configuration toolset.

  • Mobility: Integrated wireless services including 802.11a/b/g and 3G capabilities.

Obviously, for the purposes of this book, the focus is on the UC portion of the functionality. An ISR can provide full call control functionality to a number of phones ranging from 25 to 300 depending on the ISR platform in question. This is particularly useful in branch deployments where each branch or remote site is an island and maintains its own dial plan. The call control functionality can be configured via IOS command line or via web GUI.

If this is not the desired topology, the same ISR platform can undergo a simple configuration change to become a Survivable Remote Site Telephony (SRST)–capable device. That is, it becomes the local call control device if connectivity is lost to the CUCM cluster. In this role, basic call control capabilities are provided for up to 1200 phones, depending on the router model.

The ISR can fulfill the role of the trunk card in the traditional PBX architecture. Both analog and digital voice connections to the ISR can be accomplished in a number of ways. The same interfaces are used regardless of whether the connection will be made to the PSTN or to a legacy TDM PBX. Digital T1/E1 cards and Digital Signal Processor (DSP) resources can be installed into the router to provide digital connectivity. The DSPs are the call handlers, the basic function of which is to convert the call between TDM and IP. If analog connectivity is wanted, the use of Foreign Exchange Office (FXO) ports enable connection to the PSTN or PBX, whereas Foreign Exchange Station (FXS) ports can function as analog phone/fax/modem connections and Direct-Inward-Dial ports.

Applications

Applications embody the intelligence of a UC solution. They are sometimes canned, off-the-shelf applications and other times they’re custom applications developed with a purpose in mind. Applications make the solution into more than just a phone system.

The applications that make up the UC solution are as follows:

  • Application Extension Platform (AXP)

  • Cisco Unified Application Environment (CUAE)

  • Conferencing

  • Unified Communications Mobility

  • Voice messaging

  • Unified Presence

Application Extension Platform

Cisco AXP is an application services hosting platform that uses the Cisco ISR as the physical platform. Leveraging the modularity of the ISR and ISR G2, a number of different physical configurations of this module have been made available based on size and scope of processing demands to be made on the developed applications. Smaller versions would suffice for smaller footprints and embedded applications where high-end versions are specifically designed for applications requiring extensive processing capabilities and additional memory, high availability, and so on.

The Cisco AXP provides a service platform on which applications can run. The Cisco AXP ships with a virtualized hosting environment and a host of monitoring and configuration application programming interfaces (API) available to the applications running on the module.

The product also offers a software development kit (SDK) that enables functions and a management interface that allows for a centralized management environment. The SDK provides all the tools needed in packaging, hosting, and integrating applications into the router. This includes the ability to utilize third-party applications, Linux .rpm packages, and application troubleshooting. A CLI extension API provides tools needed in extending the AXP CLI with custom commands. The SDK also includes a number of examples of source code illustrating the usage of APIs.

The Cisco AXP bridges application systems with network systems. At an architectural level, the capability to combine two loosely coupled systems into an integrated solution has numerous advantages. Applications such as desktop/server management solutions, branch-embedded call recording for banks, secure healthcare records solutions, bill payment engines, fax over IP, and more help overcome business obstacles by meeting specific needs of an industry or vertical. These enhancements contribute directly to tighter business processes and models that keep the customer competitive.

CUAE

The Cisco Unified Application Environment (CUAE) provides a ready-to-use application suite and custom development tools that enable you to integrate UC into other applications or create your own specific applications.

A few applications are available today, with more being added on an ongoing basis.

The CUAE is composed of the following:

  • Cisco Unified Application Designer (CUAD): A visual integrated development environment (IDE) that makes creating your own applications easy through an interface that doesn’t require previous development experience

  • Cisco Unified Application Server (CUAS): An application server for converged applications that simplifies telephony protocols, provides reliability by mediating between the application and Communications Manager, and provides standard management for applications

  • Cisco Unified Media Engine (CUME): A software-only media server that provides off-the-shelf and media processing for all applications built with CUAD

Conferencing

Cisco Unified MeetingPlace is a collaboration tool that brings together voice, video, and web for Cisco UC. Cisco Unified MeetingPlace supports organizations by providing an effective means of communication enabling people to meet at any time from anywhere. The solution is deployed on-premises over a customer IP network.

Cisco Unified MeetingPlace also works as an audio bridge for WebEx, by keeping audio conferencing on-premise. WebEx, like MeetingPlace, provides a platform for audio and web collaboration, with the difference being that WebEx is software as a service application, or what some refer to as a cloud service.

The Cisco portfolio in audio and web collaboration is now robust as both MeetingPlace and WebEx can augment each other’s capabilities as a single product.

UC Mobility

The progression of telecommunications to the point where it is today shows the evolutionary growth where UC came from and where improvements can be made for the next version or generation of communications. Early telecommunications devices have evolved from rather large and awkward devices to the sleek, stylish, and highly portable smart devices we use today. In most cases, the use of both static and mobile devices is commonly used today; however, the communication habits of the next generation of information workers will continue to change.

The trend to not leverage devices such as desktop phones, which force users to be in a static location, has been a key driver for the transition into a user requesting and sometimes demanding the flexibility of being connected anywhere, anyplace, or anytime.

Hard phones, wireless-capable devices, cellular networks, and smart phones have literally unleashed a new work force for today and in the future. The traditional “workplace” restricted workers to a desk—the now “workspace” happens anywhere and anytime the request is made and in moments in time depending on where the worker is currently located. These architectural enhancements now reduce the process time by increasing the overall capabilities and availability, which is the importance of UC.

Accomplishing anytime, anywhere access through mobility required that the capabilities and functionality offered with the traditional workplace communications was critical for mobile communications to be successful. The cell phone and its capabilities began to expand into other areas such as texting, multimedia messages, and so on. PDAs such as the iPAC from HP or Palm Pilots started the mobile application creation that affected many business verticals such as healthcare, transportation such as UPS or FedEx, among many others.

The cell phone, smart phone, PDAs, and other attached devices converged together to give a users an easy-to-carry application platform that has fostered the business transformation on when, where, and how a user performs their roles.

Voice Messaging

Voice messaging is one of the older applications that has been incorporated into the UC product suite.

The ability to communicate obviously is important, but where messaging shines is that it enables multiple means of communication from a caller to the intended recipient. This type of communication was huge because it allowed for a more mobile workforce and began to unlock users from their static cubicles. Not having to answer every call so there were no gaps in the communication chain added productivity benefits that were hard to ignore. Because this was a game changer, it quickly caught the attention of companies, which is what eventually took an answering machine to the heights of Unified Messaging.

Increasingly, users demanded more functionality to be available to further enable users to be multimodal. This ushered in features such as the following:

  • Dropped call recovery: The capability to be placed back in the same point of the conversation where it was interrupted

  • Message monitor: The capability to listen to a message being left and then choose to take the call or let the caller leave a message

  • Notification: The capability to be notified of a message that was left for the user by Message Waiting Indicator (MWI), Alpha Numeric Paging, Short Message Service (SMS), or Simple Mail Transfer Protocol (SMTP)

Knowing you could be away from the desk empowered the workforce to multitask in new ways. Users becoming increasingly more mobile meant that the tools that were used needed to be more flexible and friendly to devices that fostered a decoupling of the user from the desk. Voice messaging took the natural evolution to incorporate itself in the corporate infrastructure so as to reduce administration and take advantage of the network and its components.

Unified Messaging simplified the capability to manage multiple means of communications such as e-mail, voicemail, and fax messages by taking multiple directories and using just one, taking multiple message stores and making them one, increasing the access and availability to messages and leveraged security, and authentication to reduce the complexity while still offering a secure means of access to better the user experience.

Unified Presence

Presence is the aggregation point of availability represented by a single or multiple devices associated with a user account that can be monitored by others.

Presence is the glue that uniquely folds together real time and nonreal-time communications by incorporating capabilities that in themselves are not new but can be leveraged in new ways.

Although mobility drove the release of users from their desks, it created new challenges to address, such as how best to reach a resource or what form of communication to use.

Presence was the answer to this by first leveraging a directory for contact information. Having a single source of the available users gives the user access to the entire workforce for all potential resources. This scales the addressable potential for a quicker means of resolution.

Presence preference capabilities provide the means of what device a particular user prefers to be communicated with. Knowing how each user wants to be communicated with takes out the guesswork of what will be the fastest means to communicate with a user.

Availability shows the status of the user whether they are online, offline, idle, busy, or away. Availability increases the time to resolution by enabling the user to reach out to a source that is available. It also helps to ensure that interruptions don’t stall other work by users that are currently engaged.

The capability to quickly see who is or isn’t available and on what device increases productivity, scales an organization’s resources, and speeds resolution by reaching out to the right user at the right time with the device that is best for both ends to accomplish the task at hand.

User Experience

So, where has all this discussion of history and technology gotten us? We understand now that things are changing with the times and that phones are no longer just phones. They’re network endpoints. The idea of having an office desk phone number, cell phone number, work phone number, voicemail phone number, and who knows how many others is simply reprehensible. It’s inefficient for us as workers and for our colleagues when they need to reach us by trying each number in series. The next step is to expand the idea of what the “network” entails. The network is now an entity that is dynamic in nature. We access the network in a multitude of ways every day.

Every time you pick up your phone and tweet your status or update your Facebook page, you make use of the network. Your phone is, of course, a phone, but it is also your organizer, e-mail client, calendar, and who knows what else. It has become one of the primary means of accessing network resources. Shouldn’t that experience be identical to the experience you’d have if you were sitting at your laptop or at your office desk? Therein lies the core idea of UC—one network experience regardless of means of access.

In a given day, you might use your cell phone, desk phone, home phone, softphone, and more. They should all be connected and aware of each other to some degree. I don’t want multiple phone numbers and voicemail boxes to deal with. I want many devices, one phone number, and one voicemail box. I want to tell people how and where I want to be reached; that is, if I want to be reached at all. This find me, follow me, hide me capability is at the very core of Presence and therefore this book.

Presence is a real-time indicator of a person’s willingness and availability to communicate. This is typically represented by status such as Available, In Meeting, On the Phone, At Lunch, or any other manner in which people might want to communicate their current status. Presence also includes details on users’ preferred method of contact, be it IM, voice call, video call, or other.

However, UC doesn’t stop there. It is all-encompassing with regard to means of communication. It includes integrations to phones, e-mail clients, calendars, and conferencing resources (including web, audio, and video conferencing). It includes IM and the capability to know how to reach someone the first time through the use of presence indicators that can be set manually or dynamically based on an Outlook schedule. It includes an integration to various models of cell phones enabling the use of the cell phone as if it were the desk phone. This includes contacts, call logs (received/missed/placed calls), and the ability to check the office voicemail box. The voice messages are presented in a list form so that they can be accessed in desired order and played via the cell phone. When played, the audio is streamed over the data channel so that no cellular airtime is utilized. All these features come together so that you have only a single voicemail box to administer, if so desired. When a call is made to your desk phone number, your cell phone rings simultaneously. If that call is not answered, it is taken back, and the caller is forwarded to your office voicemail. There’s no longer a need to publish multiple phone numbers. This is called Single Number Reach (SNR).

To further augment the SNR functionality, the user has the ability to move the call between phones at-will. For example, a call is answered on the desk phone but the conversation is taking longer than expected. With the push of a button on the desk phone, the call can be seamlessly transferred to the cell phone allowing the user the freedom to depart without missing any part of the call or having to reestablish the call. The opposite situation is also true. If a call made to the office phone number is answered on the cell phone, the call can be seamlessly transferred to the desk phone by simply hanging up the cell phone and picking up the desk phone handset.

In both cases, the call is passed seamlessly between the two phones.

Clients

Like any other solution, UC needs to have a front end or GUI that gives access to the services in a way that is operational without a lot of complexity. So being intuitive is a requirement; otherwise, the usefulness doesn’t outweigh the cost benefits.

The clients in the Cisco Unified Communications solution consist of the following:

  • Cisco IP Communicator (CIPC)

  • Cisco Unified Communications Integration for Microsoft Office Communicator (CUCIMOC)

  • Cisco Unified Mobile Communicator (CUMC)

  • Cisco Unified Personal Communicator (CUPC)

  • Cisco Unified Video Advantage

CIPC

Cisco IP Communicator (CIPC) is a Microsoft Windows application that delivers a software-based phone that supports the same capabilities as the desk phone through personal computers. This application enables computers with the functionality of IP Phones to provide voice calls on the road, in the office, or from wherever users have access to the corporate network.

The functionality of Cisco IP Communicator is designed for when users aren’t just trying to take their office extension with them. Besides the capability to make and retrieve calls, users will also have access to the phone services they have in the office. This advantage boosts a business’s capabilities for collaboration and responsiveness, and helps organizations keep pace with today’s work needs by enabling the flow of business to happen whether the user is in the office or elsewhere.

CUCIMOC

Cisco UC Integration for Microsoft Office Communicator (CUCIMOC) is an application based on the Cisco Client Services Framework that provides access to Cisco Unified Communications services such as soft phone, mid-call control, messaging, conferencing, desk phone control, and phone presence directly from a tabbed interface in Microsoft Office Communicator.

Cisco UC Integration for MOC augments Microsoft Office Communicator’s IM capabilities with call control and access to other applications supplied by CUC solution.

CUMC

Cisco Unified Mobile Communicator (CUMC) is an application applied to mobile handsets that extends enterprise communications capabilities and services to mobile phones and smart phones. By streamlining the communication experience, CUMC has enabled users with real-time collaboration across the enterprise regardless of whether they are in the office.

With CUMC, users benefit from being allowed to perform the following tasks:

  • Place and receive calls

  • Access company and personal directory information

  • View busy or available status

  • Securely send text messages

  • Receive and play back voicemail messages

  • View a list of messages and select the one they want to play back

  • Conference and collaborate through integration with Cisco Unified MeetingPlace and WebEx

  • Access recent call histories

CUPC

Cisco Unified Personal Communicator (CUPC) transparently integrates a wide set of applications and services into a single application whether on a PC or Mac. CUPC provides easy access from a single interface to communication services such as the following:

  • Call control

  • Video

  • Instant messaging

  • Web conferencing

  • Voicemail

CUPC enables you to easily communicate with co-workers, partners, and customers from your office or on the go using the integrated soft phone or controlling the desktop phone and by sharing availability information and IMs inside or outside your business or between businesses.

Additionally, CUPC enables you to perform one-click escalations, which enable you to start with one form of communication and quickly incorporate other capabilities. For example, you could start with an IM and then one-click escalate to audio, audio with video, and then web collaboration with audio and video.

CUPC further enables the user with the ability to Click-to-Dial from other applications such as Microsoft’s Outlook, Word, Live Meeting, Excel, Power Point, or from the web.

Click-to-Dial is an extension enabled to take advantage of call control and Presence while working in other applications, such as those in the preceding list.

Cisco Unified Video Advantage

Cisco Unified Video Advantage (CUVA) adds video to your environment by providing video telephony functionality to Cisco Unified IP Phones and Cisco IP Communicator soft phones. With CUVA, video telephony is incorporated into Cisco Communications Manager to be leveraged by being as easy as just making a phone call.

Because the Cisco solution is IP-based, enterprise organizations can take advantage of their existing networks to extend video to everyone in their organization. This also lends itself to a more rich communications environment for remote users.

User Workspace

When you walk into the office, wherever that might be (the definition varies greatly these days), and sit down at your desk, you’re making use of your workspace. The workspace includes all the tools you use day to day in the course of accomplishing your job goals. It includes your computer with client software, IM, e-mail, calendaring, and so on. It includes both your desk phone and cell phone. It includes your voice messaging as well.

This is the concept behind the idea of the User Workspace License (UWL). The workspace is treated as an entity in and of itself when under licensing consideration. The desk phone, softphone, Presence, IM, voice messaging, cell phone client, web/audio conferencing, calendar integration, and even contact center agent licensing are assembled as a unit. These features can be enabled/disabled at will, but they’re all part of the overall workspace and the user experience. This simplifies the budgetary calculations associated with each user by fixing the per-user cost based on desired feature-set.

The Cisco UWL comes in four flavors:

  • Business Edition (CUCM Business Edition): Includes call control, voice messaging, unified clients, mobility, and Presence

  • Entry (CUCM): Includes call control for one phone and mobility licensing

  • Standard (CUCM): Includes call control, voice messaging, unified clients, mobility, and Presence licensing

  • Professional (CUCM): Includes call control for unlimited phones per user, voice messaging, unified clients, mobility, Presence, mobile communicator client, web/audio/video conferencing, and contact center licensing

The workspace license is a highly simplified means of implementing advanced features at significantly lower cost.

Summary

Cisco Unified Communications encompasses a wide array of products and technologies. All these interact to provide features and functionality well beyond the confines of simple dial tone. No longer is the reliance on 19th- and 20th-century technologies a requirement. PBX technologies have a long and distinguished history; however, they must evolve or fade into obsolescence.

The user experience for UC has demanded that business communication evolve internally and externally. Communication resources and applications are forced to keep up with the rapid changes that users are no longer willing to be without. The mantra moving forward is anytime, anywhere access. The user experience will evolve to include any device and any media. Network resources and applications will be utilized in the same manner regardless of whether a user is accessing those resources from the office directly, via VPN connection from a laptop, or from a handheld device (phone, netbook, or similar device).

The network must evolve to provide a stable, reliable platform for voice and video interaction in real time for all users. This includes the capability to seamlessly collaborate across a varying array of user platforms while delivering an identical experience to each user and endpoint.

The network is the only way to deliver the platform from which UC are accessed. This enables you to scale and control the reliability of the platform for which voice, video, call control, messaging, IM, Presence, and gateways interact in real time for all users. This includes the capability for seamless collaboration across a variety of user hardware while delivering an identical experience to all users. In support of this, Cisco has created an exceedingly simplified licensing model based on the idea that users can access the network in their chosen manner. The Cisco Unified Workspace License enables the users to choose their means and degree of collaborative interaction.

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